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From: <jo...@op...> - 2007-11-16 03:22:05
|
OSBC should be able to intelligently use the appropriate interface based on the destination of the packet. It will act as a natural bridge between the two networks assuming you have a sane routing table. Let me know if you prove me otherwise. joegen sales@ER wrote: > Hello > > I read that OSBC binds to all NIC. Will it bind when one NIC is set up an Internet addr and other NIC with a private(192.x.x.x) addr. Or is there a need for a router between the public and private networks. Are there instruction? Last question is there a top level scheme (macro) that shows a network diagram with possible OSBC typical layout? > > Warren Kreckler > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > |
|
From: <jo...@op...> - 2007-11-16 03:19:32
|
What version of GCC is involved and on which OS? Can you paste the entire compilation console log? sales@ER wrote: > Hello > > I found a howto in the wiki that explained the steps. > > I have install both the download source code and then the CVS into the same /tmp directory before i compiled. > > I first ./configue the opensipstack > > then i changed into the opensbc directory. > > I ran the autoconf in the opensbc directory then ran the ./configure and then make oooptnoshared, which failed to Make > > error is in main..o cannot declare varible 'instant' to be of abstract type "OpenSBCDeamon" > > Can someone help me? > > Warren Kreckler > > P.S. i set PWLIBDIR AND OPALDIR = /tmp/opensipstack > > ------------------------------------------------------------------------- > 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: Ilian J. C. P. <ip...@so...> - 2007-11-15 03:57:20
|
Hi Whit, ATLSIP uses the G.729 library from VoiceAge. Try looking at their site for licensing concerns (http://www.voiceage.com). - Ilian Whit Thiele wrote: > If I want to use G.729 licensing in a commercial release of ATLSIP. Who and > how do I pay for the associated G.729 license cost? > > > ------------------------------------------------------------------------- > 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: <sa...@ER...> - 2007-11-14 23:00:50
|
Hello I read that OSBC binds to all NIC. Will it bind when one NIC is set up = an Internet addr and other NIC with a private(192.x.x.x) addr. Or is = there a need for a router between the public and private networks. Are = there instruction? Last question is there a top level scheme (macro) = that shows a network diagram with possible OSBC typical layout? Warren Kreckler |
|
From: <sa...@ER...> - 2007-11-14 21:35:14
|
Hello I found a howto in the wiki that explained the steps. I have install both the download source code and then the CVS into the = same /tmp directory before i compiled. I first ./configue the opensipstack=20 then i changed into the opensbc directory. I ran the autoconf in the opensbc directory then ran the ./configure and = then make oooptnoshared, which failed to Make error is in main..o cannot declare varible 'instant' to be of abstract = type "OpenSBCDeamon" Can someone help me? Warren Kreckler=20 P.S. i set PWLIBDIR AND OPALDIR =3D /tmp/opensipstack |
|
From: Whit T. <de...@wh...> - 2007-11-14 20:17:29
|
If I want to use G.729 licensing in a commercial release of ATLSIP. Who and how do I pay for the associated G.729 license cost? |
|
From: Yogesh C. <ych...@gm...> - 2007-11-13 07:25:52
|
Hi joegen,
Thanks for your Reply, I appreciate your quick reply. I have
done changes in source code and test it its seems working fine , i have
tested it with two Enumserver and one VoipProvider and it is working
according to plan.
Yogesh
On 11/13/07, jo...@op... <joe...@gm...> wrote:
>
> Ugh! I guess you are right. Indeed enum lookup failure are not handled
> correctly. Can you try the following block of code instead and let me
> know if it fixes the problem?
>
> if( routeURI.GetScheme() *= "enum" )
> {
> PStringArray enumServer;
> enumServer.AppendString( routeURI.GetHost().c_str() );
> if( !SIPTransport::ENUMLookup(
> enumServer,
> targetName,
> routeURI ) )
> {
> continue; /// enum lookup failed. continue to the next
> route
> }
> }
>
> Yogesh Chaturvedi wrote:
> > hi,
> >
> > Thanks for reply
> >
> > *Use a comma delimited list for your route instead of having separate
> > route entries. The sample route below will allow you to route to enum,
> > then to 192.168... and finally to gafachi.
> >
> > [sip:*] enum:**mydomain.com* <http://mydomain.com/>*,
> > sip:**192.168.96.112*<http://192.168.96.112/>
> > *, sip:** gafachi.com* <http://gafachi.com/>
> >
> > //////////////////
> >
> > I have tried to do and give setting in B2BUA route as :
> >
> > Route_1 : [sip:*] validnumber.com, sip:sip.gafachi.com
> >
> > but after getting Enum Lookup fail it create a URI like
> >
> > SIP/SDPRequest: INVITE
> > enum:141...@va...<141...@va...>
> > (according to wireshark)
> >
> > After looking in source code i found that :
> > In *Route.cxx *class in *FindRoute* function it do not change the SIPURI
> > structure in case of enumlookup fail and working as :
> > Route.cxx Line 377:
> >
> > it call SIPTransport::ENUMLookup(..., routeURI) function and if
> enumlookup
> > fail then it set the username as Called number and do not change the
> rest
> > SIPURI structure and create SIP INVITE Request as given above:
> >
> > *We need to find a way to call to next available route if enumLookup
> fail
> > (not call fail)*
> >
> > So as far as i think we need to do changes in source code in both case
> > either we want to use comma seperator in same route input box(GUI) or we
> > want to go on next route(next route box in GUI) if enumLookup fail.
> >
> > For time being i have done changes in this block and add a single line
> to go
> > to next route box entry to find next route search if enumlookup fail as:
> >
> > Route.cxx line Line: 391
> >
> > if( routeURI.GetUser().IsEmpty() )
> > {
> >
> > break; // To go on next route(route_2)
> >
> > }
> >
> > I am using a bool variable here to keep track if inner loop get fail due
> to
> > enumLookup fail and go to next route of outer for loop.
> >
> > is it break the original functionality(please add your comments) and
> what is
> > the Best solution for this problem.
> >
> >
> > Thanks
> > Yogesh
> >
> >
> >
> >
> >
> >
> > On 11/6/07, Yogesh Chaturvedi <ych...@gm...> wrote:
> >
> >> Hi,
> >>
> >> I want to configure OpenSbc such as it first call using enumlookup and
> if
> >> it fail then
> >> it should call through my voip provider.
> >>
> >>
> >> I have done setting in B2BUA Route section as :
> >>
> >> [sip:*] enum:mydomain.com \\First Preference :
> >>
> >> [sip:*@192.168.96.112] sip:sip.gafachi.com \\Second preference if
> first
> >> get failed:
> >>
> >> I dial a number that is not in DNS Opensbc try to resolve this 3 time
> then
> >> give response "no such name" and then
> >> create a Invite request As:
> >>
> >> SIP/SDP Request: INVITE enum:141...@e1...
> >>
> >>
> >> how can i achieve that in case of enumLookup fail it send request using
> my
> >> voip Provider.
> >>
> >> Yogesh
> >>
> >>
> >
> -------------------------------------------------------------------------
> > This SF.net email is sponsored by: Splunk Inc.
> > Still grepping through log files to find problems? Stop.
> > Now Search log events and configuration files using AJAX and a browser.
> > Download your FREE copy of Splunk now >> http://get.splunk.com/
> > _______________________________________________
> > opensipstack-devel mailing list
> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >
> >
> >
> >
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems? Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >> http://get.splunk.com/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
|
|
From: <jo...@op...> - 2007-11-13 02:17:26
|
Ugh! I guess you are right. Indeed enum lookup failure are not handled
correctly. Can you try the following block of code instead and let me
know if it fixes the problem?
if( routeURI.GetScheme() *= "enum" )
{
PStringArray enumServer;
enumServer.AppendString( routeURI.GetHost().c_str() );
if( !SIPTransport::ENUMLookup(
enumServer,
targetName,
routeURI ) )
{
continue; /// enum lookup failed. continue to the next
route
}
}
Yogesh Chaturvedi wrote:
> hi,
>
> Thanks for reply
>
> *Use a comma delimited list for your route instead of having separate
> route entries. The sample route below will allow you to route to enum,
> then to 192.168... and finally to gafachi.
>
> [sip:*] enum:**mydomain.com* <http://mydomain.com/>*,
> sip:**192.168.96.112*<http://192.168.96.112/>
> *, sip:** gafachi.com* <http://gafachi.com/>
>
> //////////////////
>
> I have tried to do and give setting in B2BUA route as :
>
> Route_1 : [sip:*] validnumber.com, sip:sip.gafachi.com
>
> but after getting Enum Lookup fail it create a URI like
>
> SIP/SDPRequest: INVITE
> enum:141...@va...<141...@va...>
> (according to wireshark)
>
> After looking in source code i found that :
> In *Route.cxx *class in *FindRoute* function it do not change the SIPURI
> structure in case of enumlookup fail and working as :
> Route.cxx Line 377:
>
> it call SIPTransport::ENUMLookup(..., routeURI) function and if enumlookup
> fail then it set the username as Called number and do not change the rest
> SIPURI structure and create SIP INVITE Request as given above:
>
> *We need to find a way to call to next available route if enumLookup fail
> (not call fail)*
>
> So as far as i think we need to do changes in source code in both case
> either we want to use comma seperator in same route input box(GUI) or we
> want to go on next route(next route box in GUI) if enumLookup fail.
>
> For time being i have done changes in this block and add a single line to go
> to next route box entry to find next route search if enumlookup fail as:
>
> Route.cxx line Line: 391
>
> if( routeURI.GetUser().IsEmpty() )
> {
>
> break; // To go on next route(route_2)
>
> }
>
> I am using a bool variable here to keep track if inner loop get fail due to
> enumLookup fail and go to next route of outer for loop.
>
> is it break the original functionality(please add your comments) and what is
> the Best solution for this problem.
>
>
> Thanks
> Yogesh
>
>
>
>
>
>
> On 11/6/07, Yogesh Chaturvedi <ych...@gm...> wrote:
>
>> Hi,
>>
>> I want to configure OpenSbc such as it first call using enumlookup and if
>> it fail then
>> it should call through my voip provider.
>>
>>
>> I have done setting in B2BUA Route section as :
>>
>> [sip:*] enum:mydomain.com \\First Preference :
>>
>> [sip:*@192.168.96.112] sip:sip.gafachi.com \\Second preference if first
>> get failed:
>>
>> I dial a number that is not in DNS Opensbc try to resolve this 3 time then
>> give response "no such name" and then
>> create a Invite request As:
>>
>> SIP/SDP Request: INVITE enum:141...@e1...
>>
>>
>> how can i achieve that in case of enumLookup fail it send request using my
>> voip Provider.
>>
>> Yogesh
>>
>>
> -------------------------------------------------------------------------
> 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: Piotr C. <c...@dg...> - 2007-11-12 15:44:01
|
Hello Joegen, my name is Piotr and J continuing works after Tom, would you like to drop me some words about API to make voice recording - is it ready jet? Best regards piotr |
|
From: Yogesh C. <ych...@gm...> - 2007-11-12 11:27:19
|
hi, Thanks for reply *Use a comma delimited list for your route instead of having separate route entries. The sample route below will allow you to route to enum, then to 192.168... and finally to gafachi. [sip:*] enum:**mydomain.com* <http://mydomain.com/>*, sip:**192.168.96.112*<http://192.168.96.112/> *, sip:** gafachi.com* <http://gafachi.com/> ////////////////// I have tried to do and give setting in B2BUA route as : Route_1 : [sip:*] validnumber.com, sip:sip.gafachi.com but after getting Enum Lookup fail it create a URI like SIP/SDPRequest: INVITE enum:141...@va...<141...@va...> (according to wireshark) After looking in source code i found that : In *Route.cxx *class in *FindRoute* function it do not change the SIPURI structure in case of enumlookup fail and working as : Route.cxx Line 377: it call SIPTransport::ENUMLookup(..., routeURI) function and if enumlookup fail then it set the username as Called number and do not change the rest SIPURI structure and create SIP INVITE Request as given above: *We need to find a way to call to next available route if enumLookup fail (not call fail)* So as far as i think we need to do changes in source code in both case either we want to use comma seperator in same route input box(GUI) or we want to go on next route(next route box in GUI) if enumLookup fail. For time being i have done changes in this block and add a single line to go to next route box entry to find next route search if enumlookup fail as: Route.cxx line Line: 391 if( routeURI.GetUser().IsEmpty() ) { break; // To go on next route(route_2) } I am using a bool variable here to keep track if inner loop get fail due to enumLookup fail and go to next route of outer for loop. is it break the original functionality(please add your comments) and what is the Best solution for this problem. Thanks Yogesh On 11/6/07, Yogesh Chaturvedi <ych...@gm...> wrote: > > Hi, > > I want to configure OpenSbc such as it first call using enumlookup and if > it fail then > it should call through my voip provider. > > > I have done setting in B2BUA Route section as : > > [sip:*] enum:mydomain.com \\First Preference : > > [sip:*@192.168.96.112] sip:sip.gafachi.com \\Second preference if first > get failed: > > I dial a number that is not in DNS Opensbc try to resolve this 3 time then > give response "no such name" and then > create a Invite request As: > > SIP/SDP Request: INVITE enum:141...@e1... > > > how can i achieve that in case of enumLookup fail it send request using my > voip Provider. > > Yogesh > |
|
From: <jo...@op...> - 2007-11-12 09:38:57
|
Gaurav Kheterpal wrote: > Hello Joegen, > > Thanks for your reply. I hope SIPIt 21 went well for you and Ilian. > > Ref: Issue regarding incoming call > > 1) The TO URI is an IP Address rather than a domain > 2) Port is 31265 > > Both these are specific to Gafachi's behavior though I believe 1) is > perfectly valid as per RFC 3261. > Yeah, no argument about that. It's a valid URI. However, if Gafachi will follow the 3261 specs for constructing requests destined to a registered UA, it would have used the To-URI sent in the REGISTER, and set the startline-uri to point to the Contact address. This is very important for SIP trunking since OpenSBC can support multiple trunks and the only way to distinguish which trunk is involved in a call is via the to-uri > Anyway, we resolved the issue by commenting out the following check in > opensbc/SBCSIPTrunkEndPoint.cxx -OnCreateB2BUA() > > //if( trunkReg == NULL ) > // return NULL; > > Lines 99-100 commented out. > > After making the above change, trunking works fine in case of incoming > calls. I'm not sure what impact this change will have on the rest of the > code. Comments? > > I guess this would be safe. It allows you to use basic B2BUA routing to handle the call. I'll take a deeper look when i get the chance. > Regarding the outgoing call issue, I'll try out your suggestion and let you > know if it works. > > Regards, > Gaurav > > >> -----Original Message----- >> From: ope...@li... >> [mailto:ope...@li...] On Behalf Of >> jo...@op... >> Sent: Monday, November 12, 2007 8:09 AM >> To: jo...@op... >> Cc: ope...@li...; joe...@gm...; >> din...@gm...; ope...@li... >> Subject: Re: [OpenSIPStack] OpenSBC SIP Trunking - Null Sessionin >> CallSessionManager.cxx >> >> Hi Gaurav, >> >> I just found out that you CC'ed my gmail account and the attachments >> made it. For your inbound call, attached is the INVITE >> >> INVITE sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0 >> From: "unknown" >> <sip:416...@si...>;tag=gss4a7f914ce003e05a02fa166fc6c27f14 >> To: <sip:16462781042@59.95.152.178:31265> >> Via: SIP/2.0/UDP 64.192.112.13:5060;branch=z9hG4bKd21aa1af >> CSeq: 102 INVITE >> Call-ID: d82a729522b3f145f0798b363d25b6a1@64.192.112.13 >> Contact: <sip:4169074204@64.192.112.13> >> Date: Tue, 06 Nov 2007 06:21:48 GMT >> User-Agent: Gafachi UAC v110.05 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE >> Content-Type: application/sdp >> Content-Length: 240 >> >> >> There are two things that are not in proper place in this INVITE from >> gafachi. First, The to-uri host is an ip address and not a domain. >> This call will not be properly identified by the SBC as a trunk call. >> Another strange thing is that it has a port (31265) which definitely is >> not an OSBC listener port. Can you give more information as to why >> gafachi will be sending this to-uri? >> >> For your outbound call, indeed the call was not properly identified as a >> trunk call.... see my previous response. >> >> >> >> jo...@op... wrote: >> >>> Hi Gaurav, >>> >>> I apologize for the late response. We just arrived from SIPIT 21. >>> My answers inline. >>> >>> >>> Gaurav Kheterpal wrote: >>> >>>> Hello Joegen, >>>> >>>> >>>> >>>> I grabbed the latest source code from CVS and configured OpenSBC for >>>> >> SIP >> >>>> trunking. While it may not be prime time, it works quiet well except >>>> for a >>>> couple of issues:- >>>> >>>> >>>> >>>> 1) Upon initialization, OpenSBC is able to register successfully with >>>> various service providers. I configured a couple of Xlite softphones to >>>> register with OpenSBC and used them for testing inbound/ outbound >>>> calls with >>>> various service providers. >>>> >>>> >>>> >>>> * Placing an outbound call from one of the Xlite softphones to an >>>> external service provider (Gafachi) works fine (attached log - >>>> outgoing.log) >>>> * Placing an inbound call from an external service provider to >>>> opensbc >>>> results in the following error in the log (attached log - incoming.log) >>>> >>>> >>>> >>>> >>> Many things could go wrong in a SIP trunk configuration. Routing rely >>> solely on the correct formating of the To URI. You need to let me >>> know about the specifics of your configuration like the domain of the >>> SIP provider and the kind of INVITE the provider is sending to reach >>> your trunk. BTW, your attachment did not make it. You can send it >>> to me directly and i'll see what I can figure out. An ethereal >>> capture would also be nice just in case we are investigating low level >>> interop issues aside from configuration issues. >>> >>> >>> >>>> 4:43:39.304 DTL: [CID=0x0af8] Event: ---> Inbound - INVITE >>>> sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0 >>>> >>>> 4:43:39.306 DBG: [CID=0x0af8] Session CREATED >>>> >>>> 4:43:39.306 INF: [CID=0x0af8] *** CREATED (UAS) CALL *** >>>> d82a729522b3f145f0798b363d25b6a1@64.192.112.13 >>>> >>>> 4:43:39.306 INF: [CID=0x0af8] *** DESTROYED CALL *** >>>> d82a729522b3f145f0798b363d25b6a1@64.192.112.13 >>>> >>>> 4:43:39.307 DBG: [CID=0x0af8] CALL: (inbound) : Session DESTROYED >>>> >>>> 4:43:39.308 ERR: [CID=0x0000] GC: .\src\CallSessionManager.cxx:528 >>>> CallSessionManager::OnCreateServerSession::CallSession Attempt to >>>> CreateReference() a NULL Pointer or none descendant of PObject!!! >>>> >>>> 4:43:39.308 DBG: [CID=0x06cb] *** MESSAGE ARRIVAL *** No Session >>>> available >>>> to handle INVITE sip:16462781042@192.168.96.115:5066;transport=udp >>>> SIP/2.0 >>>> >>>> 4:43:39.312 PWL: [CID=0x0000] Using Iface: 192.168.96.115 to send >>>> to Dest: >>>> 64.192.112.13 >>>> >>>> >>>> >>>> Can you confirm if it's a bug/ configuration issue? The log file is >>>> attached >>>> for reference >>>> >>>> >>>> >>>> 2) While exploring various service providers, I found an issue with >>>> authentication in SIP trunking mode. While placing an outbound call >>>> to an >>>> external service provider from one of the UAs registered to SBC, if the >>>> external service provider requests authentication and returns a 407, >>>> the 407 >>>> is relayed back by OpenSBC to the UA. This should not happen as all the >>>> credentials for service provider (trunk-account information) is >>>> present with >>>> the SBC itself. Any comments? >>>> >>>> >>>> >>>> >>> I guessing OpenSBC was not able to identify the call as a trunk call >>> properly. You are correct that the trunk should have handled the >>> authentication instead of relaying the 407. If you are using the Main >>> trunk to route your calls to the SIP Trunk, you may try to use >>> "sip-trunk" parameter in our b2bua route >>> >>> Example: [sip:1212*] sip:mytrunkprovider.com;sip-trunk=true >>> >>> >>> This would automatically tell the b2bua to route all calls bound to >>> New York to be routed to the SIP Trunk. >>> >>> >>> >>>> I look forward to hearing from you regarding these issues. Please let >>>> >> me >> >>>> know if you need any other information regarding the same. >>>> >>>> >>>> >>>> Regards, >>>> >>>> Gaurav >>>> >>>> >>>> ----------------------------------------------------------------------- >>>> >> - >> >>>> ----------------------------------------------------------------------- >>>> >> -- >> >>>> This SF.net email is sponsored by: Splunk Inc. >>>> Still grepping through log files to find problems? Stop. >>>> Now Search log events and configuration files using AJAX and a browser. >>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>> ----------------------------------------------------------------------- >>>> >> - >> >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> ----------------------------------------------------------------------- >>>> >> - >> >>>> No virus found in this incoming message. >>>> Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: >>>> 269.15.20/1107 - Release Date: 11/3/2007 11:22 AM >>>> >>>> >>> >> ------------------------------------------------------------------------- >> 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: Gaurav K. <gkh...@is...> - 2007-11-12 08:04:54
|
Hello Joegen, Thanks for your reply. I hope SIPIt 21 went well for you and Ilian. Ref: Issue regarding incoming call 1) The TO URI is an IP Address rather than a domain 2) Port is 31265 Both these are specific to Gafachi's behavior though I believe 1) is perfectly valid as per RFC 3261. Anyway, we resolved the issue by commenting out the following check in opensbc/SBCSIPTrunkEndPoint.cxx -OnCreateB2BUA() //if( trunkReg == NULL ) // return NULL; Lines 99-100 commented out. After making the above change, trunking works fine in case of incoming calls. I'm not sure what impact this change will have on the rest of the code. Comments? Regarding the outgoing call issue, I'll try out your suggestion and let you know if it works. Regards, Gaurav > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > jo...@op... > Sent: Monday, November 12, 2007 8:09 AM > To: jo...@op... > Cc: ope...@li...; joe...@gm...; > din...@gm...; ope...@li... > Subject: Re: [OpenSIPStack] OpenSBC SIP Trunking - Null Sessionin > CallSessionManager.cxx > > Hi Gaurav, > > I just found out that you CC'ed my gmail account and the attachments > made it. For your inbound call, attached is the INVITE > > INVITE sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0 > From: "unknown" > <sip:416...@si...>;tag=gss4a7f914ce003e05a02fa166fc6c27f14 > To: <sip:16462781042@59.95.152.178:31265> > Via: SIP/2.0/UDP 64.192.112.13:5060;branch=z9hG4bKd21aa1af > CSeq: 102 INVITE > Call-ID: d82a729522b3f145f0798b363d25b6a1@64.192.112.13 > Contact: <sip:4169074204@64.192.112.13> > Date: Tue, 06 Nov 2007 06:21:48 GMT > User-Agent: Gafachi UAC v110.05 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE > Content-Type: application/sdp > Content-Length: 240 > > > There are two things that are not in proper place in this INVITE from > gafachi. First, The to-uri host is an ip address and not a domain. > This call will not be properly identified by the SBC as a trunk call. > Another strange thing is that it has a port (31265) which definitely is > not an OSBC listener port. Can you give more information as to why > gafachi will be sending this to-uri? > > For your outbound call, indeed the call was not properly identified as a > trunk call.... see my previous response. > > > > jo...@op... wrote: > > Hi Gaurav, > > > > I apologize for the late response. We just arrived from SIPIT 21. > > My answers inline. > > > > > > Gaurav Kheterpal wrote: > >> Hello Joegen, > >> > >> > >> > >> I grabbed the latest source code from CVS and configured OpenSBC for > SIP > >> trunking. While it may not be prime time, it works quiet well except > >> for a > >> couple of issues:- > >> > >> > >> > >> 1) Upon initialization, OpenSBC is able to register successfully with > >> various service providers. I configured a couple of Xlite softphones to > >> register with OpenSBC and used them for testing inbound/ outbound > >> calls with > >> various service providers. > >> > >> > >> > >> * Placing an outbound call from one of the Xlite softphones to an > >> external service provider (Gafachi) works fine (attached log - > >> outgoing.log) > >> * Placing an inbound call from an external service provider to > >> opensbc > >> results in the following error in the log (attached log - incoming.log) > >> > >> > >> > > > > > > Many things could go wrong in a SIP trunk configuration. Routing rely > > solely on the correct formating of the To URI. You need to let me > > know about the specifics of your configuration like the domain of the > > SIP provider and the kind of INVITE the provider is sending to reach > > your trunk. BTW, your attachment did not make it. You can send it > > to me directly and i'll see what I can figure out. An ethereal > > capture would also be nice just in case we are investigating low level > > interop issues aside from configuration issues. > > > > > >> 4:43:39.304 DTL: [CID=0x0af8] Event: ---> Inbound - INVITE > >> sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0 > >> > >> 4:43:39.306 DBG: [CID=0x0af8] Session CREATED > >> > >> 4:43:39.306 INF: [CID=0x0af8] *** CREATED (UAS) CALL *** > >> d82a729522b3f145f0798b363d25b6a1@64.192.112.13 > >> > >> 4:43:39.306 INF: [CID=0x0af8] *** DESTROYED CALL *** > >> d82a729522b3f145f0798b363d25b6a1@64.192.112.13 > >> > >> 4:43:39.307 DBG: [CID=0x0af8] CALL: (inbound) : Session DESTROYED > >> > >> 4:43:39.308 ERR: [CID=0x0000] GC: .\src\CallSessionManager.cxx:528 > >> CallSessionManager::OnCreateServerSession::CallSession Attempt to > >> CreateReference() a NULL Pointer or none descendant of PObject!!! > >> > >> 4:43:39.308 DBG: [CID=0x06cb] *** MESSAGE ARRIVAL *** No Session > >> available > >> to handle INVITE sip:16462781042@192.168.96.115:5066;transport=udp > >> SIP/2.0 > >> > >> 4:43:39.312 PWL: [CID=0x0000] Using Iface: 192.168.96.115 to send > >> to Dest: > >> 64.192.112.13 > >> > >> > >> > >> Can you confirm if it's a bug/ configuration issue? The log file is > >> attached > >> for reference > >> > >> > >> > >> 2) While exploring various service providers, I found an issue with > >> authentication in SIP trunking mode. While placing an outbound call > >> to an > >> external service provider from one of the UAs registered to SBC, if the > >> external service provider requests authentication and returns a 407, > >> the 407 > >> is relayed back by OpenSBC to the UA. This should not happen as all the > >> credentials for service provider (trunk-account information) is > >> present with > >> the SBC itself. Any comments? > >> > >> > >> > > > > I guessing OpenSBC was not able to identify the call as a trunk call > > properly. You are correct that the trunk should have handled the > > authentication instead of relaying the 407. If you are using the Main > > trunk to route your calls to the SIP Trunk, you may try to use > > "sip-trunk" parameter in our b2bua route > > > > Example: [sip:1212*] sip:mytrunkprovider.com;sip-trunk=true > > > > > > This would automatically tell the b2bua to route all calls bound to > > New York to be routed to the SIP Trunk. > > > > > >> I look forward to hearing from you regarding these issues. Please let > me > >> know if you need any other information regarding the same. > >> > >> > >> > >> Regards, > >> > >> Gaurav > >> > >> > >> ----------------------------------------------------------------------- > - > >> > >> ----------------------------------------------------------------------- > -- > >> > >> This SF.net email is sponsored by: Splunk Inc. > >> Still grepping through log files to find problems? Stop. > >> Now Search log events and configuration files using AJAX and a browser. > >> Download your FREE copy of Splunk now >> http://get.splunk.com/ > >> ----------------------------------------------------------------------- > - > >> > >> _______________________________________________ > >> opensipstack-devel mailing list > >> ope...@li... > >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > >> > >> ----------------------------------------------------------------------- > - > >> > >> No virus found in this incoming message. > >> Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: > >> 269.15.20/1107 - Release Date: 11/3/2007 11:22 AM > >> > > > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
|
From: <jo...@op...> - 2007-11-12 02:39:30
|
Hi Gaurav, I just found out that you CC'ed my gmail account and the attachments made it. For your inbound call, attached is the INVITE INVITE sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0 From: "unknown" <sip:416...@si...>;tag=gss4a7f914ce003e05a02fa166fc6c27f14 To: <sip:16462781042@59.95.152.178:31265> Via: SIP/2.0/UDP 64.192.112.13:5060;branch=z9hG4bKd21aa1af CSeq: 102 INVITE Call-ID: d82a729522b3f145f0798b363d25b6a1@64.192.112.13 Contact: <sip:4169074204@64.192.112.13> Date: Tue, 06 Nov 2007 06:21:48 GMT User-Agent: Gafachi UAC v110.05 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: 240 There are two things that are not in proper place in this INVITE from gafachi. First, The to-uri host is an ip address and not a domain. This call will not be properly identified by the SBC as a trunk call. Another strange thing is that it has a port (31265) which definitely is not an OSBC listener port. Can you give more information as to why gafachi will be sending this to-uri? For your outbound call, indeed the call was not properly identified as a trunk call.... see my previous response. jo...@op... wrote: > Hi Gaurav, > > I apologize for the late response. We just arrived from SIPIT 21. > My answers inline. > > > Gaurav Kheterpal wrote: >> Hello Joegen, >> >> >> >> I grabbed the latest source code from CVS and configured OpenSBC for SIP >> trunking. While it may not be prime time, it works quiet well except >> for a >> couple of issues:- >> >> >> >> 1) Upon initialization, OpenSBC is able to register successfully with >> various service providers. I configured a couple of Xlite softphones to >> register with OpenSBC and used them for testing inbound/ outbound >> calls with >> various service providers. >> >> >> >> * Placing an outbound call from one of the Xlite softphones to an >> external service provider (Gafachi) works fine (attached log - >> outgoing.log) >> * Placing an inbound call from an external service provider to >> opensbc >> results in the following error in the log (attached log - incoming.log) >> >> >> > > > Many things could go wrong in a SIP trunk configuration. Routing rely > solely on the correct formating of the To URI. You need to let me > know about the specifics of your configuration like the domain of the > SIP provider and the kind of INVITE the provider is sending to reach > your trunk. BTW, your attachment did not make it. You can send it > to me directly and i'll see what I can figure out. An ethereal > capture would also be nice just in case we are investigating low level > interop issues aside from configuration issues. > > >> 4:43:39.304 DTL: [CID=0x0af8] Event: ---> Inbound - INVITE >> sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0 >> >> 4:43:39.306 DBG: [CID=0x0af8] Session CREATED >> >> 4:43:39.306 INF: [CID=0x0af8] *** CREATED (UAS) CALL *** >> d82a729522b3f145f0798b363d25b6a1@64.192.112.13 >> >> 4:43:39.306 INF: [CID=0x0af8] *** DESTROYED CALL *** >> d82a729522b3f145f0798b363d25b6a1@64.192.112.13 >> >> 4:43:39.307 DBG: [CID=0x0af8] CALL: (inbound) : Session DESTROYED >> >> 4:43:39.308 ERR: [CID=0x0000] GC: .\src\CallSessionManager.cxx:528 >> CallSessionManager::OnCreateServerSession::CallSession Attempt to >> CreateReference() a NULL Pointer or none descendant of PObject!!! >> >> 4:43:39.308 DBG: [CID=0x06cb] *** MESSAGE ARRIVAL *** No Session >> available >> to handle INVITE sip:16462781042@192.168.96.115:5066;transport=udp >> SIP/2.0 >> >> 4:43:39.312 PWL: [CID=0x0000] Using Iface: 192.168.96.115 to send >> to Dest: >> 64.192.112.13 >> >> >> >> Can you confirm if it's a bug/ configuration issue? The log file is >> attached >> for reference >> >> >> >> 2) While exploring various service providers, I found an issue with >> authentication in SIP trunking mode. While placing an outbound call >> to an >> external service provider from one of the UAs registered to SBC, if the >> external service provider requests authentication and returns a 407, >> the 407 >> is relayed back by OpenSBC to the UA. This should not happen as all the >> credentials for service provider (trunk-account information) is >> present with >> the SBC itself. Any comments? >> >> >> > > I guessing OpenSBC was not able to identify the call as a trunk call > properly. You are correct that the trunk should have handled the > authentication instead of relaying the 407. If you are using the Main > trunk to route your calls to the SIP Trunk, you may try to use > "sip-trunk" parameter in our b2bua route > > Example: [sip:1212*] sip:mytrunkprovider.com;sip-trunk=true > > > This would automatically tell the b2bua to route all calls bound to > New York to be routed to the SIP Trunk. > > >> I look forward to hearing from you regarding these issues. Please let me >> know if you need any other information regarding the same. >> >> >> >> Regards, >> >> Gaurav >> >> >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> ------------------------------------------------------------------------ >> >> No virus found in this incoming message. >> Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: >> 269.15.20/1107 - Release Date: 11/3/2007 11:22 AM >> > > |
|
From: <jo...@op...> - 2007-11-12 02:29:51
|
Hi Gaurav, I apologize for the late response. We just arrived from SIPIT 21. My answers inline. Gaurav Kheterpal wrote: > Hello Joegen, > > > > I grabbed the latest source code from CVS and configured OpenSBC for SIP > trunking. While it may not be prime time, it works quiet well except for a > couple of issues:- > > > > 1) Upon initialization, OpenSBC is able to register successfully with > various service providers. I configured a couple of Xlite softphones to > register with OpenSBC and used them for testing inbound/ outbound calls with > various service providers. > > > > * Placing an outbound call from one of the Xlite softphones to an > external service provider (Gafachi) works fine (attached log - outgoing.log) > * Placing an inbound call from an external service provider to opensbc > results in the following error in the log (attached log - incoming.log) > > > Many things could go wrong in a SIP trunk configuration. Routing rely solely on the correct formating of the To URI. You need to let me know about the specifics of your configuration like the domain of the SIP provider and the kind of INVITE the provider is sending to reach your trunk. BTW, your attachment did not make it. You can send it to me directly and i'll see what I can figure out. An ethereal capture would also be nice just in case we are investigating low level interop issues aside from configuration issues. > 4:43:39.304 DTL: [CID=0x0af8] Event: ---> Inbound - INVITE > sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0 > > 4:43:39.306 DBG: [CID=0x0af8] Session CREATED > > 4:43:39.306 INF: [CID=0x0af8] *** CREATED (UAS) CALL *** > d82a729522b3f145f0798b363d25b6a1@64.192.112.13 > > 4:43:39.306 INF: [CID=0x0af8] *** DESTROYED CALL *** > d82a729522b3f145f0798b363d25b6a1@64.192.112.13 > > 4:43:39.307 DBG: [CID=0x0af8] CALL: (inbound) : Session DESTROYED > > 4:43:39.308 ERR: [CID=0x0000] GC: .\src\CallSessionManager.cxx:528 > CallSessionManager::OnCreateServerSession::CallSession Attempt to > CreateReference() a NULL Pointer or none descendant of PObject!!! > > 4:43:39.308 DBG: [CID=0x06cb] *** MESSAGE ARRIVAL *** No Session available > to handle INVITE sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0 > > 4:43:39.312 PWL: [CID=0x0000] Using Iface: 192.168.96.115 to send to Dest: > 64.192.112.13 > > > > Can you confirm if it's a bug/ configuration issue? The log file is attached > for reference > > > > 2) While exploring various service providers, I found an issue with > authentication in SIP trunking mode. While placing an outbound call to an > external service provider from one of the UAs registered to SBC, if the > external service provider requests authentication and returns a 407, the 407 > is relayed back by OpenSBC to the UA. This should not happen as all the > credentials for service provider (trunk-account information) is present with > the SBC itself. Any comments? > > > I guessing OpenSBC was not able to identify the call as a trunk call properly. You are correct that the trunk should have handled the authentication instead of relaying the 407. If you are using the Main trunk to route your calls to the SIP Trunk, you may try to use "sip-trunk" parameter in our b2bua route Example: [sip:1212*] sip:mytrunkprovider.com;sip-trunk=true This would automatically tell the b2bua to route all calls bound to New York to be routed to the SIP Trunk. > I look forward to hearing from you regarding these issues. Please let me > know if you need any other information regarding the same. > > > > Regards, > > Gaurav > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.503 / Virus Database: 269.15.20/1107 - Release Date: 11/3/2007 11:22 AM > |
|
From: Joegen E. B. <joe...@gm...> - 2007-11-07 05:11:06
|
Hi Everyone, Ilian and myself are now attending SIPIT 21. I apologize for not responding to recent emails. I promise to attend to them as soon as I get back to the office. Joegen |
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From: Yogesh C. <ych...@gm...> - 2007-11-06 12:21:01
|
Hi, I want to configure OpenSbc such as it first call using enumlookup and if it fail then it should call through my voip provider. I have done setting in B2BUA Route section as : [sip:*] enum:mydomain.com \\First <file://First/> Preference : [sip:*@192.168.96.112] sip:sip.gafachi.com \\Second <file://Second/>preference if first get failed: I dial a number that is not in DNS Opensbc try to resolve this 3 time then give response "no such name" and then create a Invite request As: SIP/SDP Request: INVITE enum:141...@e1... how can i achieve that in case of enumLookup fail it send request using my voip Provider. Yogesh |
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From: Gaurav K. <gkh...@is...> - 2007-11-06 08:18:52
|
Hello Joegen, I grabbed the latest source code from CVS and configured OpenSBC for SIP trunking. While it may not be prime time, it works quiet well except for a couple of issues:- 1) Upon initialization, OpenSBC is able to register successfully with various service providers. I configured a couple of Xlite softphones to register with OpenSBC and used them for testing inbound/ outbound calls with various service providers. * Placing an outbound call from one of the Xlite softphones to an external service provider (Gafachi) works fine (attached log - outgoing.log) * Placing an inbound call from an external service provider to opensbc results in the following error in the log (attached log - incoming.log) 4:43:39.304 DTL: [CID=0x0af8] Event: ---> Inbound - INVITE sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0 4:43:39.306 DBG: [CID=0x0af8] Session CREATED 4:43:39.306 INF: [CID=0x0af8] *** CREATED (UAS) CALL *** d82a729522b3f145f0798b363d25b6a1@64.192.112.13 4:43:39.306 INF: [CID=0x0af8] *** DESTROYED CALL *** d82a729522b3f145f0798b363d25b6a1@64.192.112.13 4:43:39.307 DBG: [CID=0x0af8] CALL: (inbound) : Session DESTROYED 4:43:39.308 ERR: [CID=0x0000] GC: .\src\CallSessionManager.cxx:528 CallSessionManager::OnCreateServerSession::CallSession Attempt to CreateReference() a NULL Pointer or none descendant of PObject!!! 4:43:39.308 DBG: [CID=0x06cb] *** MESSAGE ARRIVAL *** No Session available to handle INVITE sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0 4:43:39.312 PWL: [CID=0x0000] Using Iface: 192.168.96.115 to send to Dest: 64.192.112.13 Can you confirm if it's a bug/ configuration issue? The log file is attached for reference 2) While exploring various service providers, I found an issue with authentication in SIP trunking mode. While placing an outbound call to an external service provider from one of the UAs registered to SBC, if the external service provider requests authentication and returns a 407, the 407 is relayed back by OpenSBC to the UA. This should not happen as all the credentials for service provider (trunk-account information) is present with the SBC itself. Any comments? I look forward to hearing from you regarding these issues. Please let me know if you need any other information regarding the same. Regards, Gaurav |
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From: voice <vo...@ne...> - 2007-11-05 20:24:43
|
Hi Joe I finally have a change to contribute to this effort In your instructions for a Linux install please insert before number 1 1. Make sure to Open up port 2401 on your firewall etc...! r ----- Original Message ----- From: "Joegen E. Baclor" <joe...@gm...> To: <ope...@li...> Sent: Sunday, November 04, 2007 9:27 PM Subject: Re: [OpenSIPStack] The CSeq headers for the MESSAGE requests > Hi GCC, > > I don't have the chance to look at the code but this seem highly > probable. MESSAGE are sent out of the dialog. Thus there is no need > to maintain CSeq state. Transactions are identifiable via their unique > from tags. However, I think it is a good idea to increment the CSeq for > out of the dialog requests as well. I'll have this changed when i get > the chance. Thanks for pointing it out. > > Joegen > > > Woo Chen wrote: > > I've found that every time I send the instant messages, the CSeq headers for the MESSAGE requests are always the same. I use OpenSIPStack-1.1.7-11, but I think the latest CVS should have the same behavior. > > > > Is it normal? > > > > I attached the .pcap file. The scenario is that I continually sent myself the message. > > > > Regards, > > GCC > > > > ------------------------------------------------------------------------ > > > > ------------------------------------------------------------------------- > > This SF.net email is sponsored by: Splunk Inc. > > Still grepping through log files to find problems? Stop. > > Now Search log events and configuration files using AJAX and a browser. > > Download your FREE copy of Splunk now >> http://get.splunk.com/ > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > opensipstack-devel mailing list > > ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------------------------------------------------ > > > > No virus found in this incoming message. > > Checked by AVG Free Edition. > > Version: 7.5.503 / Virus Database: 269.15.20/1107 - Release Date: 11/3/2007 11:22 AM > > > > > ------------------------------------------------------------------------- > 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 |
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From: voice <vo...@ne...> - 2007-11-05 16:22:42
|
Hi What is the openSipStack Russel release on sourceforge.net? And is it up to date with the latest patches? r |
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From: Joegen E. B. <joe...@gm...> - 2007-11-05 04:27:46
|
Hi GCC, I don't have the chance to look at the code but this seem highly probable. MESSAGE are sent out of the dialog. Thus there is no need to maintain CSeq state. Transactions are identifiable via their unique from tags. However, I think it is a good idea to increment the CSeq for out of the dialog requests as well. I'll have this changed when i get the chance. Thanks for pointing it out. Joegen Woo Chen wrote: > I've found that every time I send the instant messages, the CSeq headers for the MESSAGE requests are always the same. I use OpenSIPStack-1.1.7-11, but I think the latest CVS should have the same behavior. > > Is it normal? > > I attached the .pcap file. The scenario is that I continually sent myself the message. > > Regards, > GCC > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.503 / Virus Database: 269.15.20/1107 - Release Date: 11/3/2007 11:22 AM > |
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From: Woo C. <vir...@in...> - 2007-11-05 03:53:44
|
I've found that every time I send the instant messages, the CSeq headers = for the MESSAGE requests are always the same. I use OpenSIPStack-1.1.7-11, = but I think the latest CVS should have the same behavior. Is it normal? I attached the .pcap file. The scenario is that I continually sent = myself the message. Regards, GCC |
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From: voice <vo...@ne...> - 2007-11-03 16:41:49
|
The message coming back from CVS cvs login: autherization failed: server opensipstack.cvs.sourceforge.net rejected access to /cvsroot/opensipstack for user anonymous I have change /cvsroot/ to /tmp/ /cvsroot to $CVSROOT/ am i correct that /cvsroot/opensipstack is pointing to my /tmp/opensipstack on my server? Could it be a premission issue? I am doing this as root user. r ----- Original Message ----- From: "jo...@op..." <joe...@gm...> To: <ope...@li...> Sent: Saturday, November 03, 2007 8:39 AM Subject: Re: [OpenSIPStack] Dialog events I tried to make some sense out of "aonocvs" and decided that since it does not make any sense, to read your question based on the nearest context. I presupposed that you might have meant a shorthand of "anonymous cvs" and I said yes it supports anonymous CVS. What exactly don't you understand in the following instructions? ----- OpenSIPStack CVS is hosted at sourceforge. It can be accessed via anonymous login. For unix users, the following command line should download the latest CVS Head from sourceforge. unix-shell>#cvs -d:pserver:ano...@op...: /cvsroot/opensipstack login unix-shell>#cvs -z3 -d:pserver:ano...@op...: /cvsroot/opensipstack co -P opensipstack You may also get cvs-get <http://www.opensipstack.org/releases/tools/cvs-get.tar.gz> from here. cvs-get is a PERL script that automates downloads of OpenSIPStack modules. Simply extract cvs-get.tar.gz and execute: unix-shell>#cvs-get grab opensipstack unix-shell>#cvs-get grab opensbc unix-shell>#cvs-get grab atlsip ------ More instructions here http://sourceforge.net/cvs/?group_id=156710 voice wrote: > I was hoping you would have said something else. > > aonocvs does not work as password. > > It said cvs login authentiction failed... > > where do i set authentication to pserver? > > I have set invironments for .bashrc and .cshrc to point to /tmp for CVSROOT > > I am doing all this as root user. > > r > > ----- Original Message ----- > From: "jo...@op..." <joe...@gm...> > To: <ope...@li...> > Sent: Saturday, November 03, 2007 12:24 AM > Subject: Re: [OpenSIPStack] Dialog events > > > Yes, Anonymous access is supported. Make sure your authentication is set > to pserver: > > voice wrote: > >> Hi Joe >> >> CVS asks' for a password? is it aonocvs? >> >> r >> >> ----- Original Message ----- >> From: "jo...@op..." <joe...@gm...> >> To: <ope...@li...> >> Sent: Thursday, November 01, 2007 9:45 PM >> Subject: Re: [OpenSIPStack] Dialog events >> >> >> voice wrote: >> >> >>> Hi Joe >>> >>> The CVS is a version control website. I have installed openSBC from >>> >>> >> source >> >> >>> and compiled it aok. Is this CVS going to update that install? >>> >>> >> CVS will only get the source files. You will need to recompile it after >> the download and change your installed binary with the newly created one >> >> >> >> >>> Does it >>> over write existing source or does it update complied libraries etc.... >>> >>> >>> >> It will update existing source not the binaries. >> >> >> >>> Does it update contributions from others on this list and is it tested >>> before it is added to the CVS? >>> >>> >> You are assured that all codes commited to CVS are tested code. >> >> >> >> >>> And finally is it selective in what gets >>> updated? >>> >>> >>> >> Yes >> >> >> >>> r >>> >>> >>> ----- Original Message ----- >>> From: "Joegen E. Baclor" <joe...@gm...> >>> To: <ope...@li...> >>> Cc: <jo...@op...> >>> Sent: Wednesday, October 31, 2007 10:32 PM >>> Subject: Re: [OpenSIPStack] Dialog events >>> >>> >>> Instructions how to download from CVS can be found at >>> http://www.opensipstack.org/cvs.html >>> >>> voice wrote: >>> >>> >>> >>>> Hi Joe >>>> >>>> Could you please also send me that elaboration explaining steps to >>>> >>>> >>>> >>> download >>> >>> >>> >>>> latest versions of the codes >>>> using CVS >>>> >>>> Robert. >>>> >>>> >>>> ----- Original Message ----- >>>> From: "jo...@op..." <joe...@gm...> >>>> To: <ope...@li...> >>>> Sent: Wednesday, October 17, 2007 5:54 PM >>>> Subject: Re: [OpenSIPStack] Dialog events >>>> >>>> >>>> Of course. See how RFC3265Agent and RFC3680Package is used in >>>> RegisterSessionManager. >>>> >>>> >>>> sebastian pastor wrote: >>>> >>>> >>>> >>>> >>>>> Thanks, Joegen. >>>>> >>>>> >>>>> >>>>> Anyway, I'd like to know a bit more… >>>>> >>>>> My aim is, as developer, add a *new service* to the phone (for >>>>> > acceeding > >>>>> >>>>> >>>> to >>>> >>>> >>>> >>>> >>>>> media contents) and this will be based on exchange of SUBs and NTFYs >>>>> >>>>> >> with >> >> >>>>> event=dialog but with a diferent xml body than the standard. >>>>> >>>>> >>>>> >>>>> So, do you think it would be feasible or *realistic* trying to program >>>>> >>>>> >>>>> >>>>> >>>> this >>>> >>>> >>>> >>>> >>>>> whole behaviour based on existent OpenSipStack code ? It would be an >>>>> >>>>> >>>>> >>> state >>> >>> >>> >>>>> machine regarding to SUBSCRIBEs, renews, NTFYs, response codes… that I >>>>> >>>>> >>>>> >>>>> >>>> could >>>> >>>> >>>> >>>> >>>>> catch from the REGISTER one. >>>>> >>>>> Or it's better waiting for you to have this functionality implemented? >>>>> >>>>> >>>>> >>>>> How much *time* do you estimate that we are talking about? >>>>> >>>>> >>>>> >>>>> Best Regards, >>>>> >>>>> >>>>> >>>>> -Sebastián Pastor- >>>>> >>>>> Junior Telecomunications Engineer (Malaga, >>>>> >>>>> >> Spain) >> >> >>>>> p.s BTW, *Lalith *you are a bit off-topic. But i attached a quick doc >>>>> >>>>> >>>>> >>> that >>> >>> >>> >>>>> i've elaborated explaining steps to download latest versions of the >>>>> >>>>> >> codes >> >> >>>>> using CVS. Hope to help you. >>>>> >>>>> >>>>> 2007/10/17, Joegen E. Baclor <joe...@gm...>: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> Hi Sebastian, >>>>>> >>>>>> I am sorry but this feature has not made it yet to the mile stone >>>>>> release because SIP Trunking was deemed a more popular need of users >>>>>> > of > >>>>>> OpenSBC. >>>>>> >>>>>> >>>>>> >>>>>> --------------------------------------------------------------------- - >>>>>> > - > >> - >> >> >>>>>> --------------------------------------------------------------------- - >>>>>> > - > >> - >> >> >>> - >>> >>> >>> >>>>>> This SF.net email is sponsored by: Splunk Inc. >>>>>> Still grepping through log files to find problems? Stop. >>>>>> Now Search log events and configuration files using AJAX and a >>>>>> > browser. > >>>>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>>>> --------------------------------------------------------------------- - >>>>>> > - > >> - >> >> >>>>>> _______________________________________________ >>>>>> opensipstack-devel mailing list >>>>>> ope...@li... >>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>>> >>>>>> --------------------------------------------------------------------- - >>>>>> > - > >> - >> >> >>>>>> No virus found in this incoming message. >>>>>> Checked by AVG Free Edition. >>>>>> Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: >>>>>> >>>>>> >>>>>> >>>>>> >>>> 10/16/2007 2:14 PM >>>> >>>> >>>> >>>> >>>> ----------------------------------------------------------------------- - >>>> > - > >>>> This SF.net email is sponsored by: Splunk Inc. >>>> Still grepping through log files to find problems? Stop. >>>> Now Search log events and configuration files using AJAX and a browser. >>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> >>>> ----------------------------------------------------------------------- - >>>> > - > >>>> This SF.net email is sponsored by: Splunk Inc. >>>> Still grepping through log files to find problems? Stop. >>>> Now Search log events and configuration files using AJAX and a browser. >>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> >>>> >>>> >>>> >>> ------------------------------------------------------------------------ - >>> This SF.net email is sponsored by: Splunk Inc. >>> Still grepping through log files to find problems? Stop. >>> Now Search log events and configuration files using AJAX and a browser. >>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >>> ------------------------------------------------------------------------ - >>> This SF.net email is sponsored by: Splunk Inc. >>> Still grepping through log files to find problems? Stop. >>> Now Search log events and configuration files using AJAX and a browser. >>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >>> >>> >>> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> >> > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
|
From: voice <vo...@ne...> - 2007-11-03 16:29:01
|
Look I can follow directions just like anyone else. I toll you what the error said. I'm using Linux Centos 5 could that be a problem? I have gone to sourceforge and read all their CVS instructions i got that password from reading their examples. When i run the code below IT REQUEST a password to be entered. Why do you think i asked for the password. I did not think that anonymous uses password maybe except the anonymous email address as username or password. I set the Linux invironmentals according to sourceforge CVS examples. This is what i complained about before. Some where along the way you think that people are just to stupid. I'm not your 3 years that you have to repeat the instructions on the sit because i'm to young to read or follow directions. I followed the instruction and there is a problem. I'm ask you to assist me in determining what when wrong. Why am i being asked to provide a password? r ----- Original Message ----- From: "jo...@op..." <joe...@gm...> To: <ope...@li...> Sent: Saturday, November 03, 2007 8:39 AM Subject: Re: [OpenSIPStack] Dialog events I tried to make some sense out of "aonocvs" and decided that since it does not make any sense, to read your question based on the nearest context. I presupposed that you might have meant a shorthand of "anonymous cvs" and I said yes it supports anonymous CVS. What exactly don't you understand in the following instructions? ----- OpenSIPStack CVS is hosted at sourceforge. It can be accessed via anonymous login. For unix users, the following command line should download the latest CVS Head from sourceforge. unix-shell>#cvs -d:pserver:ano...@op...: /cvsroot/opensipstack login unix-shell>#cvs -z3 -d:pserver:ano...@op...: /cvsroot/opensipstack co -P opensipstack You may also get cvs-get <http://www.opensipstack.org/releases/tools/cvs-get.tar.gz> from here. cvs-get is a PERL script that automates downloads of OpenSIPStack modules. Simply extract cvs-get.tar.gz and execute: unix-shell>#cvs-get grab opensipstack unix-shell>#cvs-get grab opensbc unix-shell>#cvs-get grab atlsip ------ More instructions here http://sourceforge.net/cvs/?group_id=156710 voice wrote: > I was hoping you would have said something else. > > aonocvs does not work as password. > > It said cvs login authentiction failed... > > where do i set authentication to pserver? > > I have set invironments for .bashrc and .cshrc to point to /tmp for CVSROOT > > I am doing all this as root user. > > r > > ----- Original Message ----- > From: "jo...@op..." <joe...@gm...> > To: <ope...@li...> > Sent: Saturday, November 03, 2007 12:24 AM > Subject: Re: [OpenSIPStack] Dialog events > > > Yes, Anonymous access is supported. Make sure your authentication is set > to pserver: > > voice wrote: > >> Hi Joe >> >> CVS asks' for a password? is it aonocvs? >> >> r >> >> ----- Original Message ----- >> From: "jo...@op..." <joe...@gm...> >> To: <ope...@li...> >> Sent: Thursday, November 01, 2007 9:45 PM >> Subject: Re: [OpenSIPStack] Dialog events >> >> >> voice wrote: >> >> >>> Hi Joe >>> >>> The CVS is a version control website. I have installed openSBC from >>> >>> >> source >> >> >>> and compiled it aok. Is this CVS going to update that install? >>> >>> >> CVS will only get the source files. You will need to recompile it after >> the download and change your installed binary with the newly created one >> >> >> >> >>> Does it >>> over write existing source or does it update complied libraries etc.... >>> >>> >>> >> It will update existing source not the binaries. >> >> >> >>> Does it update contributions from others on this list and is it tested >>> before it is added to the CVS? >>> >>> >> You are assured that all codes commited to CVS are tested code. >> >> >> >> >>> And finally is it selective in what gets >>> updated? >>> >>> >>> >> Yes >> >> >> >>> r >>> >>> >>> ----- Original Message ----- >>> From: "Joegen E. Baclor" <joe...@gm...> >>> To: <ope...@li...> >>> Cc: <jo...@op...> >>> Sent: Wednesday, October 31, 2007 10:32 PM >>> Subject: Re: [OpenSIPStack] Dialog events >>> >>> >>> Instructions how to download from CVS can be found at >>> http://www.opensipstack.org/cvs.html >>> >>> voice wrote: >>> >>> >>> >>>> Hi Joe >>>> >>>> Could you please also send me that elaboration explaining steps to >>>> >>>> >>>> >>> download >>> >>> >>> >>>> latest versions of the codes >>>> using CVS >>>> >>>> Robert. >>>> >>>> >>>> ----- Original Message ----- >>>> From: "jo...@op..." <joe...@gm...> >>>> To: <ope...@li...> >>>> Sent: Wednesday, October 17, 2007 5:54 PM >>>> Subject: Re: [OpenSIPStack] Dialog events >>>> >>>> >>>> Of course. See how RFC3265Agent and RFC3680Package is used in >>>> RegisterSessionManager. >>>> >>>> >>>> sebastian pastor wrote: >>>> >>>> >>>> >>>> >>>>> Thanks, Joegen. >>>>> >>>>> >>>>> >>>>> Anyway, I'd like to know a bit more… >>>>> >>>>> My aim is, as developer, add a *new service* to the phone (for >>>>> > acceeding > >>>>> >>>>> >>>> to >>>> >>>> >>>> >>>> >>>>> media contents) and this will be based on exchange of SUBs and NTFYs >>>>> >>>>> >> with >> >> >>>>> event=dialog but with a diferent xml body than the standard. >>>>> >>>>> >>>>> >>>>> So, do you think it would be feasible or *realistic* trying to program >>>>> >>>>> >>>>> >>>>> >>>> this >>>> >>>> >>>> >>>> >>>>> whole behaviour based on existent OpenSipStack code ? It would be an >>>>> >>>>> >>>>> >>> state >>> >>> >>> >>>>> machine regarding to SUBSCRIBEs, renews, NTFYs, response codes… that I >>>>> >>>>> >>>>> >>>>> >>>> could >>>> >>>> >>>> >>>> >>>>> catch from the REGISTER one. >>>>> >>>>> Or it's better waiting for you to have this functionality implemented? >>>>> >>>>> >>>>> >>>>> How much *time* do you estimate that we are talking about? >>>>> >>>>> >>>>> >>>>> Best Regards, >>>>> >>>>> >>>>> >>>>> -Sebastián Pastor- >>>>> >>>>> Junior Telecomunications Engineer (Malaga, >>>>> >>>>> >> Spain) >> >> >>>>> p.s BTW, *Lalith *you are a bit off-topic. But i attached a quick doc >>>>> >>>>> >>>>> >>> that >>> >>> >>> >>>>> i've elaborated explaining steps to download latest versions of the >>>>> >>>>> >> codes >> >> >>>>> using CVS. Hope to help you. >>>>> >>>>> >>>>> 2007/10/17, Joegen E. Baclor <joe...@gm...>: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> Hi Sebastian, >>>>>> >>>>>> I am sorry but this feature has not made it yet to the mile stone >>>>>> release because SIP Trunking was deemed a more popular need of users >>>>>> > of > >>>>>> OpenSBC. >>>>>> >>>>>> >>>>>> >>>>>> --------------------------------------------------------------------- - >>>>>> > - > >> - >> >> >>>>>> --------------------------------------------------------------------- - >>>>>> > - > >> - >> >> >>> - >>> >>> >>> >>>>>> This SF.net email is sponsored by: Splunk Inc. >>>>>> Still grepping through log files to find problems? Stop. >>>>>> Now Search log events and configuration files using AJAX and a >>>>>> > browser. > >>>>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>>>> --------------------------------------------------------------------- - >>>>>> > - > >> - >> >> >>>>>> _______________________________________________ >>>>>> opensipstack-devel mailing list >>>>>> ope...@li... >>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>>> >>>>>> --------------------------------------------------------------------- - >>>>>> > - > >> - >> >> >>>>>> No virus found in this incoming message. >>>>>> Checked by AVG Free Edition. >>>>>> Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: >>>>>> >>>>>> >>>>>> >>>>>> >>>> 10/16/2007 2:14 PM >>>> >>>> >>>> >>>> >>>> ----------------------------------------------------------------------- - >>>> > - > >>>> This SF.net email is sponsored by: Splunk Inc. >>>> Still grepping through log files to find problems? Stop. >>>> Now Search log events and configuration files using AJAX and a browser. >>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> >>>> ----------------------------------------------------------------------- - >>>> > - > >>>> This SF.net email is sponsored by: Splunk Inc. >>>> Still grepping through log files to find problems? Stop. >>>> Now Search log events and configuration files using AJAX and a browser. >>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> >>>> >>>> >>>> >>> ------------------------------------------------------------------------ - >>> This SF.net email is sponsored by: Splunk Inc. >>> Still grepping through log files to find problems? Stop. >>> Now Search log events and configuration files using AJAX and a browser. >>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >>> ------------------------------------------------------------------------ - >>> This SF.net email is sponsored by: Splunk Inc. >>> Still grepping through log files to find problems? Stop. >>> Now Search log events and configuration files using AJAX and a browser. >>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >>> >>> >>> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> >> > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
|
From: <jo...@op...> - 2007-11-03 14:39:22
|
I tried to make some sense out of "aonocvs" and decided that since it=20 does not make any sense, to read your question based on the nearest=20 context. I presupposed that you might have meant a shorthand of=20 "anonymous cvs" and I said yes it supports anonymous CVS. What exactly=20 don't you understand in the following instructions? ----- OpenSIPStack CVS is hosted at sourceforge. It can be accessed via=20 anonymous login. For unix users, the following command line should download the latest=20 CVS Head from sourceforge. unix-shell>#cvs -d:pserver:ano...@op...: /cvsroot/opensipstack login unix-shell>#cvs -z3 -d:pserver:ano...@op...: /cvsroot/opensipstack co -P opensipstack You may also get cvs-get=20 <http://www.opensipstack.org/releases/tools/cvs-get.tar.gz> from here.=20 cvs-get is a PERL script that automates downloads of OpenSIPStack=20 modules. Simply extract cvs-get.tar.gz and execute: unix-shell>#cvs-get grab opensipstack unix-shell>#cvs-get grab opensbc unix-shell>#cvs-get grab atlsip ------ More instructions here http://sourceforge.net/cvs/?group_id=3D156710 voice wrote: > I was hoping you would have said something else. > > aonocvs does not work as password. > > It said cvs login authentiction failed... > > where do i set authentication to pserver? > > I have set invironments for .bashrc and .cshrc to point to /tmp for CVS= ROOT > > I am doing all this as root user. > > r > > ----- Original Message -----=20 > From: "jo...@op..." <joe...@gm...> > To: <ope...@li...> > Sent: Saturday, November 03, 2007 12:24 AM > Subject: Re: [OpenSIPStack] Dialog events > > > Yes, Anonymous access is supported. Make sure your authentication is se= t > to pserver: > > voice wrote: > =20 >> Hi Joe >> >> CVS asks' for a password? is it aonocvs? >> >> r >> >> ----- Original Message -----=20 >> From: "jo...@op..." <joe...@gm...> >> To: <ope...@li...> >> Sent: Thursday, November 01, 2007 9:45 PM >> Subject: Re: [OpenSIPStack] Dialog events >> >> >> voice wrote: >> >> =20 >>> Hi Joe >>> >>> The CVS is a version control website. I have installed openSBC from >>> >>> =20 >> source >> >> =20 >>> and compiled it aok. Is this CVS going to update that install? >>> >>> =20 >> CVS will only get the source files. You will need to recompile it afte= r >> the download and change your installed binary with the newly created o= ne >> >> >> >> =20 >>> Does it >>> over write existing source or does it update complied libraries etc..= =2E. >>> >>> >>> =20 >> It will update existing source not the binaries. >> >> >> =20 >>> Does it update contributions from others on this list and is it teste= d >>> before it is added to the CVS? >>> >>> =20 >> You are assured that all codes commited to CVS are tested code. >> >> >> >> =20 >>> And finally is it selective in what gets >>> updated? >>> >>> >>> =20 >> Yes >> >> >> =20 >>> r >>> >>> >>> ----- Original Message -----=20 >>> From: "Joegen E. Baclor" <joe...@gm...> >>> To: <ope...@li...> >>> Cc: <jo...@op...> >>> Sent: Wednesday, October 31, 2007 10:32 PM >>> Subject: Re: [OpenSIPStack] Dialog events >>> >>> >>> Instructions how to download from CVS can be found at >>> http://www.opensipstack.org/cvs.html >>> >>> voice wrote: >>> >>> >>> =20 >>>> Hi Joe >>>> >>>> Could you please also send me that elaboration explaining steps to >>>> >>>> >>>> =20 >>> download >>> >>> >>> =20 >>>> latest versions of the codes >>>> using CVS >>>> >>>> Robert. >>>> >>>> >>>> ----- Original Message -----=20 >>>> From: "jo...@op..." <joe...@gm...> >>>> To: <ope...@li...> >>>> Sent: Wednesday, October 17, 2007 5:54 PM >>>> Subject: Re: [OpenSIPStack] Dialog events >>>> >>>> >>>> Of course. See how RFC3265Agent and RFC3680Package is used in >>>> RegisterSessionManager. >>>> >>>> >>>> sebastian pastor wrote: >>>> >>>> >>>> >>>> =20 >>>>> Thanks, Joegen. >>>>> >>>>> >>>>> >>>>> Anyway, I'd like to know a bit more=85 >>>>> >>>>> My aim is, as developer, add a *new service* to the phone (for >>>>> =20 > acceeding > =20 >>>>> >>>>> =20 >>>> to >>>> >>>> >>>> >>>> =20 >>>>> media contents) and this will be based on exchange of SUBs and NTFY= s >>>>> >>>>> =20 >> with >> >> =20 >>>>> event=3Ddialog but with a diferent xml body than the standard. >>>>> >>>>> >>>>> >>>>> So, do you think it would be feasible or *realistic* trying to prog= ram >>>>> >>>>> >>>>> >>>>> =20 >>>> this >>>> >>>> >>>> >>>> =20 >>>>> whole behaviour based on existent OpenSipStack code ? It would be a= n >>>>> >>>>> >>>>> =20 >>> state >>> >>> >>> =20 >>>>> machine regarding to SUBSCRIBEs, renews, NTFYs, response codes=85 t= hat I >>>>> >>>>> >>>>> >>>>> =20 >>>> could >>>> >>>> >>>> >>>> =20 >>>>> catch from the REGISTER one. >>>>> >>>>> Or it's better waiting for you to have this functionality implement= ed? >>>>> >>>>> >>>>> >>>>> How much *time* do you estimate that we are talking about? >>>>> >>>>> >>>>> >>>>> Best Regards, >>>>> >>>>> >>>>> >>>>> -Sebasti=E1n Pastor- >>>>> >>>>> Junior Telecomunications Engineer (Malaga, >>>>> >>>>> =20 >> Spain) >> >> =20 >>>>> p.s BTW, *Lalith *you are a bit off-topic. But i attached a quick d= oc >>>>> >>>>> >>>>> =20 >>> that >>> >>> >>> =20 >>>>> i've elaborated explaining steps to download latest versions of the= >>>>> >>>>> =20 >> codes >> >> =20 >>>>> using CVS. Hope to help you. >>>>> >>>>> >>>>> 2007/10/17, Joegen E. Baclor <joe...@gm...>: >>>>> >>>>> >>>>> >>>>> >>>>> =20 >>>>>> Hi Sebastian, >>>>>> >>>>>> I am sorry but this feature has not made it yet to the mile stone >>>>>> release because SIP Trunking was deemed a more popular need of use= rs >>>>>> =20 > of > =20 >>>>>> OpenSBC. >>>>>> >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------= ---- >>>>>> =20 > - > =20 >> - >> >> =20 >>>>>> ------------------------------------------------------------------= ---- >>>>>> =20 > - > =20 >> - >> >> =20 >>> - >>> >>> >>> =20 >>>>>> This SF.net email is sponsored by: Splunk Inc. >>>>>> Still grepping through log files to find problems? Stop. >>>>>> Now Search log events and configuration files using AJAX and a >>>>>> =20 > browser. > =20 >>>>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>>>> ------------------------------------------------------------------= ---- >>>>>> =20 > - > =20 >> - >> >> =20 >>>>>> _______________________________________________ >>>>>> opensipstack-devel mailing list >>>>>> ope...@li... >>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>>> >>>>>> ------------------------------------------------------------------= ---- >>>>>> =20 > - > =20 >> - >> >> =20 >>>>>> No virus found in this incoming message. >>>>>> Checked by AVG Free Edition. >>>>>> Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: >>>>>> >>>>>> >>>>>> >>>>>> =20 >>>> 10/16/2007 2:14 PM >>>> >>>> >>>> >>>> >>>> --------------------------------------------------------------------= ---- >>>> =20 > - > =20 >>>> This SF.net email is sponsored by: Splunk Inc. >>>> Still grepping through log files to find problems? Stop. >>>> Now Search log events and configuration files using AJAX and a brows= er. >>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> >>>> --------------------------------------------------------------------= ---- >>>> =20 > - > =20 >>>> This SF.net email is sponsored by: Splunk Inc. >>>> Still grepping through log files to find problems? Stop. >>>> Now Search log events and configuration files using AJAX and a brows= er. >>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> >>>> >>>> >>>> =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 >>> >>> >>> >>> ---------------------------------------------------------------------= ---- >>> This SF.net email is sponsored by: Splunk Inc. >>> Still grepping through log files to find problems? Stop. >>> Now Search log events and configuration files using AJAX and a browse= r. >>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >>> >>> >>> =20 >> >> ----------------------------------------------------------------------= --- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser= =2E >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> ----------------------------------------------------------------------= --- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser= =2E >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> >> =20 > > > > -----------------------------------------------------------------------= -- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser.= > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > -----------------------------------------------------------------------= -- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser.= > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > =20 |
|
From: voice <vo...@ne...> - 2007-11-03 14:16:46
|
I was hoping you would have said something else. aonocvs does not work as password. It said cvs login authentiction failed... where do i set authentication to pserver? I have set invironments for .bashrc and .cshrc to point to /tmp for CVSROOT I am doing all this as root user. r ----- Original Message ----- From: "jo...@op..." <joe...@gm...> To: <ope...@li...> Sent: Saturday, November 03, 2007 12:24 AM Subject: Re: [OpenSIPStack] Dialog events Yes, Anonymous access is supported. Make sure your authentication is set to pserver: voice wrote: > Hi Joe > > CVS asks' for a password? is it aonocvs? > > r > > ----- Original Message ----- > From: "jo...@op..." <joe...@gm...> > To: <ope...@li...> > Sent: Thursday, November 01, 2007 9:45 PM > Subject: Re: [OpenSIPStack] Dialog events > > > voice wrote: > >> Hi Joe >> >> The CVS is a version control website. I have installed openSBC from >> > source > >> and compiled it aok. Is this CVS going to update that install? >> > > CVS will only get the source files. You will need to recompile it after > the download and change your installed binary with the newly created one > > > >> Does it >> over write existing source or does it update complied libraries etc.... >> >> > > > It will update existing source not the binaries. > > >> Does it update contributions from others on this list and is it tested >> before it is added to the CVS? >> > > You are assured that all codes commited to CVS are tested code. > > > >> And finally is it selective in what gets >> updated? >> >> > > Yes > > >> r >> >> >> ----- Original Message ----- >> From: "Joegen E. Baclor" <joe...@gm...> >> To: <ope...@li...> >> Cc: <jo...@op...> >> Sent: Wednesday, October 31, 2007 10:32 PM >> Subject: Re: [OpenSIPStack] Dialog events >> >> >> Instructions how to download from CVS can be found at >> http://www.opensipstack.org/cvs.html >> >> voice wrote: >> >> >>> Hi Joe >>> >>> Could you please also send me that elaboration explaining steps to >>> >>> >> download >> >> >>> latest versions of the codes >>> using CVS >>> >>> Robert. >>> >>> >>> ----- Original Message ----- >>> From: "jo...@op..." <joe...@gm...> >>> To: <ope...@li...> >>> Sent: Wednesday, October 17, 2007 5:54 PM >>> Subject: Re: [OpenSIPStack] Dialog events >>> >>> >>> Of course. See how RFC3265Agent and RFC3680Package is used in >>> RegisterSessionManager. >>> >>> >>> sebastian pastor wrote: >>> >>> >>> >>>> Thanks, Joegen. >>>> >>>> >>>> >>>> Anyway, I'd like to know a bit more… >>>> >>>> My aim is, as developer, add a *new service* to the phone (for acceeding >>>> >>>> >>>> >>> to >>> >>> >>> >>>> media contents) and this will be based on exchange of SUBs and NTFYs >>>> > with > >>>> event=dialog but with a diferent xml body than the standard. >>>> >>>> >>>> >>>> So, do you think it would be feasible or *realistic* trying to program >>>> >>>> >>>> >>> this >>> >>> >>> >>>> whole behaviour based on existent OpenSipStack code ? It would be an >>>> >>>> >> state >> >> >>>> machine regarding to SUBSCRIBEs, renews, NTFYs, response codes… that I >>>> >>>> >>>> >>> could >>> >>> >>> >>>> catch from the REGISTER one. >>>> >>>> Or it's better waiting for you to have this functionality implemented? >>>> >>>> >>>> >>>> How much *time* do you estimate that we are talking about? >>>> >>>> >>>> >>>> Best Regards, >>>> >>>> >>>> >>>> -Sebastián Pastor- >>>> >>>> Junior Telecomunications Engineer (Malaga, >>>> > Spain) > >>>> >>>> p.s BTW, *Lalith *you are a bit off-topic. But i attached a quick doc >>>> >>>> >> that >> >> >>>> i've elaborated explaining steps to download latest versions of the >>>> > codes > >>>> using CVS. Hope to help you. >>>> >>>> >>>> 2007/10/17, Joegen E. Baclor <joe...@gm...>: >>>> >>>> >>>> >>>> >>>>> Hi Sebastian, >>>>> >>>>> I am sorry but this feature has not made it yet to the mile stone >>>>> release because SIP Trunking was deemed a more popular need of users of >>>>> OpenSBC. >>>>> >>>>> >>>>> >>>>> ---------------------------------------------------------------------- - >>>>> > - > >>>>> ---------------------------------------------------------------------- - >>>>> > - > >> - >> >> >>>>> This SF.net email is sponsored by: Splunk Inc. >>>>> Still grepping through log files to find problems? Stop. >>>>> Now Search log events and configuration files using AJAX and a browser. >>>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>>> ---------------------------------------------------------------------- - >>>>> > - > >>>>> _______________________________________________ >>>>> opensipstack-devel mailing list >>>>> ope...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>> >>>>> ---------------------------------------------------------------------- - >>>>> > - > >>>>> No virus found in this incoming message. >>>>> Checked by AVG Free Edition. >>>>> Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: >>>>> >>>>> >>>>> >>> 10/16/2007 2:14 PM >>> >>> >>> >>> >>> ------------------------------------------------------------------------ - >>> This SF.net email is sponsored by: Splunk Inc. >>> Still grepping through log files to find problems? Stop. >>> Now Search log events and configuration files using AJAX and a browser. >>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >>> ------------------------------------------------------------------------ - >>> This SF.net email is sponsored by: Splunk Inc. >>> Still grepping through log files to find problems? Stop. >>> Now Search log events and configuration files using AJAX and a browser. >>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >>> >>> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> >> > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------------------------------------------------- > 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 |
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