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From: ehernaez <ope...@op...> - 2007-11-19 18:40:24
|
The next release of OSBC will allow routing rules based on the from-uri. This will allow you to exclude any domains that are not trusted. |
|
From: Joegen E. B. <joe...@gm...> - 2007-11-19 14:38:19
|
We use OpenSBC to traverse NAT everyday. If you think you are reporting a bug, then send more information such as ethereal captures and logs. Kevin Brennan wrote: > When a private IP address (example 192.168.1.112) is sent during registration openSBC does not seem to fix it. The AOR is placed in "Registration Status" using the private IP and registration OK fails to take place. Is there a setting I'm missing....or is openSBC just realy bad at NAT traversal. > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2005. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
|
From: Joegen E. B. <joe...@gm...> - 2007-11-19 14:37:14
|
Yes it is but current CVS version 1.1.5 is pretty much stable and it has numerous improvements and bug fixes based on bug reports posted in this list and from the just concluded sipit 21 interop. I strongly advise using CVS head instead. Kevin Brennan wrote: > How are stable releases indicated ? Is Version 1.1.4 the latest stable release ? > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2005. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
|
From: Joegen E. B. <joe...@gm...> - 2007-11-19 14:34:40
|
RTP Port range is at 30000-35000. There is currently no config provision to change this Kevin Brennan wrote: > Which range of RTP ports are used by OpenSBC by default (for firewall purposes). Can these be changed ? > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2005. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
|
From: Kevin B. <ke...@ei...> - 2007-11-19 14:24:57
|
When a private IP address (example 192.168.1.112) is sent during registration openSBC does not seem to fix it. The AOR is placed in "Registration Status" using the private IP and registration OK fails to take place. Is there a setting I'm missing....or is openSBC just realy bad at NAT traversal. |
|
From: Kevin B. <ke...@ei...> - 2007-11-19 10:17:11
|
How are stable releases indicated ? Is Version 1.1.4 the latest stable release ? |
|
From: Kevin B. <ke...@ei...> - 2007-11-19 10:04:57
|
Which range of RTP ports are used by OpenSBC by default (for firewall purposes). Can these be changed ? |
|
From: Kevin B. <ke...@ei...> - 2007-11-18 17:59:28
|
well it would seem more correct to use the R-URI for doing the lookup of local registered users in openSBC on INVITE from port 65086 instead of 'To'. I added a special check in OpenSER - of the INVITE is being routed to the SBC then 'To-URI' is overwrittend with the R-uri and this WORKS. On security - it would be nice if openSBC had separate trusted domains (checking From-URI) for incoming INVITES on port 65086. If you are using openSBC in this configuration INVITES on 5060 should all be from local UA's but on port 65086 they could be coming from ANY UA's (using foreign From-URI's). The only way to have incoming calls from foreign From-URI's is to 'Accept All Calls' - this basically leaves you with an open proxy. |
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From: Joegen E. B. <joe...@gm...> - 2007-11-18 06:24:36
|
Work is currently in the pipeline to support this scenario. Right now,
not resolving the To-URI for routing is not supported. If you can make
SER rewrite the to-uri to be the same as the r-uri, then this problem
will be solved.
Other problems
SBC crashed several times
one example
2007/11/17 15:45:42.544 HTTP Service:8945460 Error Accept failed for HTTP: Too many open files
2007/11/17 15:45:42.545 HTTP Service:8945460 Message Assertion fail: Operating System error, file tlibthrd.cxx, line 833, Error=24
get the message on the terminal "got here" alot
Some segmentation faults also
"OpenSBC" v1.1.4
Make sure your file handle limit per process is high enough. The
default is normally 1024 for linux. This is not high enough specially
if opensbc is dynamically creating threads to proxy rtp. There are
also bug fixes in 1.1.5 for zombie rtp threads which might occur during
hold/unhold. I suggest you upgrade using CVS.
Kevin Brennan wrote:
> What I would like to do is have OpenSCB act as an Outbound proxy NAT traversal solution in front of OpenSER (OpenSER acts as the Registrar and main proxy).
> Clients who need NAT traversal can set the "outbound proxy" on their SIP UA.
>
> The documentation for OSBC is pretty poor, I'm putting this down partly to help others and partly to get some answers to problems.
> Please give comments/advise. I didn't do much testing other than this, so there could be heaps of other issues.
>
> The setup is as follows
>
> UA--------------- OSCB ------------------------------------ OSER ---------------------------- PSTN
> [http://oscb.dn.com] [http://oser.dn.com] [http://pstn.dn.com]
>
> The UA is setup as follows
> username:123
> domain:oser.dn.com
> outbound proxy:oscb.dn.com
>
> There are no changes to OpenSER configuration to make this work, if you want you can register directly and make/recieve calls by omitting the "outbound proxy:oscb.dn.com" section on the UA config.
>
> REGISTRATION
> ============
> (note: the usr/psw are configured on OpenSER)
> OpenSBC's configuration
> SBC mode: B2BUpperReg
> Always Proxy Media: Yes
> (that's it, for registration, the rest is as default - don't even have Upper Registration Routes as oser.dn.com does not need translation(I guess))
>
> The UA registers with no problem,
>
> Here's a simplified snip from the register message
>
> REGISTER sip:oser.dn.com SIP/2.0. <-- Request URI
> From: "123"
> To: .
>
> As you cans see neither "Request URI" nor "To URI" are resolved to SBC's IP hence it's sent to the upper registration routes.
>
> When I check OpenSBC registration status, there are two entries (with same call-id)
> uri=123@213.224.224.7(ip of SBC) aor=sip:123@78.16.113.195:5065
> ur...@os... aor=sip:123@78.16.113.195:5065
>
> When I check OpenSER there is one contact in location DB
> sip:123@213.224.224.7:65086 (why port 65086 ??)
>
> You can see here, as expected, that the client registered with IP 78.16.113.195, this was then stored then masqueraded by OpenSBC and sent to OpenSER where the Contact IP is stored as 213.224.224.7 (The IP of OpenSBC).
>
> Any INVITES to OpenSER looking for 12...@os... will resolve it to sip:123@213.224.224.7:65086 (OpenSBC) and openSBC should then proxy it to the UA.
>
> INVITE, BASIC OUTGOING UA TEST CALL
> ================================
> To be able to accept an INVITE you need to make sure the domain used in the 'From URI' appears in the 'Trusted Domain' section.
> SBC config
> Trusted Domains oser.dn.com > add
>
> I then made a call to an echo test which is running on an Asterisk box.
> OpenSER handles the routing so no special routing needed to be configured on OpenSBC
>
> snip from invite
> INVITE sip:77...@os... SIP/2.0.
> From: "123"
> To: .
> Contact: .
>
> Since 'To URI' was not 'local' it's sent off to openser which then proxies to the Asterisk echo test.
>
> Resource Counters (during call)
> ICT=1 (what are these ? ) NICT=0 IST=0 NIST=0
> CallSession=2 Connection=1 Registration=0 RTPSession=2
> EventQueue=0 Cache=5 Garbage=3
>
> INVITE, BASIC UA-UA CALL
> =====================
> I made a call from UA to UA, both using 'outbound proxy' (ie. through SBC)
>
> No problem call worked 2 way voice
>
> Resource Counters (during call)
> ICT=6 NICT=0 IST=0 NIST=0
> CallSession=4 Connection=2 Registration=0 RTPSession=4
> EventQueue=0 Cache=9 Garbage=6
>
>
>
> PSTN INCOMING ** PROBLEM **
> =========================
> The PSTN sends calls with from URI=pstn.dn.com so this had to be set to allowed
> Trusted Domains pstn.dn.com > add
>
> Ok, so this was where I hit the first snag, perhaps someone can help on this one
>
> snip from INVITE
> INVITE sip:123@213.224.224.7:65086 SIP/2.0.
> From: "0860111345"
> To:
> Contact: .
>
> To explain what happens before this INVITE arrived
> An INVITE is sent from pstn.dn.com to oser.pn.com, as you can see the INVITE is to sip:35...@os..., we use the alias function in OpenSER to allow a UA have multiple numbers (this is equaly true for other text aliases, the 'to uri' could be jo...@os...). OpenSER performs an internal lookup on this sip:35...@os..., translates it to 12...@os... then a lookup is performed to find the contact sip:123@213.224.224.7:65086 and this is sent to the SBC
>
> As you can see the request URI (sip:123@213.224.224.7:65086) is correct, it matches the IP of the SBC.
> I would think this would be considered local and do a lookup in the local SBC registar and it would then be proxied to the UA, but alas no.
> It seems the SBC will only lookup the "to URI" which can't be found in the SBC and worse is sent back to OpenSER which sends it back to the SBC etc..
>
>
>
> ** IS THERE A SOLUTION TO THIS ??? ** (no, we will not re-write the 'to uri' in OpenSER)
>
> Other problems
> SBC crashed several times
>
> one example
> 2007/11/17 15:45:42.544 HTTP Service:8945460 Error Accept failed for HTTP: Too many open files
> 2007/11/17 15:45:42.545 HTTP Service:8945460 Message Assertion fail: Operating System error, file tlibthrd.cxx, line 833, Error=24
> get the message on the terminal "got here" alot
> Some segmentation faults also
>
> "OpenSBC" v1.1.4
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2005.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> Opensipstack-osbcdevel mailing list
> Ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel
>
>
>
|
|
From: ehernaez <ope...@op...> - 2007-11-18 05:09:29
|
The short answer to your question is that OSBC uses the to-uri for all routing. So, if you want to use it for NAT traversal, the registrar must put the AOR in the to line. There are changes underway that will allow more granular routing rules in OSBC, but it's not there currently. Also, when performing Upper Registration, OSBC uses the base SIP port plus 60020 as a back-door in order to recognize calls for which itshould have a local registration. This explains whythe AOR seen in SER was 65086. |
|
From: Kevin B. <ke...@ei...> - 2007-11-17 19:43:56
|
test test |
|
From: Kevin B. <ke...@ei...> - 2007-11-17 16:30:28
|
What I would like to do is have OpenSCB act as an Outbound proxy NAT traversal solution in front of OpenSER (OpenSER acts as the Registrar and main proxy).
Clients who need NAT traversal can set the "outbound proxy" on their SIP UA.
The documentation for OSBC is pretty poor, I'm putting this down partly to help others and partly to get some answers to problems.
Please give comments/advise. I didn't do much testing other than this, so there could be heaps of other issues.
The setup is as follows
UA--------------- OSCB ------------------------------------ OSER ---------------------------- PSTN
[http://oscb.dn.com] [http://oser.dn.com] [http://pstn.dn.com]
The UA is setup as follows
username:123
domain:oser.dn.com
outbound proxy:oscb.dn.com
There are no changes to OpenSER configuration to make this work, if you want you can register directly and make/recieve calls by omitting the "outbound proxy:oscb.dn.com" section on the UA config.
REGISTRATION
============
(note: the usr/psw are configured on OpenSER)
OpenSBC's configuration
SBC mode: B2BUpperReg
Always Proxy Media: Yes
(that's it, for registration, the rest is as default - don't even have Upper Registration Routes as oser.dn.com does not need translation(I guess))
The UA registers with no problem,
Here's a simplified snip from the register message
REGISTER sip:oser.dn.com SIP/2.0. <-- Request URI
From: "123"
To: .
As you cans see neither "Request URI" nor "To URI" are resolved to SBC's IP hence it's sent to the upper registration routes.
When I check OpenSBC registration status, there are two entries (with same call-id)
uri=123@213.224.224.7(ip of SBC) aor=sip:123@78.16.113.195:5065
ur...@os... aor=sip:123@78.16.113.195:5065
When I check OpenSER there is one contact in location DB
sip:123@213.224.224.7:65086 (why port 65086 ??)
You can see here, as expected, that the client registered with IP 78.16.113.195, this was then stored then masqueraded by OpenSBC and sent to OpenSER where the Contact IP is stored as 213.224.224.7 (The IP of OpenSBC).
Any INVITES to OpenSER looking for 12...@os... will resolve it to sip:123@213.224.224.7:65086 (OpenSBC) and openSBC should then proxy it to the UA.
INVITE, BASIC OUTGOING UA TEST CALL
================================
To be able to accept an INVITE you need to make sure the domain used in the 'From URI' appears in the 'Trusted Domain' section.
SBC config
Trusted Domains oser.dn.com > add
I then made a call to an echo test which is running on an Asterisk box.
OpenSER handles the routing so no special routing needed to be configured on OpenSBC
snip from invite
INVITE sip:77...@os... SIP/2.0.
From: "123"
To: .
Contact: .
Since 'To URI' was not 'local' it's sent off to openser which then proxies to the Asterisk echo test.
Resource Counters (during call)
ICT=1 (what are these ? ) NICT=0 IST=0 NIST=0
CallSession=2 Connection=1 Registration=0 RTPSession=2
EventQueue=0 Cache=5 Garbage=3
INVITE, BASIC UA-UA CALL
=====================
I made a call from UA to UA, both using 'outbound proxy' (ie. through SBC)
No problem call worked 2 way voice
Resource Counters (during call)
ICT=6 NICT=0 IST=0 NIST=0
CallSession=4 Connection=2 Registration=0 RTPSession=4
EventQueue=0 Cache=9 Garbage=6
PSTN INCOMING ** PROBLEM **
=========================
The PSTN sends calls with from URI=pstn.dn.com so this had to be set to allowed
Trusted Domains pstn.dn.com > add
Ok, so this was where I hit the first snag, perhaps someone can help on this one
snip from INVITE
INVITE sip:123@213.224.224.7:65086 SIP/2.0.
From: "0860111345"
To:
Contact: .
To explain what happens before this INVITE arrived
An INVITE is sent from pstn.dn.com to oser.pn.com, as you can see the INVITE is to sip:35...@os..., we use the alias function in OpenSER to allow a UA have multiple numbers (this is equaly true for other text aliases, the 'to uri' could be jo...@os...). OpenSER performs an internal lookup on this sip:35...@os..., translates it to 12...@os... then a lookup is performed to find the contact sip:123@213.224.224.7:65086 and this is sent to the SBC
As you can see the request URI (sip:123@213.224.224.7:65086) is correct, it matches the IP of the SBC.
I would think this would be considered local and do a lookup in the local SBC registar and it would then be proxied to the UA, but alas no.
It seems the SBC will only lookup the "to URI" which can't be found in the SBC and worse is sent back to OpenSER which sends it back to the SBC etc..
** IS THERE A SOLUTION TO THIS ??? ** (no, we will not re-write the 'to uri' in OpenSER)
Other problems
SBC crashed several times
one example
2007/11/17 15:45:42.544 HTTP Service:8945460 Error Accept failed for HTTP: Too many open files
2007/11/17 15:45:42.545 HTTP Service:8945460 Message Assertion fail: Operating System error, file tlibthrd.cxx, line 833, Error=24
get the message on the terminal "got here" alot
Some segmentation faults also
"OpenSBC" v1.1.4
|
|
From: Joegen E. B. <joe...@gm...> - 2007-11-17 00:34:53
|
Hi, Solegy and Tandberg booths re right beside each other in SIPIt 21. We did get the chance to interop and there were almost no problems encountered aside from 1. OpenSBC cannot handle very large SIP messages particularly those which are higher than the MTU. If you are using Tandberg user agents, make sure to only set audio and video codecs that you need to shrink the packets. Joegen Curt Shaffer wrote: > We were looking to test interaction between openSBC and a Tandberg. Has anyone on the list attempted this before? Any gotchas or advice? > > > > > > Thanks > > > > > > Curt > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2005. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
|
From: Curt S. <csh...@gm...> - 2007-11-16 22:35:23
|
We were looking to test interaction between openSBC and a Tandberg. Has anyone on the list attempted this before? Any gotchas or advice? Thanks Curt |
|
From: Gaurav K. <gkh...@is...> - 2007-11-16 05:59:14
|
Hi Joegen, We were able to get trunking working for inbound calls with OpenSBC. For the outbound trunking issue, you mentioned:- " 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" I subsequently tried that and found that sbc crashes in SBCSIPTrunkReg::GetEgressAuthInfo function at the start of For loop. *Configuration* OpenSBC 192.168.96.123 Xlite 192.168.96.36 Public LAN IP 59.95.152.178 voipvoip.com 69.90.209.57 *B2BAUA Route* [sip:*@192.168.96.123] sip:sip3.voipvoip.com:5060;sip-trunk=true *Trunking* <root> <siptrunk trunk-name="SIP" route-set="sip3.voipvoip.com" sip-domain="59.95.152.178" expires="3600"> <trunk-accounts> <account user-name="phonenumber" auth-user-name="phonenumber" auth-password="password" inbound-route="sip:102@192.168.96.123:5060" expires="3600"/> </trunk-accounts> </siptrunk> </root> The pcap file is attached for your reference. As always, thanks in advance for your help! Regards, Gaurav |
|
From: Joegen E. B. <joe...@gm...> - 2007-11-12 08:23:24
|
Dinesh, 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, sip:192.168.96.112, sip:gafachi.com HTH Joegen Dinesh Dialani wrote: > Hi, > > I want to configure OpenSBC such that it first call using enumlookup > and if it fails then > it should call through voip provider. > > I have done setting in B2BUA Route section as : > > [sip:*] enum: mydomain.com <http://mydomain.com> //First Preference : > > [sip:*@192.168.96.112 <mailto:sip:*@192.168.96.112>] > sip:sip.gafachi.com <http://sip.gafachi.com> //Second preference if > first get failed: > > When I dial a number that is not in DNS, OpenSBC tries to resolve it 3 > times and then gives response "NO SUCH NAME" and then > create a INVITE request As: > > SIP/SDPRequest: INVITE enum:141...@e1... > <mailto:enum:141...@e1...> > > and send it to my DNS server. > > how can i achieve that in case of enumLookup fails it should send > request using my voip Provider. > > Thanks in Advance > > Dinesh > > > > > ----- Original Message ----- > > > From: "Joegen E. Baclor" <joe...@gm... > <mailto:joe...@gm...>> > > > To: < ope...@li... > <mailto:ope...@li...>>; > > <sub...@wi... <mailto:sub...@wi...>> > > > Sent: Thursday, September 20, 2007 2:57 PM > > > Subject: Re: [OpenSBC] Enum routinng in Opensbc > > > > > > > > > > Try this in B2BUA Route > > > > > > > > [sip:1111*] enum:e164.org <http://e164.org> > > > > > > > > This means you want to resolve 1111 using e164.org > <http://e164.org> enum server. > > > > > > > > Joegen > > > > > > > > > > > > sub...@wi... <mailto:sub...@wi...> wrote: > > > > > Hi, > > > > > Has anybody worked with ENUM routing in opensbc?? If so, kindly > > > > > provide the details of implementation. > > > > > > > > > > Regards, > > > > > Subha, > > > > > Wipro Technologies, > > > > > Bangalore > > > > > > > > ------------------------------------------------------------------------ > > > > > > > > > > > > > > > > ------------------------------------------------------------------------- > > > > > This SF.net email is sponsored by: Microsoft > > > > > Defy all challenges. Microsoft(R) Visual Studio 2005. > > > > > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > > > > > > > > ------------------------------------------------------------------------ > > > > > > > > > > _______________________________________________ > > > > > Opensipstack-osbcdevel mailing list > > > > > Ope...@li... > <mailto:Ope...@li...> > > > > > > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > > > > > No virus found in this incoming message. > > > > > Checked by AVG Free Edition. > > > > > Version: 7.5.487 / Virus Database: 269.13.25/1018 - Release Date: > > > 9/19/2007 3:59 PM > > > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------- > > > > This SF.net email is sponsored by: Microsoft > > > > Defy all challenges. Microsoft(R) Visual Studio 2005. > > > > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > <http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/> > > > > _______________________________________________ > > > > Opensipstack-osbcdevel mailing list > > > > Ope...@li... > <mailto:Ope...@li...> > > > > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > 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-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > ------------------------------------------------------------------------ > > 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: Gaurav K. <gkh...@is...> - 2007-11-12 08:04:56
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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 |
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From: Joegen E. B. <joe...@gm...> - 2007-11-07 05:11:06
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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: Dinesh D. <din...@gm...> - 2007-11-06 12:33:53
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Hi, I want to configure OpenSBC such that it first call using enumlookup and if it fails then it should call through 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: When I dial a number that is not in DNS, OpenSBC tries to resolve it 3 times and then gives response "NO SUCH NAME" and then create a INVITE request As: SIP/SDPRequest: INVITE enum:141...@e1... and send it to my DNS server. how can i achieve that in case of enumLookup fails it should send request using my voip Provider. Thanks in Advance Dinesh > > ----- Original Message ----- > > From: "Joegen E. Baclor" <joe...@gm...> > > To: <ope...@li...>; > <sub...@wi...> > > Sent: Thursday, September 20, 2007 2:57 PM > > Subject: Re: [OpenSBC] Enum routinng in Opensbc > > > > > > > Try this in B2BUA Route > > > > > > [sip:1111*] enum:e164.org > > > > > > This means you want to resolve 1111 using e164.org enum server. > > > > > > Joegen > > > > > > > > > sub...@wi... wrote: > > > > Hi, > > > > Has anybody worked with ENUM routing in opensbc?? If so, kindly > > > > provide the details of implementation. > > > > > > > > Regards, > > > > Subha, > > > > Wipro Technologies, > > > > Bangalore > > > > > ------------------------------------------------------------------------ > > > > > > > > > > > ------------------------------------------------------------------------- > > > > This SF.net email is sponsored by: Microsoft > > > > Defy all challenges. Microsoft(R) Visual Studio 2005. > > > > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > > > > > ------------------------------------------------------------------------ > > > > > > > > _______________________________________________ > > > > Opensipstack-osbcdevel mailing list > > > > Ope...@li... > > > > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > > > > > > > ------------------------------------------------------------------------ > > > > > > > > No virus found in this incoming message. > > > > Checked by AVG Free Edition. > > > > Version: 7.5.487 / Virus Database: 269.13.25/1018 - Release Date: > > 9/19/2007 3:59 PM > > > > > > > > > > > > > > ------------------------------------------------------------------------- > > > This SF.net email is sponsored by: Microsoft > > > Defy all challenges. Microsoft(R) Visual Studio 2005. > > > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > > > _______________________________________________ > > > Opensipstack-osbcdevel mailing list > > > Ope...@li... > > > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > > > > > > > > |
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From: Dinesh D. <din...@gm...> - 2007-11-01 11:33:05
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Hi, I have tried to use XML configuration in "SIP Trunk Config". But it does not send a register request to VoIP provider whose details is stored in [Trunk Accounts] section as seen in Wireshark Traces. Can I get help on Sip Trunking with OpenSBC ? I have gone through various classes of OpenSBC to implement it(B2B, Complete SIP part, OpenSBC configuration files and so on). So if even I will be guided on code level then I can do the changes. Thanks in Advance. Dinesh On 10/17/07, Joegen E. Baclor <joe...@gm...> wrote: > > This is a cross post. > --- > > Hi, > > SIP Trunking is not prime time yet but you may already try it using the > latest CVS copy of OpenSBC/OpenSIPStack. To Enable Trunking, you must > provide an XML configuration in "SIP Trunk Config". Below is a template > XML config. In this sample config, sip:win32.opensipstack.org is > assumed to be the internal domain of OpenSBC while > sip:opteron.opensipstack.org is the domain of your SIP Provider. > > [SIPTrunk] > * trunk-name: This is the unique name OpenSBC will use to identify you > SIP Trunk > * route-set: This is the DNS resolvable domain or IP address of your > trunk provider > * sip-domain: This is the SIP Domain used as the host part of the To > and From URIs > * expires: Global expire interval for trunk registrations in seconds > > [Trunk-Accounts] > * account - An instance of a virtual UA that will register to the Trunk > Provider domain > ** user-name - The user part of the From-URI > ** auth-user-name - User name used for Authorization and Authentication > ** auth-password - Password used for Authorization and Authentication > ** inbound-route - URI specifying the identity of the UA in the internal > domain > ** expires - If set, this will be the expires used when the virtual UA > registers to the Trunk Provider > > [Transient-Accounts] - Transient accounts are similar to normal > Trunk-Account in terms of the parameters. The only difference is that > they are also meant to be shared (in round robin fashion) by calls which > are not defined in the normal trunk-accounts. This is normally used if > you have a few accounts with a Trunk Provider and is meant to be shared > by all your external users. > > > > ------------------------START OF XML > CONFIG---------------------------------- > > <root> > <siptrunk trunk-name="opteron.opensipstack.org" > route-set="opteron.opensipstack.org" > sip-domain="opteron.opensipstack.org" > expires="10"> > > <trunk-accounts> > <account user-name="1001" > auth-user-name="1001" > auth-password="1001" > inbound-route="sip:90...@wi..." > expires="3600" /> > <account user-name="1002" > auth-user-name="1002" > auth-password="1002" > inbound-route="sip:90...@wi..." > expires="3600" /> > <account user-name="1003" > auth-user-name="1003" > auth-password="1003" > inbound-route="sip:90...@wi..." > expires="3600" /> > </trunk-accounts> > > <transient-accounts> > <account user-name="1001" > auth-user-name="1001" > auth-password="1001" > inbound-route="sip:90...@wi..." /> > <account user-name="1002" > auth-user-name="1002" > auth-password="1002" > inbound-route="sip:90...@wi..." /> > <account user-name="1003" > auth-user-name="1003" > auth-password="1003" > inbound-route="sip:90...@wi..." /> > </transient-accounts> > > </siptrunk> > </root> > > Joegen > > Dinesh Dialani wrote: > > > > Hi All, > > > > > > > > I want to use Open SBC for *SIP TRUNKING*. > > > > > > > > Here is the scenario. > > > > > > > > Internal LAN External LAN > > > > ------------------------------------ > > > > | > > > > Softphone ----> PBX ------|----> OpenSBC -----> Voip Provider > > > > > > | > > > > | > > > > ------------------------------------- > > > > > > > > I wish that only OpenSBC should be visible to external world and thus > > it should be able to register itself to VoipProvider. > > > > > > > > Also I want that OpenSBC should use Enum lookup first for E-164 > > numbers on our enum server and if OpenSBC does not receive any > > response from Enum server then it should be able to connect the call > > through Voip Provider. > > > > > > > > Now here are the questions. > > > > > > > > 1. How to register OpenSBC with VoipProvider irrespective of PBX? > > > > t means that there should be fields in Web GUI to enter registration > > request for Voip Provider and the moment OpenSBC service is started, > > it should register itself with Voip Provider. > > > > 2. How to set preference order between Enum lookup and normal call > > through VoipProvider? > > > > It means whenever an Sip INVITE is sent from PBX to OpenSBC for long > > distance calls, OpenSBC should first search that number in our enum > > servers and if it is found, call is made directly to the Receiver else > > OpenSBC should direct the call to VoipProvider and create a normal call. > > > > 3. What are the entries to be given on B2B routing page for above > > Enum and normal call to VoipProvider? > > > > > > > > Thanks in advance for you help. > > > > > > > > Dinesh > > > > > > > > ------------------------------------------------------------------------ > > > > > ------------------------------------------------------------------------- > > 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-osbcdevel mailing list > > Ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > > ------------------------------------------------------------------------ > > > > 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-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > |
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From: Joegen E. B. <joe...@gm...> - 2007-10-26 03:30:41
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This should be an accepted behavior. If OpenSBC is in media proxy mode, only the inbound leg RTP is visible to the caller. Thus, a session modification is only applicable to that leg. Can you make sure that your GW only attempted to change the address and not the codec? Asim wrote: > Hi, I have openSBC is runnung in B2BUperReg mode, UA--------OSBC-------------------PSTN Gateway, when the call is coming from the PSTN and PSTN sending re-invite to change the RTP port OpenSBC is changing RTP stream only on incoming direction not outgoing direction that cause one way audio comunication means UA can listen PSTN user but PSTN user cant any idea ? > > > > > > Regards > > > Asim > > > > ------------------------------------------------------------------------- > 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-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
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From: Asim <ar...@gm...> - 2007-10-25 16:57:37
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Hi, I have openSBC is runnung in B2BUperReg mode, UA--------OSBC-------------------PSTN Gateway, when the call is coming from the PSTN and PSTN sending re-invite to change the RTP port OpenSBC is changing RTP stream only on incoming direction not outgoing direction that cause one way audio comunication means UA can listen PSTN user but PSTN user cant any idea ? Regards Asim |
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From: Joegen E. B. <jb...@so...> - 2007-10-25 02:31:07
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Hi Tim, Go ahead, send me the logs and topology directly via email at jo...@op... --------------------------------------> Hi Joegen, Sorry for the delay I had to rebuild my test environment, and for the last week I have been trying to post the logs. Unfortunately they are too big, and they won't post, no matter hwo I try to cut them down. Can I email you directly a copy of the log along with the network design in a PDF? Tim |
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From: Joegen E. B. <joe...@gm...> - 2007-10-18 15:07:20
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That is an indication that the Call is getting Proxied instead of using the B2BUA Core. Try setting the Mode to B2BUA Only. gee...@wi... wrote: > Hi Joegen, > > Thanks for the reply. > > But I have checked the "Rewrite To URI" checkbox but still its not > rewriting the To URI. > > Geethanjali > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf > Of Joegen E. Baclor > Sent: Thursday, October 18, 2007 7:55 PM > To: ope...@li... > Subject: Re: [OpenSBC] OpenSBC B2BUA routing > > Make sure you put a check mark on the "Rewrite To URI" check box in > B2BUA routes. > > > gee...@wi... wrote: > >> Hi, >> >> I am using opensbc version 1.1.4 opensipstack version 1.1.7. >> >> I am trying to use B2BUA routes of SBC with Rewrite-to uri field >> checked. >> What I have observed is that the To-uri is not getting rewritten. Is >> > it > >> the actual functionality or a error. >> >> I have just specified the B2BUA routes and configured the SBC to be >> FullMode. >> >> Can someone help me in solving the issue. >> >> Regards, >> Geethanjali >> Wipro Technologies >> >> >> >> >> >> > ------------------------------------------------------------------------ > - > >> 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-osbcdevel mailing list >> Ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel >> >> >> >> > > > ------------------------------------------------------------------------ > - > 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-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. > > WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. > > www.wipro.com > > > |
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From: <gee...@wi...> - 2007-10-18 14:57:14
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Hi Joegen, Thanks for the reply. But I have checked the "Rewrite To URI" checkbox but still its not rewriting the To URI. Geethanjali -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Joegen E. Baclor Sent: Thursday, October 18, 2007 7:55 PM To: ope...@li... Subject: Re: [OpenSBC] OpenSBC B2BUA routing Make sure you put a check mark on the "Rewrite To URI" check box in=20 B2BUA routes. gee...@wi... wrote: > Hi, > > I am using opensbc version 1.1.4 opensipstack version 1.1.7. > > I am trying to use B2BUA routes of SBC with Rewrite-to uri field > checked. > What I have observed is that the To-uri is not getting rewritten. Is it > the actual functionality or a error. > > I have just specified the B2BUA routes and configured the SBC to be > FullMode.=20 > > Can someone help me in solving the issue. > > Regards, > Geethanjali > Wipro Technologies > > > > > ------------------------------------------------------------------------ - > 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-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > =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-osbcdevel mailing list Ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel |
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