opensipstack-osbcdevel — Discussions on OpenSBC

Re: [OpenSBC] Phone registration from behind NAT to sipX behind NAT

[OpenSBC F. <ope...@op...>]

Excellent!

I'm able to register the phone now remotely.

I think I'm missing one last translation though because I can't call my internal extensions...

Also, I didn't add any info to the Sology section...  that's what was in the INI by default.

Here's the log:

2009/01/01 19:43:57.866 INF: [CID=0x88db9862] <<< INVITE sip:0...@si... SIP/2.0 Method(INVITE)  SRC: 72.12.80.199:50941:UDP enc=0 bytes=976
2009/01/01 19:43:57.870 INF: [CID=0x88db9862] *** CREATED (UAS) CALL *** MmM4MzVmZTYzNTMzMDMwMmI4N2Y5MTBkNjc2ZDA2NTc.
2009/01/01 19:43:57.871 INF: [CID=0x88db9862] *** NO REGISTRATION FOUND *** Fetching route through local database URI sip:0...@si...
2009/01/01 19:43:57.871 INF: [CID=0x88db9862] *** NO STATIC ROUTE *** URI sip:0...@si...
2009/01/01 19:43:57.872 PWL: [CID=0x00000000] *** DNSSRV *** _sip._udp.sipxecs.info -> 76.178.252.229:5060
2009/01/01 19:43:57.872 PWL: [CID=0x00000000] *** DNSSRV *** _sip._udp.sipxecs.info -> 76.178.252.229:5060
2009/01/01 19:43:57.873 INF: [CID=0x88db9862] *** CALL TEAR DOWN *** MmM4MzVmZTYzNTMzMDMwMmI4N2Y5MTBkNjc2ZDA2NTc.-connection
2009/01/01 19:43:57.874 INF: [CID=0x88db9862] *** B2BUA CONNECTION OnDestroySession ***
2009/01/01 19:43:57.876 INF: [CID=0x88db9862] >>> SIP/2.0 100 Trying Method(INVITE) DST: 72.12.80.199:50941:UDP SRC=76.178.252.229:5061 enc=0 bytes=321
2009/01/01 19:44:02.877 INF: [CID=0x88db9862] *** DESTROYED B2BUA CONNECTION *** 0x0x92d8000
2009/01/01 19:44:02.878 INF: [CID=0x88db9862] *** COUNTERS *** ICT=0 NICT=0 IST=4 NIST=0 TIMERS=4 CALL=0 CONN=0 REG=0 RTP=0 QUEUE=1 CACHE=2 GC=6 TOTALCONN=9 TOTALREG=0 TOTALSZR=3 AVGDUR=12483 HIGHDUR=20950 IDLETIME=5
2009/01/01 19:44:02.878 INF: [CID=0x88db9862] *** DESTROYED CALL *** MmM4MzVmZTYzNTMzMDMwMmI4N2Y5MTBkNjc2ZDA2NTc.
2009/01/01 19:44:09.686 INF: [CID=0x88db9862] <<< CANCEL sip:0...@si... SIP/2.0 Method(CANCEL)  SRC: 72.12.80.199:50941:UDP enc=0 bytes=353
2009/01/01 19:44:09.691 INF: [CID=0x88db9862] >>> SIP/2.0 405 Method Not Allowed Method(CANCEL) DST: 72.12.80.199:50941:UDP SRC=76.178.252.229:5061 enc=0 bytes=333



Here's my current ini:

[Solegy]
RTTS-Client-Address=76.178.252.229

[OpenSBC-General-Parameters]
SIP-Log-Level=1
PTRACE-Log-Level=1
Log-File-Prefix=b2bua
SBC-Application-Mode=B2BUpperReg Mode
Interface-Address Array Size=0
Enable-Backdoor-Port=True
Enable-Trunk-Port=True
Enable-Calea-Port=True
RTP-Min-Port=30000
RTP-Max-Port=35000
Enable-Local-Refer=False
Max-Forwards=70
Encryption-Mode=XOR
Encryption-Key=GS
Transaction-Thread-Count=10
Session-Thread-Count=10
Alerting-Timeout=30000
Seize-Timeout=60000
SIP-Timer-B=Default
SIP-Timer-H=Default
Session-Keep-Alive=1800
Session-Max-Life-Span=10800
Max-Concurrent-Session=100
Max-Call-Rate-Per-Second=10

[Upper-Registration]
Enable-Stateful-Reg=True
Rewrite-TO-Domain=True
Rewrite-FROM-Domain=True
Route-List Array Size=1
Route-List 1=[sip:*@sipxecs.info] sip:xyzcompany.com;domain=xyzcompany.com

[Internal-DNS-Mapping]
Internal-DNS-Map Array Size=2
Internal-DNS-Map 1=[sip:xyzcompany.com] sip:172.16.1.2:5060
Internal-DNS-Map 2=[sip:sipx.xyzcompany.com] sip:172.16.1.2:5060

[Proxy-Relay-Routes]
Drop-Routes-On-Ping-Timeout=False
Proxy-Resolve-To-URI=True
Route-List Array Size=0

[B2BUA-Routes]
Enable-Route-Scripting=False
Route-Script=b2bua-route.cscript
Route-List Array Size=0
Route-List 1=[sip:sipxecs.info] sip:xyzcompany.com
Insert-Route-Header=True
Rewrite-TO-URI=True
Prepend-ISUP-OLI=False
Route-By-Request-URI=False
Route-By-To-URI=False
Drop-Routes-On-Ping-Timeout=False
Use-External-XML=False
External-XML-File=b2bua-route.xml

[Media-Server]
Enable-Media-Server=False
Media-Server-Number=5000
Codec-List Array Size=0
No-RTP-Proxy-On-All-Transfers=False
Enable-Announcement-Service=False
4xx-Error-Map=prompts/basic/cant_complete.wav
5xx-Error-Map=prompts/basic/cant_complete.wav
6xx-Error-Map=prompts/basic/cant_complete.wav
Announcement-Error-Map Array Size=0

[Outbound-Proxies]
Outbound-Proxies Array Size=0

[Local-Domain-Accounts]
Accept-All-Registration=True
Account-List Array Size=0

Re: [OpenSBC] Phone registration from behind NAT to sipX behind NAT

[OpenSBC F. <ope...@op...>]

Mike,


In the Local-Domain-Accounts section, try changing Accept-All-Registration to =TRUE.   


 Also, there is no need to enter anything in the Solegy section.  These parameters are useful  only to Solegy customers.


 


Regards,


~Eric 


 


 


 

[OpenSBC] Phone registration from behind NAT to sipX behind NAT

[OpenSBC F. <ope...@op...>]

Ok, since I got everything else working today (with a help from Joegen), I figured I'd go for the last piece I really care about...  far-end NAT traversal.

I have been referring to Raymond's excellent work over on the sipX Wiki ( [SipXecs-OpenSBC Setup Guide|http://sipx-wiki.calivia.com/index.php/SipXecs-OpenSBC_Setup_Guide_for_Different_Call_Scenario] ).

It is my understanding that the Upper Registration page should be able to mangle/rewrite the registration requests forward in towards my PBX (which has a different domain name inside than what I have outside).

So all calls / registrations in to *@sipxecs.info should be rewritten to be *@xyzcompany.com.

I believe the remote registrations would be working now if my internal domain was the same judging by the log below.

2009/01/01 18:23:02.239 INF: [CID=0x4ee4b71e] <<< REGISTER sip:sipxecs.info SIP/2.0 Method(REGISTER)  SRC: 72.12.80.199:11649:UDP enc=0 bytes=563
2009/01/01 18:23:02.241 PWL: [CID=0x00000000] *** DNSSRV *** _sip._udp.sipxecs.info -> 76.178.252.229:5060
2009/01/01 18:23:02.241 INF: [CID=0x4ee4b71e] *** LOCAL REG *** sip:20...@si...
2009/01/01 18:23:02.244 INF: [CID=0x4ee4b71e] >>> SIP/2.0 401 Unauthorized Method(REGISTER) DST: 72.12.80.199:11649:UDP SRC=76.178.252.229:5061 enc=0 bytes=488
2009/01/01 18:23:03.166 INF: [CID=0x4ee4b71e] <<< REGISTER sip:sipxecs.info SIP/2.0 Method(REGISTER)  SRC: 72.12.80.199:11649:UDP enc=0 bytes=786
2009/01/01 18:23:03.168 PWL: [CID=0x00000000] *** DNSSRV *** _sip._udp.sipxecs.info -> 76.178.252.229:5060
2009/01/01 18:23:03.168 INF: [CID=0x4ee4b71e] *** LOCAL REG *** sip:20...@si...
2009/01/01 18:23:03.170 INF: [CID=0x4ee4b71e] >>> SIP/2.0 403 Forbidden Method(REGISTER) DST: 72.12.80.199:11649:UDP SRC=76.178.252.229:5061 enc=0 bytes=338


Here's my ini file...

[Solegy]
RTTS-Client-Address=76.178.252.229

[OpenSBC-General-Parameters]
SIP-Log-Level=1
PTRACE-Log-Level=1
Log-File-Prefix=b2bua
SBC-Application-Mode=B2BUpperReg Mode
Interface-Address Array Size=0
Enable-Backdoor-Port=True
Enable-Trunk-Port=True
Enable-Calea-Port=True
RTP-Min-Port=30000
RTP-Max-Port=35000
Enable-Local-Refer=False
Max-Forwards=70
Encryption-Mode=XOR
Encryption-Key=GS
Transaction-Thread-Count=10
Session-Thread-Count=10
Alerting-Timeout=30000
Seize-Timeout=60000
SIP-Timer-B=Default
SIP-Timer-H=Default
Session-Keep-Alive=1800
Session-Max-Life-Span=10800
Max-Concurrent-Session=100
Max-Call-Rate-Per-Second=10

[Upper-Registration]
Enable-Stateful-Reg=True
Rewrite-TO-Domain=True
Rewrite-FROM-Domain=True
Route-List Array Size=1
Route-List 1=[sip:*@sipxecs.info] sip:xyzcompany.com;domain=xyzcompany.com

[Internal-DNS-Mapping]
Internal-DNS-Map Array Size=0

[Proxy-Relay-Routes]
Drop-Routes-On-Ping-Timeout=False
Proxy-Resolve-To-URI=True
Route-List Array Size=0

[B2BUA-Routes]
Enable-Route-Scripting=False
Route-Script=b2bua-route.cscript
Route-List Array Size=0
Route-List 1=[sip:sipxecs.info] sip:xyzcompany.com
Insert-Route-Header=True
Rewrite-TO-URI=True
Prepend-ISUP-OLI=False
Route-By-Request-URI=False
Route-By-To-URI=False
Drop-Routes-On-Ping-Timeout=False
Use-External-XML=False
External-XML-File=b2bua-route.xml

[Media-Server]
Enable-Media-Server=False
Media-Server-Number=5000
Codec-List Array Size=0
No-RTP-Proxy-On-All-Transfers=False
Enable-Announcement-Service=False
4xx-Error-Map=prompts/basic/cant_complete.wav
5xx-Error-Map=prompts/basic/cant_complete.wav
6xx-Error-Map=prompts/basic/cant_complete.wav
Announcement-Error-Map Array Size=0

[Outbound-Proxies]
Outbound-Proxies Array Size=0

[Local-Domain-Accounts]
Accept-All-Registration=False
Account-List Array Size=0

Thanks guys,
  Mike

Re: [OpenSBC] Calls to outside with SRV record addresses

[OpenSBC F. <ope...@op...>]

Success!


 Thanks Joegen!

Re: [OpenSBC] Calls to outside with SRV record addresses

[OpenSBC F. <ope...@op...>]

oops, hold on...  looks like a new dependency from opensipstack...  i'll cvs that and recompile that too...

Re: [OpenSBC] Calls to outside with SRV record addresses

[OpenSBC F. <ope...@op...>]

Looks like some other issue got introduced?  Or do I need to do something different to compile in the same place again?

fw1:/usr/src/opensbc# make bothnoshared
make optnoshared debugnoshared
make[1]: Entering directory `/usr/src/opensbc'
make P_SHAREDLIB=0 opt
make[2]: Entering directory `/usr/src/opensbc'
g++ -D_REENTRANT -D_REENTRANT -Wall  -I/usr/src/opensipstack/include -DPTRACING -I/usr/src/opensipstack/include -I ./ -fpermissive -Os    -c SBCIVRHandler.cxx -o obj_linux_x86_r/SBCIVRHandler.o
SBCIVRHandler.cxx: In member function 'virtual void SBCIVRHandler::RefreshSupportedCodecs()':
SBCIVRHandler.cxx:188: error: 'OpalG729_RAW' was not declared in this scope
make[2]: *** [http://obj_linux_x86_r/SBCIVRHandler.o|http://obj_linux_x86_r/SBCIVRHandler.o] Error 1
make[2]: Leaving directory `/usr/src/opensbc'
make[1]: *** [optnoshared] Error 2
make[1]: Leaving directory `/usr/src/opensbc'
make: *** [bothnoshared] Error 2

 
Thanks,
   Mike

Re: [OpenSBC] Calls to outside with SRV record addresses

[OpenSBC F. <ope...@op...>]

Thanks Joegen.


 I'll download current CVS, recompile & test.


Mike

Re: [OpenSBC] Getting closer with Vyatta and OpenSBC...

[OpenSBC F. <ope...@op...>]

Hi Joegen,


 Haven't tried registering from the outside yet.  I've been working on basic inbound / outbound calling so far.


 I'll keep you & the list posted.


 Thanks,


   Mike

Re: [OpenSBC] OpenSBC Routing

[OpenSBC F. <ope...@op...>]

It appears that your calls to Topex would fail because you created a B2BUA route that sends any call with a 0 prefix to 192.168.1.10.    I assume that the 405 error is coming from Topex when the call hairpins back to it.

When OpenSBC is in B2B Only mode, it will prioritize explicitly defined B2BUA routes over the local registrar.  One way to fix this would be to use a prefix like 999 to route calls to Topex, this way OpenSBC will be able to distinguish between inbound calls to a locally registered UA and outbound calls meant to be routed to Topex UA.

The various modes of operation are as follows:

----------------------------------

*Full Mode* - By default OpenSBC runs in full mode exposing its capability both as a
relay SIP proxy, Registrar and as a B2B User Agent. When OpenSBC receives an INVITE
or a REGISTER request it would follow the following procedure to make a decision how
to route a request:

● If the Request-URI resolves to a remote domain, the request will be relayed. If a
relay route is available, the request is sent to that route. If a relay route is not
available, then the URI is resolved via DNS.
● If the Startline-URI resolves as a local address and port, the To URI is checked
if it resolves to a local domain and port. If not, the request would be proxied
using Relay Routes or via DNS resolution. The Request URI would be rewritten to point
to the resolved route.
● INVITE: If both Request URI and To URI resolves to a local listener and port, the
B2BUA Route is used to route the INVITE.
● REGISTER: If both Request URI and To URI resolves to a local listener and port, the
local Registrar will process the registration. This would include Authorization of
the user.

*B2BOnly Mode* - This mode removes the relay capability but exposes the Registrar and
the B2BUA functionalities. This mode does not do the checks performed by Full Mode. It
will always process REGISTER and INVITE as local.

● INVITE: This mode always use B2BUA Route to route calls. If there is not corresponding
route found, a DNS resolutions is done against the Request URI or the To URI in case the
Request-URI resolves to a local address.
● REGISTER: Registrations are always handled by the local registrar.

*Proxy Only Mode* - This mode removes the B2BUA functionality but exposes Registrar and the
relay SIP Proxy functionalities
● Always uses Relay Routes for all messages including REGISTER. If a relay route is not
configured, Requests will be relayed using DNS resolution. If a registrations is resolved
as local, the registrar would handle the registration including authorization

*B2BUpperReg Mod*e - This is almost the same as the B2BOnly mode but with the additional
capability of relaying registrations to upper registrars.

● INVITE: This mode always uses B2BUA Route.
● REGISTER: For registrations, it performs the Request URI and To URI checking and relay
for a remote domain or process the registration locally for local domains.
● Upper-Registration: This mode also has the capability to hijack-registrations towards
upstream registrars.

Re: [OpenSBC] Is this application feasible?

[OpenSBC F. <ope...@op...>]

Hi Paolo,

Yes, OpenSBC can be configured to register to different trunk providers.  OpenSBC can also fork calls if you register multiple phones using the same AOR.

Joegen

> {quote:title=Paolo wrote:}{quote}
> 
> Hi all, I am quite new to OpenSBC and I don't really understand if what I would like to do with it is really feasible.
> 
> 
> Currently all our users register to a provider's registrar and use its proxy.
> 
> 
> I would like to have the users register to OpenSBC, then OpenSBC converts its own logins to remote logins,
> 
> 
> so when the users are calling out they effectively direct their calls through the provider's proxy ( in fact, depending
> 
> 
> on destination prefix,  we would like to be able to choose between many providers) using their own login, and when
> 
> 
> receiving calls from the proxy they should operate in the same way as effectively registered directly.
> 
> 
> The purpose of all this thing is solving some incompatibilities in timeouts between UA and proxy, if registrations
> 
> 
> are done too quickly they lock up for many minutes, so OpenSBC should keep registrations and give its own to
> 
> 
> UAs powering on and off;  then we would like to have many UAs receiving calls at the same time for the same
> 
> 
> incoming number, the first one answers, the others then stop ringing.
> 
> 
> Thanks for your help. 
> 
>

Re: [OpenSBC] OpenSBC

[Joegen E. B. <joe...@gm...>]

inline....

Tony Turner wrote:
>
> Hi Guys,
>
>  
>
> I need an SBC to register SIP clients and send all traffic starting 0 
> to a trunk. It must only send traffic to the trunk for registered 
> users only.
>
>  
>
> I have downloaded OpenSBC MSI and installed on a Windows 2003 server 
> with two interface cards.
>
>  
>
> I have OpenSBC set up in B2BOnly Mode
>
>  
>
> I can register two SIP clients.
>
> sip:20...@si...:5060 sip:192.168.1.89:7055 3260
> sip:10...@si... sip:100@192.168.1.67:5060 2990
>
>  
>
> I have a trunk which registers:
>
> sip:08453870000@REMOVED sip:08453870000@REMOVED:5066 
> f95953a8-56fd-1810-91ab-a931058ef8ec@212.13.196.40 
> <mailto:f95953a8-56fd-1810-91ab-a931058ef8ec@212.13.196.40> SIP/2.0 200 OK
>
>  
>
> Trunk Config
>
>  
>
> <root>
>
> <siptrunk trunk-name="trunk.MobileSqueeze"
>
> route-set="sip:REMOVED"
>
> sip-domain=" REMOVED"
>
> expires="10">
>
> <trunk-accounts>
>
> <account user-name="08453870000"
>
> auth-user-name="08453870000"
>
> auth-password=" REMOVED"
>
> inbound-route="sip:10...@si..."
>
> expires="3600"/>
>
> </trunk-accounts>
>
> <transient-accounts>
>
> <account user-name="08453870000"
>
> auth-user-name="08453870000"
>
> auth-password=" REMOVED"
>
> inbound-route="10...@si... 
> <mailto:10...@si...>"
>
> expires="3600"/>
>
> </transient-accounts>
>
> </siptrunk>
>
> </root> 
>
>  
>
> Each client can call each other.
>
>  
>
> I use an external route file route.xml
>
>  
>
> <routes>
>
> <route filter="sip:100*"></route>
>
> <route filter="sip:200*"></route>
>
> </routes>
>
>  
>
> How do I route all calls starting 0 to the registered trunk?
>

<route filter="sip:0* 
<sip:100*>">sip:yoursiptrunkdomain.com;sip-trunk=true</route>

> Is my trunk config correct?
>
> I cannot find any logs on the Windows server, I log SIP-Log-Level 5, 
> do I have to start the service different to get it to log?
>

Log is in Application Data/OpenSIPStack/OpenSBCData


> Is B2BOnly Mode the correct mode for using OpenSBC as registrar and 
> send traffic for "registered users only" to the trunk?
>

Yes
>
>  
>
> I can compile OpenSBC from source just wanted to get it working first.
>
>  
>
> Thank you
>
>  
>
> Tony
>
>  
>

Re: [OpenSBC] OpenSBC Routing

[OpenSBC F. <ope...@op...>]

Hi Tony,

Sorry for the late reply.  You have to send the level 5 log for the none working call from your Topex UA towards the registered UA so we can determine what's causing the 405 reponse.  You may mail it to joegen @ opensipstack.org as an attachment.

Joegen

> {quote:title=telswitch wrote:}{quote}
> Hi Guys
> 
> I don't quite understand the routing of OpenSBC.
> 
> IP Calls to TOPEX IP - works great
> Calls from 1 Client to another work Great via OpenSBC
> Calls from Topex IP to OpenSBC registered client I get 405 not allowed
> 
> My Topex does not register on the OpenSBC. 
> 
> 
> How can I allow all calls from 192.168.1.10 to any registered clients on the OpenSBC.
> 
> 
> Private Interface on Topex is 192.168.1.10
> Private Interface on OpenSBC is 192.168.1.11
> Public IP on OpenSBC sip.mobilesqueeze.co.uk
> 
> 
> My setup is like this:
> 
> 
> Works Great
> SIP Clients via OpenSBC to [TOPEX IP GW] 
> 
> Topex to OpenSBC - can't get route to work to work 405 not allowed
> SS7 to [Topex IP GW] to [OpenSBC] to SIP Clients
> 
> 
> Setup B2B Only mode
> Allow all calls - un ticked
> 
> Local accounts
> sip:02033902000:xx...@si...
> sip:02033902001:xx...@si...
> 
> Routing
> --sip:0*@sip.mobilesqueeze.co.uk-- sip:192.168.1.10 (TOPEX) 
> 
> If I dial from a SIP client an 0xxxx number if goes into the TOPEX - 192.168.1.10 - great
> 
> 
> But if i send a call from my mobile to 02033902000, my topex it sends the call to the to the registered client on OpenSBC but i get 405 method not allowed 
> 
> 
> Thanks
> 
> 
> Tony

Re: [OpenSBC] Getting closer with Vyatta and OpenSBC...

[OpenSBC F. <ope...@op...>]

> {quote:title=mpicher wrote:}{quote}
> 
> I am getting calls to route in at this point but audio quality doesn't seem great.  Still tweaking though.  Also, I can't seem to register to other things like my sipphone.com account through the firewall now with OpenSBC installed on it.  I'm sure it's just config stuff I need to get right though.


Mike, you seem to have indicated in your latest post that you got things working for inbound.  Does that mean you got your phones to register as well?

Re: [OpenSBC] Calls to outside with SRV record addresses

[OpenSBC F. <ope...@op...>]

Hi Mike,

Indeed.  This is a confirmed bug.  OpenSBC currently prioritizes A records over DNS SRV.  Since sipphone.com is also an A record, OpenSBC uses that instead of the DNS/SRV record.  I have applied a patch for this in CVS.  Thanks for pointing it out.

Joegen


> {quote:title=mpicher wrote:}{quote}
> Alright, got inbound / outbound working.
> 
> One problem with outbound calling however, if I try calling a location by SRV record it doesn't want to complete the call but if I call to an A record the call goes through.
> 
> For example...  if i dial out to mp...@si... the call does not complete (see log below)
> 
> But if I try mp...@pr... the call will complete (see log below).
> 
> Any thoughts?
> 
> My only other problem (well at the moment, and with opensbc) is calls to the AutoAttendant on sipX seem really scratchy, whereas call into my UA have good call quality.
> 
> Thanks,
>    Mike
> 
> Here's a log of the call to mp...@si... not working:
> 
> 2009/01/01 12:40:43.144 INF: [CID=0xa651360e] <<< INVITE sip:mp...@si... SIP/2.0 Method(INVITE)  SRC: 172.16.1.2:5060:UDP enc=0 bytes=1760
> 2009/01/01 12:40:43.148 INF: [CID=0xa651360e] >>> SIP/2.0 100 Trying Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=756
> 2009/01/01 12:40:43.150 INF: [CID=0xa651360e] *** CREATED (UAS) CALL *** NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.
> 2009/01/01 12:40:43.152 INF: [CID=0xa651360e] *** NO REGISTRATION FOUND *** Fetching route through local database URI sip:mp...@si...
> 2009/01/01 12:40:43.152 INF: [CID=0xa651360e] *** NO STATIC ROUTE *** URI sip:mp...@si...
> 2009/01/01 12:40:43.153 INF: [CID=0xa651360e] *** UPPER REGISTRATION RELAY *** -->> Callee: sip:MP...@xy... AOR: sip:mp...@si... BINDING: sip:mp...@si...
> 2009/01/01 12:40:43.156 INF: [CID=0xa651360e] *** CREATED (UAC) CALL *** NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.-0x0003
> 2009/01/01 12:40:43.158 INF: [CID=0xa651360e] *** RTP Session CREATED *** l-addr=172.16.1.254 r-addr=172.16.1.129/172.16.1.129 r-port=50334
> 2009/01/01 12:40:43.169 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
> 2009/01/01 12:40:43.669 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
> 2009/01/01 12:40:44.669 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
> 2009/01/01 12:40:46.669 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
> 2009/01/01 12:40:50.669 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
> 2009/01/01 12:40:58.669 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
> 2009/01/01 12:41:13.169 WRN: [CID=0x99b5a101] *** TIMER EXPIRATION *** for SIP Session NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.-0x0003
> 2009/01/01 12:41:13.170 ERR: [CID=0xa651360e] *** ALERTING/CONNECT TIMEOUT!!! ***
> 2009/01/01 12:41:13.170 INF: [CID=0xa651360e] *** RTP (Audio) Statistics *** addr=76.178.252.229:30010->0.0.0.0:0 enc=0 rx=0 tx=0 lost=0 outOfOrder=0 late=0 rxTime=0/0 txTime=0/0 jitter=0/0
> 2009/01/01 12:41:13.171 INF: [CID=0xa651360e] Connection: Rejected / Code = 408
> 2009/01/01 12:41:13.175 INF: [CID=0xa651360e] *** CALL TEAR DOWN *** NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.-connection
> 2009/01/01 12:41:13.175 INF: [CID=0x8428bd16] *** B2BUA CONNECTION OnDestroySession ***
> 2009/01/01 12:41:13.175 INF: [CID=0xa651360e] *** RTP (Audio) Statistics *** addr=172.16.1.254:30008->0.0.0.0:0 enc=0 rx=0 tx=0 lost=0 outOfOrder=0 late=0 rxTime=0/0 txTime=0/0 jitter=0/0
> 2009/01/01 12:41:13.176 INF: [CID=0xa651360e] *** DESTROYED CALL *** NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.-0x0003
> 2009/01/01 12:41:13.178 INF: [CID=0xa651360e] >>> CANCEL sip:mp...@si... SIP/2.0 Method(CANCEL) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=387
> 2009/01/01 12:41:13.181 INF: [CID=0xa651360e] >>> SIP/2.0 408 Alerting Timeout Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=839
> 2009/01/01 12:41:13.192 INF: [CID=0xa651360e] <<< ACK sip:mp...@si... SIP/2.0 Method(ACK)  SRC: 172.16.1.2:5060:UDP enc=0 bytes=471
> 2009/01/01 12:41:13.668 INF: [CID=0xa651360e] >>> CANCEL sip:mp...@si... SIP/2.0 Method(CANCEL) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=387
> 2009/01/01 12:41:14.668 INF: [CID=0xa651360e] >>> CANCEL sip:mp...@si... SIP/2.0 Method(CANCEL) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=387
> 2009/01/01 12:41:16.668 INF: [CID=0xa651360e] >>> CANCEL sip:mp...@si... SIP/2.0 Method(CANCEL) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=387
> 2009/01/01 12:41:17.177 INF: [CID=0x8428bd16] *** DESTROYED B2BUA CONNECTION *** 0x0x887d130
> 2009/01/01 12:41:17.178 INF: [CID=0x8428bd16] *** COUNTERS *** ICT=0 NICT=1 IST=1 NIST=0 TIMERS=7 CALL=0 CONN=0 REG=0 RTP=0 QUEUE=1 CACHE=2 GC=4 TOTALCONN=3 TOTALREG=0 TOTALSZR=1 AVGDUR=3385 HIGHDUR=3385 IDLETIME=34
> 2009/01/01 12:41:17.178 INF: [CID=0xa651360e] *** DESTROYED CALL *** NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.
> 2009/01/01 12:41:20.668 INF: [CID=0xa651360e] >>> CANCEL sip:mp...@si... SIP/2.0 Method(CANCEL) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=387
> 
> 
> 
> 
> Here's a log of the call to mp...@pr... working:
> 
> 2009/01/01 12:45:17.906 INF: [CID=0x78099be1] <<< INVITE sip:mp...@pr... SIP/2.0 Method(INVITE)  SRC: 172.16.1.2:5060:UDP enc=0 bytes=1626
> 2009/01/01 12:45:17.910 INF: [CID=0x78099be1] *** CREATED (UAS) CALL *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.
> 2009/01/01 12:45:17.911 INF: [CID=0x78099be1] *** NO REGISTRATION FOUND *** Fetching route through local database URI sip:mp...@pr...
> 2009/01/01 12:45:17.912 INF: [CID=0x78099be1] *** NO STATIC ROUTE *** URI sip:mp...@pr...
> 2009/01/01 12:45:17.913 INF: [CID=0x78099be1] *** UPPER REGISTRATION RELAY *** -->> Callee: sip:20...@xy... AOR: sip:mp...@pr... BINDING: sip:mp...@pr...
> 2009/01/01 12:45:17.915 INF: [CID=0x78099be1] *** CREATED (UAC) CALL *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.-0x0004
> 2009/01/01 12:45:17.918 INF: [CID=0x78099be1] *** RTP Session CREATED *** l-addr=172.16.1.254 r-addr=172.16.1.129/172.16.1.129 r-port=34390
> 2009/01/01 12:45:17.926 INF: [CID=0x78099be1] >>> SIP/2.0 100 Trying Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=606
> 2009/01/01 12:45:17.929 INF: [CID=0x78099be1] >>> INVITE sip:mp...@pr... SIP/2.0 Method(INVITE) DST: 198.65.166.131:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1359
> 2009/01/01 12:45:18.428 INF: [CID=0x78099be1] >>> INVITE sip:mp...@pr... SIP/2.0 Method(INVITE) DST: 198.65.166.131:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1359
> 2009/01/01 12:45:19.212 INF: [CID=0x78099be1] <<< SIP/2.0 100 Giving a try Method(INVITE)  SRC: 198.65.166.131:5060:UDP enc=0 bytes=382
> 2009/01/01 12:45:19.216 INF: [CID=0x78099be1] <<< SIP/2.0 180 Ringing Method(INVITE)  SRC: 198.65.166.131:5060:UDP enc=0 bytes=560
> 2009/01/01 12:45:19.226 INF: [CID=0x78099be1] >>> SIP/2.0 180 Ringing Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=695
> 2009/01/01 12:45:20.429 INF: [CID=0x78099be1] <<< SIP/2.0 180 Ringing Method(INVITE)  SRC: 198.65.166.131:5060:UDP enc=0 bytes=560
> 2009/01/01 12:45:20.436 INF: [CID=0x78099be1] >>> SIP/2.0 180 Ringing Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=695
> 2009/01/01 12:45:20.881 INF: [CID=0x78099be1] <<< SIP/2.0 200 OK Method(INVITE)  SRC: 198.65.166.131:5060:UDP enc=0 bytes=855
> 2009/01/01 12:45:20.886 INF: [CID=0x78099be1] *** RTP Session CREATED *** l-addr=76.178.252.229 r-addr=130.94.88.90/76.178.252.229 r-port=16416
> 2009/01/01 12:45:20.886 INF: [CID=0x78099be1] *** CALL ESTABLISHED *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.-0x0004
> 2009/01/01 12:45:20.890 INF: [CID=0x78099be1] *** RTP PROXY STARTED ***
> 2009/01/01 12:45:20.892 INF: [CID=0x78099be1] >>> ACK sip:hostedvm17476040590@130.94.88.90 SIP/2.0 Method(ACK) DST: 198.65.166.131:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=667
> 2009/01/01 12:45:20.893 INF: [CID=0x78099be1] *** RTP PROXY STARTED ***
> 2009/01/01 12:45:20.898 INF: [CID=0x78099be1] >>> SIP/2.0 200 OK Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=1133
> 2009/01/01 12:45:21.092 INF: [CID=0x78099be1] <<< ACK sip:mpicher@172.16.1.254:5060 SIP/2.0 Method(ACK)  SRC: 172.16.1.2:5060:UDP enc=0 bytes=895
> 2009/01/01 12:45:26.515 INF: [CID=0x78099be1] <<< BYE sip:mpicher@172.16.1.254:5060 SIP/2.0 Method(BYE)  SRC: 172.16.1.2:5060:UDP enc=0 bytes=932
> 2009/01/01 12:45:26.526 INF: [CID=0x78099be1] *** CALL TEAR DOWN *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.-connection
> 2009/01/01 12:45:26.526 INF: [CID=0x0cfee90a] *** B2BUA CONNECTION OnDestroySession ***
> 2009/01/01 12:45:26.527 INF: [CID=0x78099be1] *** RTP (Audio) Statistics *** addr=172.16.1.254:30012->172.16.1.129:34390 enc=0 rx=275 tx=269 lost=0 outOfOrder=0 late=0 rxTime=20/27 txTime=20/42 jitter=0/1
> 2009/01/01 12:45:26.529 INF: [CID=0x78099be1] >>> SIP/2.0 200 OK Method(BYE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=526
> 2009/01/01 12:45:26.531 INF: [CID=0x78099be1] >>> BYE sip:hostedvm17476040590@130.94.88.90 SIP/2.0 Method(BYE) DST: 198.65.166.131:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=722
> 2009/01/01 12:45:26.533 INF: [CID=0x78099be1] *** RTP PROXY STOPPED ***
> 2009/01/01 12:45:26.533 INF: [CID=0x78099be1] *** RTP PROXY STOPPED ***
> 2009/01/01 12:45:26.539 INF: [CID=0x78099be1] *** RTP (Audio) Statistics *** addr=76.178.252.229:30014->130.94.88.90:16416 enc=0 rx=269 tx=275 lost=7 outOfOrder=0 late=0 rxTime=19/22 txTime=20/27 jitter=1/5
> 2009/01/01 12:45:26.658 INF: [CID=0x78099be1] <<< SIP/2.0 200 OK Method(BYE)  SRC: 198.65.166.131:5060:UDP enc=0 bytes=608
> 2009/01/01 12:45:31.541 INF: [CID=0x0cfee90a] *** DESTROYED B2BUA CONNECTION *** 0x0x886d4b0
> 2009/01/01 12:45:31.541 INF: [CID=0x0cfee90a] *** COUNTERS *** ICT=1 NICT=1 IST=0 NIST=1 TIMERS=12 CALL=0 CONN=0 REG=0 RTP=0 QUEUE=1 CACHE=2 GC=6 TOTALCONN=4 TOTALREG=0 TOTALSZR=2 AVGDUR=4507 HIGHDUR=5630 IDLETIME=14
> 2009/01/01 12:45:31.542 INF: [CID=0x78099be1] *** DESTROYED CALL *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.
> 2009/01/01 12:45:31.543 INF: [CID=0x78099be1] *** DESTROYED CALL *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.-0x0004

[OpenSBC] Calls to outside with SRV record addresses

[OpenSBC F. <ope...@op...>]

Alright, got inbound / outbound working.

One problem with outbound calling however, if I try calling a location by SRV record it doesn't want to complete the call but if I call to an A record the call goes through.

For example...  if i dial out to mp...@si... the call does not complete (see log below)

But if I try mp...@pr... the call will complete (see log below).

Any thoughts?

My only other problem (well at the moment, and with opensbc) is calls to the AutoAttendant on sipX seem really scratchy, whereas call into my UA have good call quality.

Thanks,
   Mike

Here's a log of the call to mp...@si... not working:

2009/01/01 12:40:43.144 INF: [CID=0xa651360e] <<< INVITE sip:mp...@si... SIP/2.0 Method(INVITE)  SRC: 172.16.1.2:5060:UDP enc=0 bytes=1760
2009/01/01 12:40:43.148 INF: [CID=0xa651360e] >>> SIP/2.0 100 Trying Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=756
2009/01/01 12:40:43.150 INF: [CID=0xa651360e] *** CREATED (UAS) CALL *** NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.
2009/01/01 12:40:43.152 INF: [CID=0xa651360e] *** NO REGISTRATION FOUND *** Fetching route through local database URI sip:mp...@si...
2009/01/01 12:40:43.152 INF: [CID=0xa651360e] *** NO STATIC ROUTE *** URI sip:mp...@si...
2009/01/01 12:40:43.153 INF: [CID=0xa651360e] *** UPPER REGISTRATION RELAY *** -->> Callee: sip:MP...@xy... AOR: sip:mp...@si... BINDING: sip:mp...@si...
2009/01/01 12:40:43.156 INF: [CID=0xa651360e] *** CREATED (UAC) CALL *** NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.-0x0003
2009/01/01 12:40:43.158 INF: [CID=0xa651360e] *** RTP Session CREATED *** l-addr=172.16.1.254 r-addr=172.16.1.129/172.16.1.129 r-port=50334
2009/01/01 12:40:43.169 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
2009/01/01 12:40:43.669 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
2009/01/01 12:40:44.669 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
2009/01/01 12:40:46.669 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
2009/01/01 12:40:50.669 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
2009/01/01 12:40:58.669 INF: [CID=0xa651360e] >>> INVITE sip:mp...@si... SIP/2.0 Method(INVITE) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1320
2009/01/01 12:41:13.169 WRN: [CID=0x99b5a101] *** TIMER EXPIRATION *** for SIP Session NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.-0x0003
2009/01/01 12:41:13.170 ERR: [CID=0xa651360e] *** ALERTING/CONNECT TIMEOUT!!! ***
2009/01/01 12:41:13.170 INF: [CID=0xa651360e] *** RTP (Audio) Statistics *** addr=76.178.252.229:30010->0.0.0.0:0 enc=0 rx=0 tx=0 lost=0 outOfOrder=0 late=0 rxTime=0/0 txTime=0/0 jitter=0/0
2009/01/01 12:41:13.171 INF: [CID=0xa651360e] Connection: Rejected / Code = 408
2009/01/01 12:41:13.175 INF: [CID=0xa651360e] *** CALL TEAR DOWN *** NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.-connection
2009/01/01 12:41:13.175 INF: [CID=0x8428bd16] *** B2BUA CONNECTION OnDestroySession ***
2009/01/01 12:41:13.175 INF: [CID=0xa651360e] *** RTP (Audio) Statistics *** addr=172.16.1.254:30008->0.0.0.0:0 enc=0 rx=0 tx=0 lost=0 outOfOrder=0 late=0 rxTime=0/0 txTime=0/0 jitter=0/0
2009/01/01 12:41:13.176 INF: [CID=0xa651360e] *** DESTROYED CALL *** NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.-0x0003
2009/01/01 12:41:13.178 INF: [CID=0xa651360e] >>> CANCEL sip:mp...@si... SIP/2.0 Method(CANCEL) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=387
2009/01/01 12:41:13.181 INF: [CID=0xa651360e] >>> SIP/2.0 408 Alerting Timeout Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=839
2009/01/01 12:41:13.192 INF: [CID=0xa651360e] <<< ACK sip:mp...@si... SIP/2.0 Method(ACK)  SRC: 172.16.1.2:5060:UDP enc=0 bytes=471
2009/01/01 12:41:13.668 INF: [CID=0xa651360e] >>> CANCEL sip:mp...@si... SIP/2.0 Method(CANCEL) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=387
2009/01/01 12:41:14.668 INF: [CID=0xa651360e] >>> CANCEL sip:mp...@si... SIP/2.0 Method(CANCEL) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=387
2009/01/01 12:41:16.668 INF: [CID=0xa651360e] >>> CANCEL sip:mp...@si... SIP/2.0 Method(CANCEL) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=387
2009/01/01 12:41:17.177 INF: [CID=0x8428bd16] *** DESTROYED B2BUA CONNECTION *** 0x0x887d130
2009/01/01 12:41:17.178 INF: [CID=0x8428bd16] *** COUNTERS *** ICT=0 NICT=1 IST=1 NIST=0 TIMERS=7 CALL=0 CONN=0 REG=0 RTP=0 QUEUE=1 CACHE=2 GC=4 TOTALCONN=3 TOTALREG=0 TOTALSZR=1 AVGDUR=3385 HIGHDUR=3385 IDLETIME=34
2009/01/01 12:41:17.178 INF: [CID=0xa651360e] *** DESTROYED CALL *** NmRiYzY2MzUzOWE5ZDMyNmNiMWEzM2M0MzdjZDhlMTc.
2009/01/01 12:41:20.668 INF: [CID=0xa651360e] >>> CANCEL sip:mp...@si... SIP/2.0 Method(CANCEL) DST: 198.65.166.139:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=387




Here's a log of the call to mp...@pr... working:

2009/01/01 12:45:17.906 INF: [CID=0x78099be1] <<< INVITE sip:mp...@pr... SIP/2.0 Method(INVITE)  SRC: 172.16.1.2:5060:UDP enc=0 bytes=1626
2009/01/01 12:45:17.910 INF: [CID=0x78099be1] *** CREATED (UAS) CALL *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.
2009/01/01 12:45:17.911 INF: [CID=0x78099be1] *** NO REGISTRATION FOUND *** Fetching route through local database URI sip:mp...@pr...
2009/01/01 12:45:17.912 INF: [CID=0x78099be1] *** NO STATIC ROUTE *** URI sip:mp...@pr...
2009/01/01 12:45:17.913 INF: [CID=0x78099be1] *** UPPER REGISTRATION RELAY *** -->> Callee: sip:20...@xy... AOR: sip:mp...@pr... BINDING: sip:mp...@pr...
2009/01/01 12:45:17.915 INF: [CID=0x78099be1] *** CREATED (UAC) CALL *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.-0x0004
2009/01/01 12:45:17.918 INF: [CID=0x78099be1] *** RTP Session CREATED *** l-addr=172.16.1.254 r-addr=172.16.1.129/172.16.1.129 r-port=34390
2009/01/01 12:45:17.926 INF: [CID=0x78099be1] >>> SIP/2.0 100 Trying Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=606
2009/01/01 12:45:17.929 INF: [CID=0x78099be1] >>> INVITE sip:mp...@pr... SIP/2.0 Method(INVITE) DST: 198.65.166.131:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1359
2009/01/01 12:45:18.428 INF: [CID=0x78099be1] >>> INVITE sip:mp...@pr... SIP/2.0 Method(INVITE) DST: 198.65.166.131:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=1359
2009/01/01 12:45:19.212 INF: [CID=0x78099be1] <<< SIP/2.0 100 Giving a try Method(INVITE)  SRC: 198.65.166.131:5060:UDP enc=0 bytes=382
2009/01/01 12:45:19.216 INF: [CID=0x78099be1] <<< SIP/2.0 180 Ringing Method(INVITE)  SRC: 198.65.166.131:5060:UDP enc=0 bytes=560
2009/01/01 12:45:19.226 INF: [CID=0x78099be1] >>> SIP/2.0 180 Ringing Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=695
2009/01/01 12:45:20.429 INF: [CID=0x78099be1] <<< SIP/2.0 180 Ringing Method(INVITE)  SRC: 198.65.166.131:5060:UDP enc=0 bytes=560
2009/01/01 12:45:20.436 INF: [CID=0x78099be1] >>> SIP/2.0 180 Ringing Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=695
2009/01/01 12:45:20.881 INF: [CID=0x78099be1] <<< SIP/2.0 200 OK Method(INVITE)  SRC: 198.65.166.131:5060:UDP enc=0 bytes=855
2009/01/01 12:45:20.886 INF: [CID=0x78099be1] *** RTP Session CREATED *** l-addr=76.178.252.229 r-addr=130.94.88.90/76.178.252.229 r-port=16416
2009/01/01 12:45:20.886 INF: [CID=0x78099be1] *** CALL ESTABLISHED *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.-0x0004
2009/01/01 12:45:20.890 INF: [CID=0x78099be1] *** RTP PROXY STARTED ***
2009/01/01 12:45:20.892 INF: [CID=0x78099be1] >>> ACK sip:hostedvm17476040590@130.94.88.90 SIP/2.0 Method(ACK) DST: 198.65.166.131:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=667
2009/01/01 12:45:20.893 INF: [CID=0x78099be1] *** RTP PROXY STARTED ***
2009/01/01 12:45:20.898 INF: [CID=0x78099be1] >>> SIP/2.0 200 OK Method(INVITE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=1133
2009/01/01 12:45:21.092 INF: [CID=0x78099be1] <<< ACK sip:mpicher@172.16.1.254:5060 SIP/2.0 Method(ACK)  SRC: 172.16.1.2:5060:UDP enc=0 bytes=895
2009/01/01 12:45:26.515 INF: [CID=0x78099be1] <<< BYE sip:mpicher@172.16.1.254:5060 SIP/2.0 Method(BYE)  SRC: 172.16.1.2:5060:UDP enc=0 bytes=932
2009/01/01 12:45:26.526 INF: [CID=0x78099be1] *** CALL TEAR DOWN *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.-connection
2009/01/01 12:45:26.526 INF: [CID=0x0cfee90a] *** B2BUA CONNECTION OnDestroySession ***
2009/01/01 12:45:26.527 INF: [CID=0x78099be1] *** RTP (Audio) Statistics *** addr=172.16.1.254:30012->172.16.1.129:34390 enc=0 rx=275 tx=269 lost=0 outOfOrder=0 late=0 rxTime=20/27 txTime=20/42 jitter=0/1
2009/01/01 12:45:26.529 INF: [CID=0x78099be1] >>> SIP/2.0 200 OK Method(BYE) DST: 172.16.1.2:5060:UDP SRC=172.16.1.254:5061 enc=0 bytes=526
2009/01/01 12:45:26.531 INF: [CID=0x78099be1] >>> BYE sip:hostedvm17476040590@130.94.88.90 SIP/2.0 Method(BYE) DST: 198.65.166.131:5060:UDP  SRC: 76.178.252.229:5060 enc=0 bytes=722
2009/01/01 12:45:26.533 INF: [CID=0x78099be1] *** RTP PROXY STOPPED ***
2009/01/01 12:45:26.533 INF: [CID=0x78099be1] *** RTP PROXY STOPPED ***
2009/01/01 12:45:26.539 INF: [CID=0x78099be1] *** RTP (Audio) Statistics *** addr=76.178.252.229:30014->130.94.88.90:16416 enc=0 rx=269 tx=275 lost=7 outOfOrder=0 late=0 rxTime=19/22 txTime=20/27 jitter=1/5
2009/01/01 12:45:26.658 INF: [CID=0x78099be1] <<< SIP/2.0 200 OK Method(BYE)  SRC: 198.65.166.131:5060:UDP enc=0 bytes=608
2009/01/01 12:45:31.541 INF: [CID=0x0cfee90a] *** DESTROYED B2BUA CONNECTION *** 0x0x886d4b0
2009/01/01 12:45:31.541 INF: [CID=0x0cfee90a] *** COUNTERS *** ICT=1 NICT=1 IST=0 NIST=1 TIMERS=12 CALL=0 CONN=0 REG=0 RTP=0 QUEUE=1 CACHE=2 GC=6 TOTALCONN=4 TOTALREG=0 TOTALSZR=2 AVGDUR=4507 HIGHDUR=5630 IDLETIME=14
2009/01/01 12:45:31.542 INF: [CID=0x78099be1] *** DESTROYED CALL *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.
2009/01/01 12:45:31.543 INF: [CID=0x78099be1] *** DESTROYED CALL *** YTU5MzRmN2RlMWUyYTU2ZTRmNTY2NThmNDgwMWNlZmU.-0x0004

Re: [OpenSBC] Getting closer with Vyatta and OpenSBC...

[OpenSBC F. <ope...@op...>]

I Joegen knows if he waits long enough we'll figure this stuff out ourselves...  I wasn't getting core dumps...

For other Linux n00bs here's what I did:

Login as root.

Copy Executables to a more acceptable location.

  cp /usr/src/opensbc/distrib/* /usr/local/bin


Fix Shell Scripts

  Edit /usr/local/bin/startup.sh  (i use 'nano -w /usr/local/bin/startup.sh')

  Modify the startup command to: ./opensbc -d -p /var/run/opensbc.pid -H 65536 -C 1024000

  Edit /usr/local/bin/shutdown.sh

  Modify the shutdown command to: ./opensbc -k -p /var/run/opensbc.pid


Log / Ini Files

  If you are logged in as root, logging and ini files are in: /root/OpenSIPStack/OpenSBC_data


I am getting calls to route in at this point but audio quality doesn't seem great.  Still tweaking though.  Also, I can't seem to register to other things like my sipphone.com account through the firewall now with OpenSBC installed on it.  I'm sure it's just config stuff I need to get right though.

[OpenSBC] OpenSBC Routing

[OpenSBC F. <ope...@op...>]

Hi Guys


I don't quite understand the routing of OpenSBC.


IP Calls to TOPEX IP - works great
Calls from 1 Client to another work Great via OpenSBC
Calls from Topex IP to OpenSBC registered client I get 405 not allowed


My Topex does not register on the OpenSBC. 


How can I allow all calls from 192.168.1.10 to any registered clients on the OpenSBC.


Private Interface on Topex is 192.168.1.10
Private Interface on OpenSBC is 192.168.1.11
Public IP on OpenSBC sip.mobilesqueeze.co.uk


 My setup is like this:


Works Great
INTERNET ------Public Address/Private Address ----------------------- Private Address 
SIP Clients ---------- OpenSBC ------------------------------------------------[TOPEX IP]  ---- SS7

405 not allowed
Topex to OpenSBC - can't get route to work
SS7 --- [TOPEX IP] -----   OpenSBC-------- SIP Clients


Setup B2B Only mode


Local accounts
sip:02033902000:xx...@si...
sip:02033902001:xx...@si...

Routing
[sip:0*@sip.mobilesqueeze.co.uk] sip:192.168.1.10 (TOPEX)
[02033902000*]


If I dial from a SIP client an 0xxxx number if goes into the TOPEX -  192.168.1.10 - great


But if i send a call from my mobile to 02033902000,  my  [http://TOPEX - 192.168.1.10|http://TOPEX - 192.168.1.10] sends the call to the to the regsietered OpenSBC but i get 405 mothod not allowed 


Thanks


Tony


 


 


 


 


 


 


 


 


 


 

[OpenSBC] Getting closer with Vyatta and OpenSBC...

[OpenSBC F. <ope...@op...>]

Core dumps?

Not sure if I'm supposed to move the distrib folder somewhere or what...  the oss-application.conf.xml file is never getting updated so I'm not sure where the real one lives.

Looks like I'm getting core dumps as soon as I try to call in to the system.

This is from my distrib folder...

fw1:/usr/src/opensbc/distrib# ./startup.sh
Core file size set to 1024000/4294967295
Daemon started with pid 6826
fw1:/usr/src/opensbc/distrib# ls
core.20847  core.6706  oss-application.conf.xml      pidfile      startup.sh
core.5249   opensbc    oss-application.conf.xml.old  shutdown.sh
fw1:/usr/src/opensbc/distrib# ls -l
total 10556
-rw------- 1 root src  8171520 Dec 29 22:09 core.20847
-rw------- 1 root src  8310784 Dec 31 09:04 core.5249
-rw------- 1 root src 11620352 Dec 31 09:59 core.6706
-rw------- 1 root src 11751424 Dec 31 10:01 core.6826
-rwxr-xr-x 1 root src  6105825 Dec 29 12:22 opensbc
-rw-r--r-- 1 root src    33164 Dec 29 12:28 oss-application.conf.xml
-rw-r--r-- 1 root src    33153 Dec 29 12:27 oss-application.conf.xml.old
-rw-r--r-- 1 root src        5 Dec 31 10:00 pidfile
-rwxr-xr-x 1 root src       26 Dec 29 12:22 shutdown.sh
-rwxr-xr-x 1 root src       46 Dec 29 12:22 startup.sh
fw1:/usr/src/opensbc/distrib# ./shutdown.sh
Could not stop process 6826 - No such process


Also, not sure where the application is supposed to be logging to.

I'm getting pretty close to a properly documented SBC setup on a proper firewall.  Any help will be appreciated!

[OpenSBC] Is this application feasible?

[OpenSBC F. <ope...@op...>]

Hi all, I am quite new to OpenSBC and I don't really understand if what I would like to do with it is really feasible.


Currently all our users register to a provider's registrar and use its proxy.


I would like to have the users register to OpenSBC, then OpenSBC converts its own logins to remote logins,


so when the users are calling out they effectively direct their calls through the provider's proxy ( in fact, depending


on destination prefix,  we would like to be able to choose between many providers) using their own login, and when


receiving calls from the proxy they should operate in the same way as effectively registered directly.


The purpose of all this thing is solving some incompatibilities in timeouts between UA and proxy, if registrations


are done too quickly they lock up for many minutes, so OpenSBC should keep registrations and give its own to


UAs powering on and off;  then we would like to have many UAs receiving calls at the same time for the same


incoming number, the first one answers, the others then stop ringing.


Thanks for your help. 

Re: [OpenSBC] OpenSBC on Vyatta Firewall

[OpenSBC F. <ope...@op...>]

I'll try that.  Do I need to do it for both OpenSIPStack and OpenSBC?

Here's where I'm at with my instructions so far...

*Install OpenSBC on Vyatta Firewall*

Install Vyatta

  Boot from Vyatta LiveCD ISO.  Press ENTER.

  At login, username 'root', password 'vyatta'.

  Enter 'install-system'

  Configure as needed...  I'll share mine when I get this all working...


Get items required for building OpenSBC (kitchen sink approach):

  Login to Firewall as user 'vyatta'

  cd /etc/apt

  su

  Password:  (enter root password)

  nano -w sources.list

  Add line: "deb ftp://ftp.us.debian.org/debian/ lenny main contrib non-free"

  Ctrl-X and Y to overwrite
  
  FW1:/etc/apt# apt-get update

  FW1:/etc/apt# apt-get install -y mc autoconf automake cvs flex expat libexpat1-dev libtool build-essential libxml2 libxml2-dev libtiff4 libtiff4-dev php5 php5-cli php5-mysql php5 php5-cli php5-mysql php5-gd mysql-server libmysqlclient15-dev php-pear php-db curl sox apache2 libssl-dev libncurses5-dev bison libaudiofile-dev subversion libnewt-dev libcurl3-dev libnet-ssleay-perl openssl ssl-cert libauthen-pam-perl libio-pty-perl libmd5-perl libpg-perl libdbd-pg-perl php5-pgsql sqlite3 libsqlite3-dev openssl ssl-cert libapache2-mod-php5 php5-cli php5-common phpMyAdmin php5-mcrypt mcrypt phppgadmin apache2 libmcrypt-dev


Get OpenSipStack and OpenSBC from CVS:

  cd /usr/src

  FW1:/home/vyatta#cvs -d:pserver:ano...@op...:/cvsroot/opensipstack login

  FW1:/home/vyatta#cvs -z3 -d:pserver:ano...@op...:/cvsroot/opensipstack co -P opensipstack

  FW1:/home/vyatta#cvs -z3 -d:pserver:ano...@op...:/cvsroot/opensipstack co -P opensbc


Compile / Make OpenSipStack and OpebSBC:

  cd /usr/src/opensipstack

  chmod +x ./configure

  ./configure

  make bothnoshared

  cd ../opensbc

  chmod +x ./configure

  ./configure

  make bothnoshared

[OpenSBC] OpenSBC

[Tony T. <ton...@te...>]

Hi Guys,

 

I need an SBC to register SIP clients and send all traffic starting 0 to a
trunk. It must only send traffic to the trunk for registered users only.

 

I have downloaded OpenSBC MSI and installed on a Windows 2003 server with
two interface cards.

 

I have OpenSBC set up in B2BOnly Mode

 

I can register two SIP clients.

sip:20...@si...:5060 sip:192.168.1.89:7055 3260 
sip:10...@si... sip:100@192.168.1.67:5060 2990

 

I have a trunk which registers:

sip:08453870000@REMOVED sip:08453870000@REMOVED:5066
f95953a8-56fd-1810-91ab-a931058ef8ec@212.13.196.40 SIP/2.0 200 OK

 

Trunk Config

 

<root>

<siptrunk trunk-name="trunk.MobileSqueeze" 

route-set="sip:REMOVED" 

sip-domain=" REMOVED" 

expires="10">

<trunk-accounts>

<account user-name="08453870000" 

auth-user-name="08453870000" 

auth-password=" REMOVED" 

inbound-route="sip:10...@si..." 

expires="3600"/>

</trunk-accounts>

<transient-accounts>

<account user-name="08453870000" 

auth-user-name="08453870000" 

auth-password=" REMOVED" 

inbound-route="10...@si..." 

expires="3600"/>

</transient-accounts>

</siptrunk>

</root>  

 

Each client can call each other.

 

I use an external route file route.xml

 

<routes>

<route filter="sip:100*"></route>

<route filter="sip:200*"></route>

</routes>

 

How do I route all calls starting 0 to the registered trunk?

Is my trunk config correct?

I cannot find any logs on the Windows server, I log SIP-Log-Level 5, do I
have to start the service different to get it to log?

Is B2BOnly Mode the correct mode for using OpenSBC as registrar and send
traffic for "registered users only" to the trunk?

 

I can compile OpenSBC from source just wanted to get it working first.

 

Thank you

 

Tony

 

Re: [OpenSBC] OpenSBC on Vyatta Firewall

[OpenSBC F. <ope...@op...>]

Hi Mike,

Quickest way is to issue a "make distrib" for OpenSBC after compiling.  This would put the required scripts and the executable in the "distrib" folder.   The binary for opensbc should be on obj_linux_x86_r folder. 

HTH,

Joegen

> {quote:title=mpicher wrote:}{quote}
> Hi guys,
> 
> Working on getting OpenSBC installed on a Vyatta firewall ISO install (Debian...  www.vyatta.com).
> 
> I finally figured out how to get all the dependencies working and am documenting things as I go so I'll post this when I'm done.
> 
> OpenSipStack and OpenSBC look like they compiled (using g++).  Hard to tell from all the stuff flying by.
> 
> But from here I'm not sure exactly what to do.  Where's the executable?  How do I get it to run as a service?
> 
> Either I'm missing something, the docs are lacking or the compile didn't go as planned.
> 
> Thanks,
>    Mike

[OpenSBC] OpenSBC on Vyatta Firewall

[OpenSBC F. <ope...@op...>]

Hi guys,

Working on getting OpenSBC installed on a Vyatta firewall ISO install (Debian...  www.vyatta.com).

I finally figured out how to get all the dependencies working and am documenting things as I go so I'll post this when I'm done.

OpenSipStack and OpenSBC look like they compiled (using g++).  Hard to tell from all the stuff flying by.

But from here I'm not sure exactly what to do.  Where's the executable?  How do I get it to run as a service?

Either I'm missing something, the are lacking or the compile didn't go as planned.

Thanks,
   Mike

Re: [OpenSBC] Getting OpenSBC to Generate a "Ring" on 180 Ringing Messages

[OpenSBC F. <ope...@op...>]

Makes sense, thanks

Re: [OpenSBC] Getting OpenSBC to Generate a "Ring" on 180 Ringing Messages

[OpenSBC F. <ope...@op...>]

This is an inherent limitation when using local REFER in OSBC.   Since the call between PSTN and OSBC is already connected, the 180 alerting message is absorbed by OSBC.   There is currently no mechanism in OSBC to play a 'ringing' file, and the caller would hear silence while the call is being transferred.