December 2005 - sr-users

10 Jan '06

Hello, Im making tests and its not a LCR problem, its a problem from my GW2, when I use it for first option, it fails too, here you have the ngrep, ClientA --> Proxy --> GW2 (192.168.10.93) (192.168.10.91) (195.219.74.166) Regards The problem is that the BYE request will be handled by your LCR logic. The BYE request should be route in the loose_route block as it is an in-dialog request. Maybe the BYE sent from the gateway is not correct. Please post a ngrep dump (ngrep -t -W byline port 5060) regards klaus Pepe wrote: > Hello, > > Im configuring Openser with LCR module and Im having an extrange > behavior, I have 2 gateways, GW1(preference1) and GW2(preference2), > > GW1(pref.1) > / \ > ClientA --> OpenSer --> Client B > \ GW2 (pref.2) / > > > When I call from Client A to Client B using GW1, all works fine, its the > same when hang up Client B or Client A, but when GW1 fail(I provoke it > changing codec) and use failure route (GW 2) then if Client A hang up > all works fine, but the problem is when is Client B who hang up, its > like a new conversation, GW 2 send BYE to openser and Openser just send > "503 Service Not avilable - No gateways" to GW2, but doesnt send nothing > to ClientA, any idea ???? > > > Thx in advance >

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[Users] direct manipulation of DB
by unplug 09 Jan '06

09 Jan '06

I am using openser with mysql as db. There is a tool call openserctl to manage the data in the openser. I wonder whether I can direct manipulate the db directly without using openserctl. I have tried to insert a record in the aliases table directly using mysql client but it doesn't work. Can I conclude direct manipulate the DB is impossible?

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[Users] deactivate an account
by unplug 05 Jan '06

05 Jan '06

Hi , Does anyone know how can I deactivate an account? Does openser already implement that function to deactivate an account, how?

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Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting
by 'Jan Janak' 02 Jan '06

02 Jan '06

Hello, Attached is a patch that implements "swap_direction" parameter of acc module. If you turn the parameter on in the configuration file: modparam("acc", "swap_direction", 1) then the acc module will swap Calling-Station-ID and Called-Station-ID values when necessary (in this case BYE comming from the callee). Put the patch in top level source directory (ser-0.9.x) and type: patch -p0 < swap.patch The patch should work with any 0.9.x release. Note that Alan DeKok is wrong in thinking that this is a bug in SER. This particular problem is a result of incomplete specification of RADIUS use with SIP. I will commit the patch in CVS so it will be included in future SER releases. Jan. On 23-12-2005 17:10, Lenir wrote: > Please read the reply below from one of the maintainers of freeradius: > > "Lenir" <lenirsantiago(a)yahoo.com> wrote: > > But if UserB hangs up on UserA: SER generates a stop-record where the > > Calling-Station-Id is UserB and the Called-Station-Id is UserA, this > > is the undesired and incorrect behavior. > > It would appear to be a bug in SER. > > > To me the Calling-Station-Id and the Called-Station-Id should be the > > same for both start and stop records, am I right by thinking that? > > Yes. > > > According to the developers of SER/OpenSER, this is the correct > > behavior, whoever sends the hangup signal (BYE or CANCEL) is > > considered the Calling-Station-Id, and they are unwilling to modify or > > create a patch to "fix" this. > > What they do for something inside of SER is their business. When they > generate RADIUS packets, they should follow RADIUS standards and > interoperability. The expectation, as you said, is that the > Calling/Called-Station-Id doesn't change during a session. If it does, it's > a bug and they should fix it. > > Alan DeKok. > - > List info/subscribe/unsubscribe? See > http://www.freeradius.org/list/users.html > > > > -----Original Message----- > From: 'Jan Janak' [mailto:jan@iptel.org] > Sent: Thursday, November 17, 2005 10:33 AM > To: Lenir > Cc: 'Klaus Darilion'; serdev(a)iptel.org; serusers(a)iptel.org; > devel(a)openser.org; users(a)openser.org > Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius > Accounting > > On 17-11-2005 10:21, Lenir wrote: > > In this case the radius proxy wont work, because you never can anticipate > > who hangs up the call, thus radius wont know who hung up the > call...Besides > > all other voice applications/hardware (SIP and H323) that use radius do > not > > behave like that, the Called-Station-ID ALWAYS remains the same, as with > the > > Calling-Station-ID. > > Could you name those SIP applications that behave the way you describe ? > > Jan. >

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[Users] Dealing with reinvites
by Matt Schulte 02 Jan '06

02 Jan '06

All, We have had a new issue arise within openser, basically we connect to a SIP carrier that uses Asterisk as a proxy. Naturally, this carrier is hellbent on offloading the RTP traffic, thus the reinvites. My issue is that they appear to be sending the second invite "too fast", before the initial RTP even gets a chance to establish. This was working just fine before, and I believe they changed something. Anyway, they're refusing to help with this issue, so now I must "fix" it on my end. OpenSER is in-part, the problem.. See below: (Sorry if formatting is off, dump from tethereal ..) 0.000000 204.13.233.13 -> 206.80.70.47 SIP/SDP Request: INVITE sip:8886963856@206.80.70.47, with session description 0.001752 206.80.70.47 -> 204.13.233.13 SIP Status: 100 trying -- your call is important to us 0.002723 206.80.70.47 -> 206.80.70.54 SIP/SDP Request: INVITE sip:+1314xxxxxxx@sip.stl.netlogic.net, with session description 0.010327 206.80.70.54 -> 206.80.70.47 SIP Status: 100 Trying 0.292175 206.80.70.54 -> 206.80.70.47 SIP Status: 180 Ringing 0.292692 206.80.70.47 -> 204.13.233.13 SIP Status: 180 Ringing 2.016246 206.80.70.54 -> 206.80.70.47 SIP/SDP Status: 200 OK, with session description 2.016893 206.80.70.47 -> 204.13.233.13 SIP/SDP Status: 200 OK, with session description 2.050871 204.13.233.13 -> 206.80.70.47 SIP Request: ACK sip:+1314xxxxxxx@206.80.70.54 2.051233 204.13.233.13 -> 206.80.70.47 SIP/SDP Request: INVITE sip:+1314xxxxxxx@206.80.70.54 2.051827 206.80.70.47 -> 204.13.233.13 SIP Status: 100 trying --<<KABOOM>>-- 2.051956 206.80.70.47 -> 206.80.70.54 SIP/SDP Request: INVITE sip:+1314xxxxxxx@206.80.70.54, 2.052112 206.80.70.47 -> 206.80.70.54 SIP Request: ACK sip:+13142664000@206.80.70.54 2.053028 206.80.70.54 -> 206.80.70.47 SIP/SDP Status: 200 OK, with session description 2.053336 206.80.70.54 -> 206.80.70.47 SIP Status: 503 Server error Where you see "kaboom", is the problem. If you notice, the time between the first ACK and second INVITE is about 1ms .. While this shouldn't be a problem, you'll notice when the ACK/INVITES are getting statefully forwarded, they're getting sent out of order. The end point is an Asterisk machine, thus the 503 error. Carrier_Ast --> OpenSER --> Netlogic_Ast I am using loose_route to forward, just like everyone else, they are even hitting loose_route in the correct order.. Thoughts? Suggestions? This seems like something internal to OpenSER, I have tried butchering the config to force the ACK out first and it just created more problems. Thanks! Matt S

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[Users] handling of command line parameters
by Klaus Darilion 31 Dec '05

31 Dec '05

Hi! Is this observation correct? 1. If an option is defined in the openser.cfg file, this definition will be used. 2. In an option is not defined in the openser.cfg file, the default value will be used, except the option is defined via a command line parameter. regards klaus

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[Users] invite timers
by Mike Williams 31 Dec '05

31 Dec '05

I've run into an interesting issue with invite timers: when a user makes an outbound call, to say, a PSTN gateway, it would be best if the OpenSER fr_invite_timer was set very high. However, this timer controls how long I want the call to ring for before timing out and sending users to my local voicemail server.I would like the timer to be set lower for this. Has anyone come up with a creative way around this? Am I missing something obvious? All help is appreciated. ---Mike

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[Users] help: anonymous call
by unplug 30 Dec '05

30 Dec '05

Below is the anonymous call invite method compared with the normal call. In the from header, there is missing username. How can I add an anonymous username (i.e. <sip:anonymous@203.193.46.4>) in the from header if the call is from anonymous? Invite method from Anonymous call ---------------------------------------------- U 203.193.46.4:54169 -> 203.193.46.6:5060 INVITE sip:36428473@203.193.46.6:5060 SIP/2.0. Via: SIP/2.0/UDP 203.193.46.4:5060. From: <sip:203.193.46.4>;tag=1924AB7C-EE5. <----------------------------- missing username To: <sip:36428473@203.193.46.6>. Date: Wed, 28 Dec 2005 09:22:19 GMT. Invite method from normal call ---------------------------------------------- U 203.193.46.4:50197 -> 203.193.46.6:5060 INVITE sip:36428473@203.193.46.6:5060 SIP/2.0. Via: SIP/2.0/UDP 203.193.46.4:5060. From: <sip:34263634@203.193.46.4>;tag=978CF54-291. To: <sip:36428473@203.193.46.6>. Date: Fri, 23 Dec 2005 06:29:08 GMT.

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[Users] Media Proxy
by Fernando Rodriguez 29 Dec '05

29 Dec '05

Is there any manual on how to setup multiple mediaproxy instances in one machine ?? For different network cards and different lans?? Thanks Fernando Rodriguez V. frod(a)aitelecom.net AITelecom S.A. de C.V. http://www.aitelecom.net

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[Users] Disabling rtpproxy or mediaproxy when both SIP UA are behind same NAT
by Rafael Risco GV 29 Dec '05

29 Dec '05

Hi I think I am very close to solve this issue, based on suggestions of Tavis and Klaus for openser in: http://openser.org/dokuwiki/doku.php?id=how_to_set_up_nathelper_rtpproxy_wh… I´ve been able to disable rtpproxy in this cases but I have a problem with accounting, if the callee hungup first appears an error and no cdr for BYE is generated until caller hungup the call, bellow the errors and config. thanks for any help rafael 4(22193) BYE - STOP ACCOUNTING 4(22193) DEBUG: add_param: tag=3658302912 4(22193) grep_sock_info - checking if host==us: 13==13 && [192.168.1.205] == [10.0.2.130] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) grep_sock_info - checking if host==us: 13==9 && [192.168.1.205] == [127.0.0.1] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) check_self: host != me 4(22193) grep_sock_info - checking if host==us: 13==13 && [10.0.2.130] == [10.0.2.130] 4(22193) grep_sock_info - checking if port 5060 matches port 5060 4(22193) check_via_address(10.0.1.25, 192.168.1.100, 0) 4(22193) ERROR: udp_send: sendto(sock,0x4051e800,588,0,0x4051f2ac,16): Operation not permitted(1) 4(22193) msg_send: ERROR: udp_send failed 4(22193) ERROR:tm:t_forward_nonack: sending request failed 4(22193) DEBUG: add_to_tail_of_timer[4]: 0x4051f2cc 4(22193) DEBUG: add_to_tail_of_timer[0]: 0x4051f2dc 4(22193) ERROR:tm:t_relay_to: t_forward_nonack returned error 4(22193) parse_headers: flags=ffffffffffffffff 4(22193) check_via_address(10.0.1.25, 192.168.1.100, 0) 4(22193) WARNING:vqm_resize: resize(0) called . . .

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