Ray,
The problem is in the Sipura's SDP :
> a=rtpmap:18 G729a/8000.
My understanding of RFC3551 is that the encoding name "G729a" is not
compliant. Therefore when the Sipura sends this to your Softphone, the
softphone which is compliant to RFC3551 rejects the request and you get
a BYE.
Maybe force your Sipura to use G711a/u.
Regards,
Craig Peacock
> I have set up both xten and sipura, I want to test onnet calls between
> them. I am fine with xten eyebeam to xten eyebeam, and sipura to
> sipura, both scenario works fine. when I do xten to sipura 2000, it
> rings and when pick up, xten sends BYE right after ACK. I've tried
> 1.0.1 verson and the CVS HEAD, happens to both version. did anyone
> encountered the same problem?
>
> thanks
>
> Ray
> ps. debug attached
>
> Server: Sipura/SPA2000-2.0.13(g).
> Content-Length: 238.
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
> Supported: x-sipura.
> Content-Type: application/sdp.
> .
> v=0.
> o=- 12432 12432 IN IP4 192.168.10.244.
> s=-.
> c=IN IP4 192.168.10.244.
> t=0 0.
> m=audio 43712 RTP/AVP 18 100 101.
> a=rtpmap:18 G729a/8000.
> a=rtpmap:100 NSE/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:30.
> a=sendrecv.
When a REFER is sent from ua1 to ua2 I need to modify
the "Refer-To:" URL.
ua1 will send the refer using the alias 222 which is
for username 5551212
Therefore, ua2 needs to receive 5551212 in the
Refer-To:.
How can I modify the Refer-To field prior to
forwarding the request to ua2?
Thanks!!
__________________________________________________
Do You Yahoo!?
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Hi!
I have problems with avp_subst:
I have $avp(s:pai) with the value <tel:+43108> (the < and > belongs to
the AVP)
Then I want to extract the number into $avp(s:cli):
avp_subst("$avp(s:pai)/$avp(s:cli)","/tel:(.*)/\1/")
The result is <+43108>
Can someone explain me why the leading < belongs to the result?
Shouldn't it be removed?
thanks
klaus
Bogdan,
I upgraded to the current cvs version, and I think that has fixed it. With no change to openser.cfg, it's been running for about 3 hours now without any reported memory issues.
Douglas.
> -----Original Message-----
> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro]
> Sent: Monday, June 26, 2006 10:20 AM
> To: Douglas Garstang
> Cc: Users(a)openser.org
> Subject: Re: [Users] Out of Memory
>
>
> Ok Douglas - please let me know if the latest changes in
> resolver have
> any effect on the leak you observed.
>
> regards,
> bogdan
>
> Douglas Garstang wrote:
>
> >Bogdan,
> >
> >Don't appear to have ever received that message.
> >
> >To answer your questions:
> >We are not using ENUM at all. We aren't using our own module
> for DNS lookups.
> >I will try the latest cvs.
> >
> >Doug
> >
> >
> >
> >>-----Original Message-----
> >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro]
> >>Sent: Monday, June 26, 2006 8:33 AM
> >>To: Douglas Garstang
> >>Cc: Users(a)openser.org
> >>Subject: Re: [Users] Out of Memory
> >>
> >>
> >>See the list. I sent you a reply :
> >> http://www.openser.org/pipermail/users/2006-June/005321.html
> >>
> >>regards,
> >>bogdan
> >>
> >>Douglas Garstang wrote:
> >>
> >>
> >>
> >>>>-----Original Message-----
> >>>>From: Douglas Garstang
> >>>>Sent: Thursday, June 22, 2006 12:09 PM
> >>>>To: Bogdan-Andrei Iancu
> >>>>Cc: Users(a)openser.org
> >>>>Subject: RE: [Users] Out of Memory
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>>-----Original Message-----
> >>>>>From: Douglas Garstang
> >>>>>Sent: Wednesday, June 21, 2006 12:48 PM
> >>>>>To: Bogdan-Andrei Iancu
> >>>>>Cc: Users(a)openser.org
> >>>>>Subject: RE: [Users] Out of Memory
> >>>>>
> >>>>>
> >>>>>Bogdan,
> >>>>>
> >>>>>I finally managed to upload the memory dumps to pastebin.
> >>>>>
> >>>>>
> >>Links are:
> >>
> >>
> >>>>>Hopefully you can access them.
> >>>>>
> >>>>>Memory dump right after OpenSER was started:
> >>>>>http://pastebin.com/723872
> >>>>>Memory dump after OpenSER running for 20min:
> >>>>>http://pastebin.com/723890
> >>>>>Memory dump after OpenSER running for 50min:
> >>>>>http://pastebin.com/723902
> >>>>>
> >>>>>The problem started to occur between the 20 and 50 minute
> >>>>>samples. About 20 calls had been processed in that time.
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>So... I'm wondering if anything has been determined with this...
> >>>>
> >>>>
> >>>>
> >>>>
> >>>Hi Bogdan. Did you manage to find anything with this?
> >>>
> >>>Douglas.
> >>>
> >>>
> >>>
> >>>
> >
> >
> >
>
>
Hello,
I am using SER-0.9.6. How to send instant message programmatically, how
to build sip message and which function can by used(like t_uac,
t_request).
Can you help where and which to be changed in ser code.
Thank you,
Regards,
Sriram Srinivas.
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I am new to SER and radius and I am having a digest problem.
My ser.cfg section:
if (method=="REGISTER" || method=="INVITE") {
log("**************************WTF mate?");
if (!radius_www_authorize("voip.fasttrackcomm.net")) {
log("WTF mate auth?");
www_challenge("voip.fasttrackcomm.net", "0");
break;
};
My users file:
Line 50-
9703751000(a)voip.fasttrackcomm.net Auth-Type := Digest, User-Password
== "5321"
Reply-Message = "Authenticated"
Radius_logging:
--- Walking the entire request list ---
Cleaning up request 96 ID 55 with timestamp 44be52aa
Sending Access-Reject of id 56 to 127.0.0.1 port 34690
Reply-Message = "Authenticated"
Waking up in 4 seconds...
rad_recv: Access-Request packet from host 127.0.0.1:34691, id=57, length=247
User-Name = "9703751000(a)voip.fasttrackcomm.net"
Digest-Attributes = 0x0a0c39373033373531303030
Digest-Attributes =
0x0118766f69702e66617374747261636b636f6d6d2e6e6574
Digest-Attributes =
0x022a3434626535336361303461366438663636346561346630313231303134366630366531
6361316333
Digest-Attributes =
0x04207369703a35303040766f69702e66617374747261636b636f6d6d2e6e6574
Digest-Attributes = 0x0308494e56495445
Digest-Response = "e311972978c1d06b6341c18aba7373c7"
Service-Type = Sip-Session
Sip-Uri-User = "9703751000"
NAS-Port = 5060
NAS-IP-Address = 127.0.0.1
Processing the authorize section of radiusd.conf
modcall: entering group authorize for request 98
modcall[authorize]: module "preprocess" returns ok for request 98
rlm_digest: Adding Auth-Type = DIGEST
modcall[authorize]: module "digest" returns ok for request 98
rlm_realm: Looking up realm "voip.fasttrackcomm.net" for User-Name =
"9703751000(a)voip.fasttrackcomm.net"
rlm_realm: Found realm "voip.fasttrackcomm.net"
rlm_realm: Proxying request from user 9703751000 to realm
voip.fasttrackcomm.net
rlm_realm: Adding Realm = "voip.fasttrackcomm.net"
rlm_realm: Authentication realm is LOCAL.
modcall[authorize]: module "suffix" returns noop for request 98
users: Matched entry 9703751000(a)voip.fasttrackcomm.net at line 50
modcall[authorize]: module "files" returns ok for request 98
modcall: leaving group authorize (returns ok) for request 98
rad_check_password: Found Auth-Type Digest
auth: type "digest"
Processing the authenticate section of radiusd.conf
modcall: entering group authenticate for request 98
rlm_digest: Converting Digest-Attributes to something sane...
Digest-User-Name = "9703751000"
Digest-Realm = "voip.fasttrackcomm.net"
Digest-Nonce = "44be53ca04a6d8f664ea4f01210146f06e1ca1c3"
Digest-URI = "sip:500@voip.fasttrackcomm.net"
Digest-Method = "INVITE"
A1 = 9703751000:voip.fasttrackcomm.net:5321
A2 = INVITE:sip:500@voip.fasttrackcomm.net
KD =
bde2dd5ada6b4377bf8167616c47236f:44be53ca04a6d8f664ea4f01210146f06e1ca1c3:d9
ed5d99fee1e1f435339ff6f206edac
rlm_digest: FAILED authentication
modcall[authenticate]: module "digest" returns reject for request 98
modcall: leaving group authenticate (returns reject) for request 98
auth: Failed to validate the user.
Delaying request 98 for 1 seconds
Finished request 98
I didn't any Bye request in ngrep and ethereal,
That Bye Request is not forwarding to server.
On 7/21/06, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote:
>
> Hi Klaus,
>
> What U 'd told , I did it.
>
> but i didn't see the Bye Request to server , By using both ngrep and
> ethereal..
>
>
>
>
> On 7/21/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
> >
> > How should I know?
> >
> > Do what I suggested. Use ngrep (or any other packet sniffer like tcpdump
> > or ethereal/whireshark) to watch where the BYE gets lost. Then I can
> > help.
> >
> > btw: please send emails to the list
> >
> > regards
> > Klaus
> >
> > raviprakash sunkara wrote:
> > > Hello Klaus,
> > > thank U..... for replying to me..
> > > CAN help On Bye Problem
> > >
> > > is openser is problem or network problem..
> > >
> > > can do that...
> > > Bye
> > >
> > > On 7/20/06, * Klaus Darilion* <klaus.mailinglists(a)pernau.at
> > > <mailto: klaus.mailinglists(a)pernau.at>> wrote:
> > >
> > > raviprakash sunkara wrote:
> > > >
> > > > Hi Users
> > > >
> > > > I'm Using the openser with Nat Bu using the RTP..
> > > >
> > > > After invite method rtp is also established between the
> > caller and
> > > > callee ...
> > > > Audio is clear..
> > > > But when I send Bye request to Callee , it not hung upping to
> > > callee it
> > > > still estallishing the call...
> > > >
> > > > I think that ....
> > > > 1) openSER server is not getting the
> > > request from
> > > > callee or caller
> > > > 2) OPenser is respones the BYE ..
> > > > 3) Router(firewall) gateways is
> > blocking the
> > > > Bye request , to pass the request to openser,
> > > > 4) Route behind the UA's
> > isbloacking....
> > >
> > >
> > > You have to check which one of this is the real case.
> > >
> > > Use ngrep at the SIP proxy to verify if the BYE is received and
> > forward.
> > > Take a look at the IP addresses and ports in the Route headers and
> > > request URI and Contact headers. Make sure the BYE is sent to the
> > proper
> > > IPaddress:socket.
> > >
> > > regards
> > > klaus
> > >
> > >
> > >
> > >
> > > --
> > > Thanks and Regards with cheers
> > > Sunkara Ravi Prakash (Voip Developer)
> > > Hyperion Technology
> > > Kondapur, Hi-tech city,
> > > Hyderabad.
> > > www.hyperion-tech.com <http://www.hyperion-tech.com>
> > > +91-9985077535
> >
> >
>
>
> --
> Thanks and Regards with cheers
> Sunkara Ravi Prakash (Voip Developer)
> Hyperion Technology
> Kondapur, Hi-tech city,
> Hyderabad.
> www.hyperion-tech.com
> +91-9985077535
>
--
Thanks and Regards with cheers
Sunkara Ravi Prakash (Voip Developer)
Hyperion Technology
Kondapur, Hi-tech city,
Hyderabad.
www.hyperion-tech.com
+91-9985077535
Hi Feriante!
Please Cc: the list too.
To test TLS you have choose where you want to use TLS:
- between the SIP clients and the SIP proxy
- between SIP proxies and to gateways
- both
between the SIP clients and the SIP proxy:
get a SIP client which Supports TLS (eyebeam, minisip, SNOM phones
(maybe also the SNOM softphone?))
between SIP proxies:
get 2 domain names for each proxy (they can be hosted on the same PC
with different ports).
make NAPTR and SRV (RFC3263) entries with TLS as preferred protocol
configure both proxies with the same rootCA
configure both proxies with a client certificate and key
use t_relay(). This should check NAPTR records and use tls as protocol.
regards
klaus
Ferianto siregar wrote:
>
> Dear Klaus Darilion,
>
> Thank you very much for your kind-hearted to reply my message.Thanks
> I have tried your suggestion and it works. Thanks
> But, would you mind if I ask you a question anymore?
> How can I test my TLS configuration, so It can run as a security in voip
> communication?
> What should I prepare, Klaus? Would you mind..
> Please help me..
>
> Thanks with all regards,
>
>
> Ferianto
>
> */Klaus Darilion <klaus.mailinglists(a)pernau.at>/* wrote:
>
> Hi Feriante!
>
> The TLS syntax has changed and we have forgotten to update the default
> config file. As I sad, read the TLS tutorial
> (http://openser.org/docs/tls.html) and also the Wiki describes the new
> syntax
> (http://openser.org/dokuwiki/doku.php?id=migrating_openser_v1.0.x_to_v1.1.x).
>
> In your case:
> tls_verify_server = on
> tls_verify_client = on
> tls_require_client_certificate = on
>
> regards
> klaus
>
> On Fri, July 21, 2006 5:40, Ferianto siregar said:
> > Dear all,
> >
> > First of all I would like to say thanks to all of you who has
> given me
> > some helps and suggestions to solve my problem in configuring my
> openser
> > system. Thank you very much.
> > I have some questions in configuring the TLS now, I do hope
> anybody can
> > help me. These are the questions:
> > 1. Now, I try to configure the openser system for using the TLS.
> So, as
> > Klaus Darilion said before, I must configure openser.cfg file
> first. So,
> > I uncomment (enable) the TLS by deleting "#" in openser.cfg file.
> After
> > that, I try to restart the openser again. But, when I run it again
> > (after restarting), I get 3 error messages. Here are the error
> message:
> > [root@localhost openser]# openser restart
> > 0(5783) parse error (27,12-13): parse error
> > 0(5783) parse error (27,12-13): unknown config
> > variable
> > 0(5783) parse error (27,14-15):
> > ERROR: bad config file (3 errors)
> > 0(5783) destroy_tls: Entered
> > 0(5783) shm_mem_destroy
> > [root@localhost openser]#
> >
> > As I see, the error is at line 27. I see that it contain
> "tls_verify=1"
> > and "tls_require_certificate=0". I don`t know what is wrong with this
> > line because As I see from all mailinglist`s messages, they didn`t
> > change this line and if they change it, they just change the
> value, for
> > example :
> > tls_verify = on
> > tls_require_certificate = on
> >
> > I have tried this effort, but I get the same error message.
> > Does anybody can give me a suggestion what sould i do? Please...
> >
> > 2. If the error can be solved, how can I test my TLS configuration? I
> > mean how I can test whether it can run correctly ( It can secure the
> > communication system in openser)?
> >
> > Please help me..I do hope anyone can help me to solve this problem.
> > Thank you.
> >
> >
> > Regards with cheers,
> >
> >
> >
> > Ferianto
> >
> >
> >
> >
> > ---------------------------------
> > Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US
> (and 30+
> > countries) for 2¢/min or
> > less._______________________________________________
> > Devel mailing list
> > Devel(a)openser.org
> > http://openser.org/cgi-bin/mailman/listinfo/devel
> >
>
>
>
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Hello,
I want to send an immediate 100 trying message when an Invite is
received, then I do a DB lookup, then I rewrite the RURI and forward the
message using t_relay.
Since I have already sent a 100 trying manually I'd like to short
circuit the 100 t_relay sends so multiple 100 trying messages aren't
sent.
Does anyone know of a way to do this?
Thanks,
T.R.
Ezequiel Colombo wrote:
> Hi Jon, you can use DNS SRV records pointing to your gateways and use
> something like t_relay_to("pool.yourgateways.com").
Hi
Thanks for that.
I have only ever seen t_relay_to() examples using IP addresses so was
not sure if they also supported DNS.
However what is the advantage to using DNS SRV as opposed to normal DNS
round robin i.e. A records with the same priority?
Regards
Jon
--
Jon Farmer
Telford, Shropshire, UK