>
> Hello All.
> I am having some trouble with call transfers when asterisk is
> the 2nd party called and I hope to benefit from your experience.
>
> I want to use asterisk for call park/pickup and have configured
> openser to relay calls made to ruri 700-720 to asterisk running on
> localhost:5069
>
>
> Call flow:
>
> phone A calls phone B (both phones are polycom)
>
> Phone B answers
>
> then phone b user presses transfer and dials 700
>
> asterisk plays back 701 as the parking lot location
>
> phone B user presses transfer again.
> at this time phone b is not disconnected from asterisk system
> phone A is also connected to asterisk and hears 702 as the
> parking lot location (as if asterisk places the user at priority 1
> for that
> context)
>
>
> From phone C calling 702 will connect phone C to phone A.
>
>
> This was a specific example but this transfer problem is not
> limited to call park only. It happens any time asterisk is the
> second party called in call transfer.
>
> Thanks in advance for your help.
>
> --
> Zahid
>
>
>
>
>
> On May 8, 2007, at 1:56 PM, Christian Schlatter wrote:
>
>
>
> I think I found out why this doesn't work as expected. After phone
> 1 receives REFER from phone 2, it sends a new INVITE to the
> asterisk server. This INVITE includes a Replaces: header that tells
> the receiver (asterisk) to replace an existing SIP dialog with the
> new one.
>
> RFC 3891 "The SIP Replaces Header", Section 3 "UAS Behavior", defines:
>
> "the UA attempts to accept the new INVITE, reassign the user
> interface and other resources of the matched dialog to the new
> INVITE, and shut down the replaced dialog."
>
> But your SIP trace shows that asterisk doesn't shut down the
> replaced dialog (by sending a BYE), which is the reason why phone 2
> does not get disconnected after hitting "transfer" the second time.
>
>
> Instead of creating a new call park slot (702) when phone 1 sends
> the Replaces: INVITE to asterisk, asterisk should be intelligent
> enough to figure out that this INVITE actually replaces the
> existing SIP dialog with phone 2. And asterisk should not create a
> new park slot 702 but directly put phone 1 on hold at park slot 701
> and send a BYE to phone 2.
>
> Although asterisk supports the Replaces: header when used e.g. as a
> gateway, I have some doubts that the call park/pickup
> implementation does so too. Especially since it was designed to be
> used in "PBX mode" where asterisk acts as B2BUA for all involved
> call legs.
>
> Maybe this should be opened as a new feature/bug request on the
> asterisk bug tracker. Or maybe there is a asterisk setting that
> controls this behavior, I'm not really an asterisk expert myself ;-)
>
> --
>
Hi all,
I wasn't
Openser is a SIP proxy.
There is Media Server. It works together with OpenSER.
The OpenSER is monitoring the flow of the messges. In case of this flow to be RTP, the Openser redirects for the Media Server.
How does it work?
Did you understand?
thanks
Danilo
Inovax Engenharia de Sistemas Ltda.
Rio de Janeiro - RJ / Brasil
Hi,
I was wondering if it's possible to create an openser script to
automatically send an error response when an INVITE request for a
specific SIP URI is received.
For example, an INVITE to "sip:e403@domain" would return a 403 response,
and requests to "sip:eXYZ@domain.com" would return a XYZ response.
I've tried using avp:
# get username
avp_printf ( "$avp(i:2)", "$oU" );
# get error code
avp_subst ("$avp(i:2)", "/e*/\1/" );
# send reply
sl_send_reply ( "$avp(i:2)", "Error" );
But apparently it's not allowed to pass an avp parameter to the
sl_send_reply function.
Is there another way to accomplish this?
kind regards,
Sigrid Thijs
Thank you Michal,
I have migrated to SER-2.0 and have compiled with HONOR_MADDR, but
having some problem making it work. I cannot get the digest
authentication working. I am using the following code for REGISTER,
if (method=="REGISTER") {
if(!www_authorize("","credentials")){
www_challenge("","0");
break;
};
};
For some reason, www_challenge is not recognised. And gives parse
error. Can anyone please tell me how can I get the digest
authentication working??
kumar
On 5/7/07, Michal Matyska <michal(a)iptel.org> wrote:
> Hi,
>
> commetns inline
>
> On Po, 2007-05-07 at 10:04 +0545, KUMAR wrote:
> > Hi Michal,
> > Sorry for my late response. Here are the packets while the transport
> > UDP is used.
> > regards
> > kumar
> >
> > On 5/4/07, Michal Matyska <michal(a)iptel.org> wrote:
> > > Hi Kumar,
> > >
> > > could you please attach also the capture (at the server is enough) of
> > > the UDP both way messages? There call flow is quite strange and not
> > > understandable.... (well not surprised that's M$ RTC :)
> > >
> > > Some comments for now:
> > > 1) it seems the CLIENT IS BROKEN and does not understand lr=on flag for
> > > loose routing - see how is the ACK generated !!! You can try change rr
> > > module parameter to put just lr into the record-route (this might
> > > explain your rr and rr_preset difference)
>
> Taking back, the client code is not broken, it is obsoleted only ;-)
> It uses strict router procedures (RFC2543), to create the Route header
> and Request-URI in subsequent within-dialog requests, but SER's
> loose_route function call should be able to escape form that (and it is,
> the rewritten request uri shows that). If you want to check, you should
> see "after_strict" message in your debug output.
>
> > > 2) M$ is used to misuse maddr, there is compile time flag HONOR_MADDR
> > > which you should have set to route such requests taking maddr into
> > > account
>
> This seems to be the main issue, the client does not put maddr into the
> UDP requests. Check the Route header from the client in the TCP and UDP
> captures to see the difference.
>
> Recompile SER with the option HONOR_MADDR set and it should work. If
> not, please provide SER output with debug=4 statement in you ser.cfg,
> while running the TCP usecase.
>
> Michal
>
>
> > > On Pá, 2007-05-04 at 09:52 +0545, KUMAR wrote:
> > > > Here are the packet captures. I was not quite sure whether I have to
> > > > include packet captures from server or the UAs so i've included all of
> > > > them. ua_A is sending INVITE to ua_B.
> > > > thank you
> > > >
> > > > regards
> > > > kumar
> > > >
> > > > On 5/3/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
> > > > >
> > > > >
> > > > > Jiri Kuthan wrote:
> > > > > > actually I was suggesting SIP message dump (PCAP file) as opposed to log files. -jiri
> > > > >
> > > > > or ngrep dump:
> > > > >
> > > > > ngrep -t -W byline -d any port 5060
> > > > >
> > > > > regards
> > > > > klaus
> > > > >
> > > > >
> > > > > >
> > > > > > At 13:10 03/05/2007, KUMAR wrote:
> > > > > >> Here is the message dump that i get when using TCP as transport.
> > > > > >>>
> > > > > >>> On 5/3/07, Jiri Kuthan <jiri(a)iptel.org> wrote:
> > > > > >>>> I think you would have to send message dumps first so that [serusers] volunteers
> > > > > >>>> have material for providing an answer.
> > > > > >>>>
> > > > > >>>> -jiri
> > > > > >>>>
> > > > > >>>> At 12:07 03/05/2007, KUMAR wrote:
> > > > > >>>>> -
> > > > > >>>>> Hi all,
> > > > > >>>>> I already posted this message yet havent got any replies. I really
> > > > > >>>>> need to find this out.
> > > > > >>>>> Please anyone reply to this problem.
> > > > > >>>>>
> > > > > >>>>> I am using SER-0.9.6 and the problem i'm having is this. When using
> > > > > >>>>> the following ser.cfg, it works well when the transport UDP is used.
> > > > > >>>>> But when transport TCP is used, then it only results in one way IM,
> > > > > >>>>> only from the UA from which the INVITE is being sent. Moreover, when
> > > > > >>>>> record_route is used instead of record_route_preset, then even that
> > > > > >>>>> one way IM doesn't work. But it works well with UDP.
> > > > > >>>>>
> > > > > >>>>> Can anyone please point me as to where the problem might be??
> > > > > >>>>> Thank you in advance.
> > > > > >>>>>
> > > > > >>>>> kumar
> > > > > >>>>>
> > > > > >>>>>
> > > > > >>>>> Content-Type: application/octet-stream; name=ser.cfg
> > > > > >>>>> X-Attachment-Id: f_f18sg2bl
> > > > > >>>>> Content-Disposition: attachment; filename="ser.cfg"
> > > > > >>>>>
> > > > > >>>>> _______________________________________________
> > > > > >>>>> Serusers mailing list
> > > > > >>>>> Serusers(a)lists.iptel.org
> > > > > >>>>> http://lists.iptel.org/mailman/listinfo/serusers
> > > > > >>>>
> > > > > >>>>
> > > > > >>>> --
> > > > > >>>> Jiri Kuthan http://iptel.org/~jiri/
> > > > > >>>>
> > > > > >>>>
> > > > > >>
> > > > > >> Content-Type: audio/x-pn-realaudio-plugin; name="msg_dump.rar"
> > > > > >> Content-Disposition: attachment; filename="msg_dump.rar"
> > > > > >> X-Attachment-Id: f_f1949112
> > > > > >>
> > > > > >> _______________________________________________
> > > > > >> Serusers mailing list
> > > > > >> Serusers(a)lists.iptel.org
> > > > > >> http://lists.iptel.org/mailman/listinfo/serusers
> > > > > >
> > > > > >
> > > > > >
> > > > > > --
> > > > > > Jiri Kuthan http://iptel.org/~jiri/
> > > > > >
> > > > > > _______________________________________________
> > > > > > Serusers mailing list
> > > > > > Serusers(a)lists.iptel.org
> > > > > > http://lists.iptel.org/mailman/listinfo/serusers
> > > > >
> > > > _______________________________________________
> > > > Serusers mailing list
> > > > Serusers(a)lists.iptel.org
> > > > http://lists.iptel.org/mailman/listinfo/serusers
> > >
> > >
>
>
Hello Users
Todays onwards I'm Started integrating SEMS.
Previously I'm using the Asterisk + openSER1.2 .For a small change , I
shifted into the SEMS for voicemail
By this url ( http://openser.org/pipermail/users/2005-September/000844.html)
i found t_write_reg
that is not in OpenSER tm modules , I found t_write_fifo/unix
How I can do the voicemail or announcement with OpenSER + SEMS
Any one give me ant hints
Regardinng this points
--
Thanks &Regards
Ravi Prakash Sunkara
Hi all,
I am using OpenSER 1.2.0.
I am going to install media proxy,
How does it work? How the Openser knows and obtains to monitor the flow and to set in motion the midia server?
Danilo
Inovax Engenharia de Sistemas Ltda.
Rio de Janeiro - RJ / Brazil
Hi all,
I am trying to configure my openSER with TLS support and get the
following error message:
May 7 15:44:39 tiger openser: WARNING:init_tls: disabling
compression due ZLIB problems
May 7 15:44:39 tiger openser: ERRRO:init_tls: null openssl compression
methods
Anyone could give me a hint of what is going wrong with my settings? Thank
you so much for the help.
Regards.
Han
Hi, all,
I could not find a good explanation of the releasing policy for OpenSER.
I mean, 1.2.0 was released on March 12, 2007. I know that the future
release is 1.3.0. But is there going to be 1.2.1 ever released?
There were many bugs fixed in 1.2.x since 1.2.0 was released. I think I
read a developer's reply that the best and most stable OpenSER 1.2.x
release sources can only be downloaded from the SVN 1.2 branch. Is this
true? And what is the trunk version in SVN? Is it only for the next
release (1.3.0) and changes to the trunk do not go to 1.2.x branch
automatically?
The download page (http://www.openser.org/mos/view/Download/)
unfortunately does not tell that to get the most stable 1.2.x release
one has to use SVN Download. I think most of the OpenSER users just
download the released 1.2.0 sources.
By the way, looks like the so called "daily" SVN snapshots have not been
posted since April 23. Is this a known problem?
Thanks,
Anatoly.