Padmaja wrote:
> Hi,
>
> Thanks for the reply,
> Could you provide me the example script for stateless forwarding of
> invite when a call is received from say a 911 operator to the server?
> I suppose this should be implemented by checking the From field of the
> invite and if it is from a certain 911, then it should be forwarded
> statelessly (or is this logic incorrect?). I am new to the Openser
> stuff, so I do not wish to change the script myself.
Your logic is correct, you can either check the From: header or the
request URI or any other part of an incoming SIP message to make the
routing decision.
A simple example looks like:
if (is_method('INVITE'))
{
# check if from URI domain equals 911operator.com
if ($fd == '911operator.com')
{
# relay stateless using request URI
forward();
exit;
}
else
{
# relay statefull using request URI
if (!t_relay())
{
sl_reply_error();
};
exit;
}
}
Please always CC users(a)lists.openser.org so that others can take
advantage of the discussion.
/Christian
>
> Thanks again,
> Padmaja
>
> ----- Original Message ----- From: "Christian Schlatter" <cs(a)unc.edu>
> To: "Padmaja" <padmaja.rv(a)vodcalabs.com>
> Cc: <users(a)lists.openser.org>
> Sent: Monday, November 05, 2007 4:53 AM
> Subject: Re: [OpenSER-Users] Openser as a conditional stateless proxy
>
>
>> Padmaja wrote:
>>> Hi all,
>>>
>>> Can any one tell me if the same running instance of Openser can be
>>> configured as a stateful proxy for some user accounts and for some
>>> numbers like emergency services, it acts like a stateless proxy, just
>>> forwarding the request to the destination?
>>
>> Yes, it only depends on what forwarding function is used in the
>> routing script:
>>
>> - t_forward(..) --> stateless
>>
>> - t_relay(..) --> stateful
>>
>> /Christian
>>
>
Hi,
This is probably more SIP related than openSER, but I'll ask it here
anyway
When we were at VON I brought up the subject of accounting and
verifying proper call tear down, and it seems the ideal solution from
a sip perspective is RFC 4028 (SIP session timers) which tracks the
actual sip session with the UA. What I'm starting to realize is that
support for this is very limited and I haven't found much evidence if
any in the devices where network failures are most likely to occur
ATAs and end user devices.
This seems to leave me relying on the media stream, but this also has
its issues with network issues occurring when calls are placed on
hold. Does anyone have any good suggestions for accurate accounting
when dealing with end users, or is what I described as good as I'm
going to get, until I find some solutions that support 4028?
Cheers
Miles Scruggs
Wide Ideas | Operations | miles.scruggs(a)wideideas.com | 509.525.6522
ext 4880
This should in (poor) theory work. I would check by watching the mysql port number traffic
what data gets actually fetched before making the next step.
At 13:01 01/11/2007, Piyush.Bansal(a)relianceada.com wrote:
>Hi All,
>
>We're trying to test Call Forwarding Unconditional (CFU) using SER-2.0.0-rc1. We're adding the subscriber's attribute using following command -
>./ser_ctl attrs set 50001(a)192.168.112.14 fwd_always_target=50(a)192.168.112.14
>
>After that we tried to call to 50001. After receiving the call, SER fails to fetch the attribute (fwd_always_target) value from the DB.
How do you know SER fails to fetch it from DB?
-jiri
>Even, the check "if ($tu.fwd_always_target)" fails in this case..
>
>The config file is also attahced with the mail.
>
>
>
>Can anybody tell me why this is happening?????? Plz reply ASAP.....
>
>
>[]
>
>
>
>The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.
>
>
>
>_______________________________________________
>Serdev mailing list
>Serdev(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serdev
--
Jiri Kuthan http://iptel.org/~jiri/
Hi All,
We are using SER-0.9.6. We want it to be configured in a way that it can
have two interfaces towards two different application.
One interface will be over TLS and another will be over UDP.
Can we do that???? If yes, then how???
thanx in advance.....
regds,
--Piyush Bansal
The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.
Spam detection software, running on the system "mail.iptel.org", has
identified this incoming email as possible spam. The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email. If you have any questions, see
the administrator of that system for details.
Content preview: I USER SER2.0, RUN serctl ps show "500 Command 'ps' not
found" why [...]
Content analysis details: (7.6 points, 5.0 required)
pts rule name description
---- ---------------------- --------------------------------------------------
-0.0 SPF_HELO_PASS SPF: HELO matches SPF record
-0.0 SPF_PASS SPF: sender matches SPF record
0.0 BAYES_50 BODY: Bayesian spam probability is 40 to 60%
[score: 0.5027]
0.2 MIME_BASE64_NO_NAME RAW: base64 attachment does not have a file name
1.9 MIME_BASE64_TEXT RAW: Message text disguised using base64 encoding
0.2 MIME_BASE64_BLANKS RAW: Extra blank lines in base64 encoding
1.9 DNS_FROM_RFC_BOGUSMX RBL: Envelope sender in
bogusmx.rfc-ignorant.org
0.2 DNS_FROM_RFC_ABUSE RBL: Envelope sender in abuse.rfc-ignorant.org
1.4 DNS_FROM_RFC_WHOIS RBL: Envelope sender in whois.rfc-ignorant.org
1.7 DNS_FROM_RFC_POST RBL: Envelope sender in
postmaster.rfc-ignorant.org
Hi guys,
Great to meet many of you this week at VON and looking forward to
getting together again soon.
We have a question about the gateway aspect which is part of the LCR
module. We are needing to route to various carriers which the module
handles great, but we are trying to figure out if there is an elegant
way to handle the situation where a carrier provides us with multiple
proxies. For instance if carrier A provides proxy1 and proxy2,
carrier B etc.... When we get the AVP back which will be our stack to
do serial forking the sequence is dependent on the proxy response (or
lack of response)
For instance if we get a 400 error back from either proxy there isn't
much use in trying the other proxy we need to more along to a
completely different carrier. For 500 errors or time outs will
continue on using the same carrier. The obvious ways to do this: to
abandon the gateway aspect all together and just build logic which we
could use to process what would appear to be a multi demential array
instead of single diminutional array. I'm just writing to see if
there is a nice out of the box way to handle this scenario, also if
not, is this something which developers would be interested in having
added to the module?
Cheers
Miles Scruggs
Wide Ideas | Operations | miles.scruggs(a)wideideas.com | 509.525.6522
ext 4880
Hello everybody
there is a bug in SER that removes the last byte from a message body if
this happens to be a binary zero.
This may be the case when the Content-Type is something binary (i.e.
*not* using "Content-Type: text/something").
The bad code is in udp_server.c. It is present in SER 0.9.x a well as in
2.0. I have reported it in the bug tracker as SER-324 and attached a
comment which points out the bad section of the code.
How shall I proceed to fix this bug?
Best regards
Stefan
Hi, all
I am using Openser 1.2.2 - tls.
I have 400 registered users.
Everything works well as traffic is low.
If traffic is high, the calls are not completed (Busy tone)
Can anyone help me?
Hi all,
I am using the openserctl command to add users to the Openser database and I
can verify if these numbers are added to the Subscriber table in the mysql
database.However, if I try to look into the Subscriber table through Webmin,
I do not see all the users but only a few. What should I do to reflect all
the users through webmin as well?
Thank You,
Padmaja
----- Original Message -----
From: <users-request(a)lists.openser.org>
To: <users(a)lists.openser.org>
Sent: Friday, November 02, 2007 3:27 PM
Subject: Users Digest, Vol 30, Issue 2
> Send Users mailing list submissions to
> users(a)lists.openser.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.openser.org/cgi-bin/mailman/listinfo/users
> or, via email, send a message with subject or body 'help' to
> users-request(a)lists.openser.org
>
> You can reach the person managing the list at
> users-owner(a)lists.openser.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Users digest..."
>
>
> Today's Topics:
>
> 1. Re: Set up two SIP to PSTN calls and then connect them
> (Andreas Granig)
> 2. Traffic (Gerson A. Matiolli)
> 3. Re: How to expose the expires value in REGISTER (Robert Dyck)
> 4. Beep in audio stream. (Marc Dirix)
> 5. Is it possible to insert avp to reply message? (Tung Tran)
> 6. Re: Set up two SIP to PSTN calls and then connect them
> (Bogdan-Andrei Iancu)
> 7. Re: Set up two SIP to PSTN calls and then connect them (CSB)
> 8. Re: Is it possible to insert avp to reply message?
> (I?aki Baz Castillo)
> 9. maddr in contact (Allan Chao ( ??? ))
> 10. Re: Set up two SIP to PSTN calls and then connect them
> (I?aki Baz Castillo)
> 11. Re: Traffic (Henning Westerholt)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 01 Nov 2007 13:07:42 +0100
> From: Andreas Granig <agranig(a)sipwise.com>
> Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then
> connect them
> To: CSB <kjcsb(a)xnet.co.nz>
> Cc: users(a)lists.openser.org
> Message-ID: <4729C18E.7040507(a)sipwise.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> It's already included, see
> http://svn.berlios.de/viewcvs/sems/trunk/apps/click2dial/
>
> Andreas
>
> CSB wrote:
>> Is there any update regarding the click2dial plugin that was planned to
>> be introduced to the trunk?
>>
>>
>>
>> Regards
>>
>>
>>
>> Cameron
>>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Thu, 01 Nov 2007 10:35:49 -0200
> From: "Gerson A. Matiolli" <gerson(a)cambridgetelecom.com.br>
> Subject: [OpenSER-Users] Traffic
> To: users(a)lists.openser.org
> Message-ID: <1193920549.5276.9.camel@jupiter2>
> Content-Type: text/plain
>
> Hi, all
>
> I am using Openser 1.2.2 - tls.
>
> I have 400 registered users.
>
> Everything works well as traffic is low.
>
> If traffic is high, the calls are not completed (Busy tone)
>
> Can anyone help me?
>
>
>
>
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Thu, 1 Nov 2007 09:12:20 -0700
> From: Robert Dyck <rob.dyck(a)telus.net>
> Subject: Re: [OpenSER-Users] How to expose the expires value in
> REGISTER
> To: Christian Schlatter <cs(a)unc.edu>
> Cc: users(a)lists.openser.org
> Message-ID: <200711010912.20409.rob.dyck(a)telus.net>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On Wednesday 31 October 2007, Christian Schlatter wrote:
>> Robert Dyck wrote:
>> > I am wondering how to expose and test the value of the expires
>> > parameter
>> > in a REGISTER request.
>> >
>> > I am experimenting with openser as the basis for a home phone network.
>> > I
>> > use multiple devices with the same user ID. They register locally (
>> > with
>> > no reply ) and with an external service provider. The contacts are
>> > mangled to show the public address of openser. Multiple registrations
>> > result in a single AOR at the external registrar. Incoming calls from
>> > the
>> > outside are forked and ring the local phones. Local phones can also
>> > call
>> > each other without the hairpin problem associated with STUN enabled
>> > phones.
>> >
>> > The problem is that a softphone will deregister when it is closed or
>> > its
>> > profile changes. This would deregister the AOR at the external
>> > registrar.
>> > The remaining phones could not receive calls from the outside until
>> > they
>> > refreshed their registrations.
>> >
>> > I would like to prevent deregistration at the external registrar unless
>> > the phone that was deregistering was the only remaining one. The first
>> > step would be to identify REGISTER messages where the expires value is
>> > equal to zero.
>>
>> Both 'Expires' header and 'expires' contact uri parameter have to be
>> checked like e.g.
>>
>> if ((is_present_hf("Expires") && $(hdr(Expires){s.int}) == 0) ||
>> ($(ct{param.value,expires}) == '0'))
>> {
>> # someone tries to unregister
>> }
>>
>> Have a look at
>> http://www.openser.org/dokuwiki/doku.php/transformations:1.2.x if you're
>> not familiar with the PV transformations introduced with 1.2.
>
> I am indeed unfamiliar with PV transformations. I will have a look it. I
> was
> afraid I might have to do something ugly with regular expressions. I
> probably
> should not put off upgrading any longer.
>
> Thanks, Rob
>
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Thu, 1 Nov 2007 21:35:06 +0100
> From: Marc Dirix <marc(a)electronics-design.nl>
> Subject: [OpenSER-Users] Beep in audio stream.
> To: users(a)lists.openser.org
> Message-ID:
> <871BA93E-E51A-4297-906D-789BA5798461(a)electronics-design.nl>
> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>
> Hi,
>
> I'm currently setting op an openser server.
> My setup at the moment is as follows:
>
> registrar (pstn) <=> yate (sip server) <=> openser <=> sip_phone.
>
> As I make a call with the sip phone to a pstn line, the rtp stream is
> forwarded from
> the yate server to openser, which acts as en media proxy with
> rtpproxy. During
> the call I get very annoying beeps every 2 or 3 seconds.
>
> The beeps sound a bit like cost-beeps or somethin.
>
> When I connect the phone directly to the yate server however, which
> then starts
> acting as media-proxy, I do not get any beeps.
>
> Furthermore, if I remove force_rtp_forward() from openser config, it
> stops being proxy for the stream, but still I get these annoying
> beeps. Excluding any problems with rtpproxy.
>
>
> Clearly, the registrar sends these beeps, but he doesn't send them
> when I connect with yate.
> Am I missing something that can trigger this behaviour?
>
> Thanks,
>
> Marc Dirix
>
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Fri, 2 Nov 2007 09:41:33 +0700
> From: Tung Tran <tr.tung(a)gmail.com>
> Subject: [OpenSER-Users] Is it possible to insert avp to reply
> message?
> To: <users(a)lists.openser.org>
> Message-ID: <200711294133.621270@VGN-TXN15P>
> Content-Type: text/plain; charset="us-ascii"
>
> An HTML attachment was scrubbed...
> URL:
> http://lists.openser.org/pipermail/users/attachments/20071102/26276838/atta…
>
> ------------------------------
>
> Message: 6
> Date: Fri, 02 Nov 2007 05:30:45 +0200
> From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>
> Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then
> connect them
> To: Andreas Granig <agranig(a)sipwise.com>
> Cc: users(a)lists.openser.org
> Message-ID: <472A99E5.9000401(a)voice-system.ro>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi,
>
> Or you can use this script, with no external dependency:
>
> http://openser.svn.sourceforge.net/viewvc/openser/branches/1.2/examples/web…
>
> regards,
> Bogdan
>
> Andreas Granig wrote:
>> It's already included, see
>> http://svn.berlios.de/viewcvs/sems/trunk/apps/click2dial/
>>
>> Andreas
>>
>> CSB wrote:
>>
>>> Is there any update regarding the click2dial plugin that was planned to
>>> be introduced to the trunk?
>>>
>>>
>>>
>>> Regards
>>>
>>>
>>>
>>> Cameron
>>>
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users(a)lists.openser.org
>> http://lists.openser.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
>
>
> ------------------------------
>
> Message: 7
> Date: Fri, 2 Nov 2007 16:32:53 +1300
> From: "CSB" <kjcsb(a)xnet.co.nz>
> Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then
> connect them
> To: "'Andreas Granig'" <agranig(a)sipwise.com>
> Cc: users(a)lists.openser.org
> Message-ID: <003401c81d01$0d9d8920$28d89b60$(a)co.nz>
> Content-Type: text/plain; charset="us-ascii"
>
> Thanks.
>
> I currently use OpenSER and Asterisk and I can get the call set up using
> ctd.sh. The question I have relates to the accounting. Using the ctd.sh
> script is there a way to get the CDR records written from OpenSER? If I
>understand correctly, OpenSER drops out of the call signalling and will not
> receive any BYEs so accounting will be impossible; am I correct? Asterisk
> will record the calls but billing them appropriately using those records
> would be problematic (I think).
>
> If using the SEMS option, am I correct in thinking that it would be
> possible
> to use the accounting records from OpenSER?
>
> Regards
>
> Cameron
>
> -----Original Message-----
> From: Andreas Granig [mailto:agranig@sipwise.com]
> Sent: Friday, 2 November 2007 1:08 a.m.
> To: CSB
> Cc: users(a)lists.openser.org
> Subject: Re: Set up two SIP to PSTN calls and then connect them
>
> It's already included, see
> http://svn.berlios.de/viewcvs/sems/trunk/apps/click2dial/
>
> Andreas
>
> CSB wrote:
>> Is there any update regarding the click2dial plugin that was planned to
>> be introduced to the trunk?
>>
>>
>>
>> Regards
>>
>>
>>
>> Cameron
>>
>
>
>
>
> ------------------------------
>
> Message: 8
> Date: Fri, 2 Nov 2007 09:39:31 +0100
> From: I?aki Baz Castillo <ibc(a)in.ilimit.es>
> Subject: Re: [OpenSER-Users] Is it possible to insert avp to reply
> message?
> To: users(a)lists.openser.org
> Message-ID: <200711020939.31335.ibc(a)in.ilimit.es>
> Content-Type: text/plain; charset="ISO-8859-1"
>
> El Friday 02 November 2007 03:41:33 Tung Tran escribi?:
>> Hi all,
>
> Please, when creating a **new** mail press "create new mail", but don't
> press "Reply" on any other mail of any other thread. If you do so your
> mail
> will appear contained in a wrong thread, broking it and make it very
> difficult to understand.
>
> Thanks.
>
>
> --
> I?aki Baz Castillo
> ibc(a)in.ilimit.es
>
>
>
> ------------------------------
>
> Message: 9
> Date: Fri, 2 Nov 2007 17:15:45 +0800
> From: Allan Chao ( ??? ) <AllanChao(a)taiwanmobile.com>
> Subject: [OpenSER-Users] maddr in contact
> To: <users(a)lists.openser.org>
> Message-ID:
> <970C4ACFBCFD2349B388E8907A2A7B5E80D2B4(a)TCCEXCH12.pcdc.com.tw>
> Content-Type: text/plain; charset="big5"
>
> Hi :
>
> I have a flow UE -> gateway (use openser and runs proxy mode) ( ip :
> 192.168.1.2) -> SIP Proxy ( ip: 192.168.1.3),
>
> if i have two user , UE1(192.168.1.5) and UE2(192.168.1.6) send REGISTER
> request to SIP proxy through gateway,
>
> but our gateway add a maddr = "192.168.1.2" string in contact header,so
> the contact in REGISTER becomes <username@ host ; maddr="192.168.1.2">.
>
> now , if UE1 send INVITE message to UE2, how does sip proxy to do if
> receive INVITE message? it will send invite message to UE2 through
> gateway ( maddr parameter) ?
> and does openser has support maddr in contact or not . thx.
>
>
> allan
>
