Hello,
my new email address for any matter is miconda(a)gmail.com -- as I was not
able to access my inbox over last week, please resend to the new address
any message you sent to me (not the ones sent to the lists, I will pick
them up from there).
Thank you,
Daniel
_________________________________________________________________
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Hello everybody,
I noticed that OpenSER replies with status code "408 Request Timeout"
when the fr_timer (tm module) expires. I am wondering if the response
code "504 Server Timeout" would be more appropriate in this case. The
status code 408 means that "the server could not produce a response,
e.g., a user location, within the time indicated in the Expires
request-header field. The client MAY repeat the request without
modifications at any later time."; and the status code 504 means "the
server did not receive a timely response from an external server it
accessed in attempting to process the request." It sounds to me that the
description of the status code 504 better describes what happens in
OpenSER when the fr_timer expires. Any comments? I am using OpenSER 1.2.x.
Best regards,
Anatoly Pidruchny.
On Friday 22 February 2008, Arturo Díaz Almagro wrote:
(forward to user list for the archives)
> This is my case. We have a SIP telephony system based on OpenSER with a
> redundant MySQL database ring, so the OpenSER queries always to the primary
> DB unless it fails. When the "master" database dies, a monit process
> detects the failure and modify properly the database entry in the DNS zone
> file. We have tested this redundancy in two way I expose you here:
>
> - mysql process is killed in the database host machine. In this case
> OpenSER realizes the failure and asks the DNS inmediately for the
> database IP address and the service keeps working as expected. No problem
> here.
>
> - shutdown the interface of the machine where mysql is hosted. In this
> case OpenSER does not ask the DNS for the new address. It keeps trying
> over and over the connection with the master and until the interface is not
> turn on the system keeps hung.
>
> The difference between the two cases is that in the first case, the host
> machine responses with an ICMP (port destination unreachable) when OpenSER
> queries the database. In the second case we have no answer from the MySQL
> host machine.
>
> So, if the mysql's host interface is shutdown the system hangs but if the
> mysql process dies the system works properly. In the first case both
> machines (OpenSER and MySQL) has IP visibility, not in the second, so the
> problem should be related to that. Do you know why this is happening? Might
> you, please, be so kind to tell me how to fix this problem or at list how
> to take a workaround to get it working properly?
Hi Arturo,
Well, this problem is probably caused because the mysql_query function works
synchronly, i.e. it blocks the process if no progress on the query could be
made. You could easily reproduce this with simple IP filtering with iptables.
If you don't get a negative reply to the query, it will simply hang for the
TCP timeout from the kernel.
We've written in the past a custom extension to the mysql module, called
mysql-mt. This elevates this problem by uses a multi-threading approach. But
this worked not that great.
For recent mysql driver versions this is not needed anymore, i've committed in
revision 3606 a improvement to the trunk version of openser that uses the
recently introduced extensions in the mysql lib. This could be easily
backported to your version. Take a look into the mysql documentation if this
different timeouts are supported on your driver version.
Cheers,
Henning
Hi All
I am using 1.3 with tls enabled, I am using the default certs that come
with the source code and I did disable all the tls_verify options, I can
logon to my UA and make calls however there is one problem, every couple
minutes the sip client disconnects and re-connects "it does have a
re-login option".
Any ideas why the UA is getting disconnected every couple of minutes ?
Note: I have also tried with UDP without tls and the same thing happened
upon monitoring SIP packets I noticed the following, as long as the
client is connected it regularly sends REGISTER packets, when the
disconnect happens the following happens
SIP/2.0 401 Unauthorized
Thx
With Regards
Ali Jawad
Hello,
I am trying to do a performance test of OpenSER (1.3.0) with TLS using SIPP as the UAS/UAC.
The TLS authentication seems to work and the calls are stablished and then terminated (SIPP default scenario). However, I am receiving the following error message from OpenSER:
Feb 19 12:58:40 [12709] ERROR:core:_tls_read: something wrong in SSL:
Feb 19 12:58:40 [12709] ERROR:core:tcp_read_req: failed to read
What do these messages mean? Can I ignored them?
The SIP command I am using is:
sipp -s 100 -sn uac 192.168.1.10:5061 -t ln -tls_cert 101-cert.pem -tls_key 101-privkey.pem
The TLS configuration I am using in my openser.cfg file is:
disable_tls = no
listen = tls:192.168.1.10:5061
tls_verify_server = 1
tls_verify_client = 1
tls_require_client_certificate = 1
tls_method = SSLv23
tls_certificate = "/openser/etc/openser/tls/proxy/proxy-cert.pem"
tls_private_key = "/openser/etc/openser/tls/proxy/proxy-privkey.pem"
tls_ca_list = "/openser/etc/openser/tls/proxy/proxy-calist.pem"
I will appreciate any help about this matter.
Thanks,
JB
_________________________________________________________________
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Thanks.
So, if the client aren't able to perform this function. Can SER do that? Or
I need to setup the Asterisk server to help me to perform the 'call pickup'
functions. Correct me if I'm wrong.
Thanks.
Roa Yu
-----Original Message-----
From: SIP [mailto:sip@arcdiv.com]
Sent: Monday, February 25, 2008 11:17 AM
To: roayu
Cc: samuel; serusers(a)lists.iptel.org
Subject: Re: [Serusers] Intercept SIP call
When your client registers to SER as a particular user, that
registration is stored in SER.
If you register multiple times with the same user, ALL of those contacts
are called when a call is sent to that user.
N.
roayu wrote:
>
> Thanks for your reply and explanation, samuel.
>
> But what do you mean by "If you want to send the initial call to both
> phone B and C, either in parallel or sequentally, then SER can easily
> do the job"? Could you please further explain on this or you can
> provide me so links to read bout this.
>
> Thanks!
>
> Roa Yu.
>
> ------------------------------------------------------------------------
>
> *From:* samuel [mailto:samu60@gmail.com]
> *Sent:* Friday, February 22, 2008 4:39 PM
> *To:* roayu
> *Cc:* serusers(a)lists.iptel.org
> *Subject:* Re: [Serusers] Intercept SIP call
>
> The way you have explained the use case would require the phones to do
> complex SIP operations.
> Phone C need to know the dialog identifiers (dialog info package
> probably would help)of the call between A and B and then use replaces
> header to "intercept the call". I presonally haven't seen phones doing
> this.
>
> If you want to send the initial call to both phone B and C, either in
> parallel or sequentally, then SER can easily do the job. Just read a
> few mails or the getting started document and you'll see how to do this.
>
> Hope it helps,
> Samuel.
>
> 2008/2/20, roayu <roayu(a)ctisys.net <mailto:roayu@ctisys.net>>:
>
> Hi there!
>
> Recently, I had setup a SIP environment with few sip phones, SIPphone
> A, SIPphone B, and SIPphone C. For instance, when SIPphone A called to
> SIPphone B, but B is not able to answer the call. Can SIPphone C
> answer A's call?
>
> I would like to know whether is SER able to have the intercept sip
> calls function? If yes, what module or configuration that I need to set?
>
> Thanks in advance!
>
> Best regards,
>
> RoaYu
>
>
> _______________________________________________
> Serusers mailing list
> Serusers(a)lists.iptel.org <mailto:Serusers@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
>
Greger, et. al.,
I like your idea enough to volunteer as a tester + documentation writer.
I do a lot of tech and user writing. Active in VoIP since the beginning.
I am active VoIP engineer and developer. Also PSTN & VoIP network
designer-builder.
I have my own private machines, network, and VoIP gateway resources.
Prefer CentOS or FC*
..mike..
>Hi guys,
>I have been playing with the following idea:
>Create a ready-to-run OS image with everything that is needed for
>iptel.org apps pre-installed + a complete installation of:
>* SER 2.0 (release)
>* rtpproxy
>* SEMS
>* SERweb
>* maybe sipsak, some monitoring tools, etc
>
>
>The idea is to create a small script 'config_iptelorg' for
>configuring the installation to your needs. You should then be able
>to download the ready image, boot it, go through the script and have
>an up and running iptel.org proxy and app server just like the
>iptel.org free SIP service in maybe 10-15 minutes. This way you
>could host a SIP service for your own domain with close to no setup at all.
>
>Some questions to you:
>* Is there any interest for this at all?
>* I was thinking about using Ubuntu 7.10 server as the OS. Any
>thoughts/preferences?
>* Should the image be an Amazon EC3 image (you could use
>http://www.rightscale.com and get it running in no time with 10
>run-hours free) or should it be a VMware appliance to be run with
>free VMware Player?
>* Other suggestions/comments?
>
>I could need some help with this, anyone interested in lending me a
>hand? (could be anything, documentation, testing, installation, etc)
>g-)
Hi, im trying to stablish the real time connection between Openser and
Ejabberd both with mysql support, any suggestions ?
--
Thanks
Frank Gonzalez
414-6260492
Thanks for your reply and explanation, samuel.
But what do you mean by "If you want to send the initial call to both phone
B and C, either in parallel or sequentally, then SER can easily do the job"?
Could you please further explain on this or you can provide me so links to
read bout this.
Thanks!
Roa Yu.
_____
From: samuel [mailto:samu60@gmail.com]
Sent: Friday, February 22, 2008 4:39 PM
To: roayu
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] Intercept SIP call
The way you have explained the use case would require the phones to do
complex SIP operations.
Phone C need to know the dialog identifiers (dialog info package probably
would help)of the call between A and B and then use replaces header to
"intercept the call". I presonally haven't seen phones doing this.
If you want to send the initial call to both phone B and C, either in
parallel or sequentally, then SER can easily do the job. Just read a few
mails or the getting started document and you'll see how to do this.
Hope it helps,
Samuel.
2008/2/20, roayu <roayu(a)ctisys.net <mailto:roayu@ctisys.net> >:
Hi there!
Recently, I had setup a SIP environment with few sip phones, SIPphone A,
SIPphone B, and SIPphone C. For instance, when SIPphone A called to SIPphone
B, but B is not able to answer the call. Can SIPphone C answer A's call?
I would like to know whether is SER able to have the intercept sip calls
function? If yes, what module or configuration that I need to set?
Thanks in advance!
Best regards,
RoaYu
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