Brett,
I have to agree that from web perspective the php.ini model is the
best. But to migrate to something like that, we need to find a way to
keep synchronize the web content with the README files from source tree
If I understand your suggestion , is to add to the current README
content
I was a little confused. I was trying to explain to the operator that the
reply doesn't have a RURI and that I send it with the same To/From headers
they send me.
I did check the headers and they are they same.. I'm going to be doing some
more testing on this tomorrow. It's my understanding that th
Hi Brett,
Ok, more clear now :)
" ...because the "To:" field doesn't match the RURI."
I guess they refer at RURI from IINVITE request and TO hdr from reply ?
If so, Both this entities are generated by UAC. The TO hdr from reply is
the TO header from INVITE (plus the TO tag).
So, can you visua
Bogdan,Problem is, it's a real hunt to find those comment pages.. or at
least I think it is.. I really like the php.net model that Matti mentions in
his email. That way, the comments and suggestions are right there with the
docs. Almost like the doc page and comment stream and just concatenated.
I
Hi,
we need to run both a SIP client and a SIP server on a embedded systems and
we have done this sucessully in the past with embedded Linux (OpenSER and
PJSUA), but for this project we have a customer requirement to use Windows
CE as operating system, so my question is if any of you guys out t
Ugh! I didn't make that easy, did I.. Yes, in failure route I t_reply (not
relay) with a 503. They ignore the REPLY, there is no new branch. The UAC is
ignoring my REPLY and the operator of that device is telling me that it's
because the "To:" field doesn't match the RURI.
On Thu, Feb 12, 2009 at
Hi,
Great idea. People could post there example conf's regarding that
matter. Much like php.net works.
/Matti
-Original Message-
From: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro]
Sent: den 11 februari 2009 17:45
To: Brett Nemeroff
Cc: Matti Zemack; users@lists.opensips.org
Subje
Hi Amit,
We are using weSIP for this? or? Because there is no mention about
the software you are using and definitely is not OpenSIPS as you have
JAVA there :D
Regards,
Bogdan
Amit Bansal wrote:
> Hi All,
>
> I am trying to build a SIP application which will act as proxy to the
> pho
Hi Brett,
Brett Nemeroff wrote:
> All,
> I'm having an issue with a customer's nextone sbc. They send a call
> out, I send it to my upstream. My upstream is broken (separate issue
> althogether). They send me 183..183.. 500. When I get the 500, I send
> a 503 to the originator of the request (m
Hi Inaki,
OpenSIPS transport stack has no support for dealing with ICMP error.
Mainly because of two reasons:
- catching ICMP error differs from OS to OS, hard to get a common
ground
- you need to keep state at transport level; mapping between
transactions/messages and destination.
Reg
Hi All,
I am running two pcscf's by changing the port.Actually my
requirement is i am testing REFER message for session continuity.so i need
two proxy servers.
But when i use "*dig @IP proxyserver address"* it is giving status:NOERROR
for one proxy and status:NXDOMAIN for other proxy
Hi all,
I have installed opensips-1.4.4 using debian packages. I have also
installed mysql package for debain support. For Wesip support which package
should i add so that i can get wesip support (i.e seas.so module) in the
libraries.
--
Thanking You,
Ashwini BR Naidu
__
Hi All,
I am trying to build a SIP application which will act as proxy to the phones
those are connected to it like a server that is in call centers.
I have have made a servlet code is as follows..
package com.sip.servlet;
import javax.servlet.sip.SipServlet;
import javax.servlet.sip.SipServletR
All,I'm having an issue with a customer's nextone sbc. They send a call out,
I send it to my upstream. My upstream is broken (separate issue
althogether). They send me 183..183.. 500. When I get the 500, I send a 503
to the originator of the request (my customer).. and they ignore the
request, so I
Folks,
I am not sure, if this is the right forum to post this query.
Just wanted to know, if anyone faced this issue and found a workaround .
I am trying to install OpenXCAP server in my ubuntu machine (Hardy 8.04).
I tried installing through apt-get install openxcap, I am getting dependency
error
Ah, I see.. I totally forgot about an internally generated 408. I was just
trying to suggest that if you receive a 408 from an endpoint, it isn't
necessarily grounds for being blacklisted. However, given the application,
it probably makes sense.
On Wed, Feb 11, 2009 at 5:04 PM, Iñaki Baz Castillo
There have been a couple of people (myself included) that have taken
advantage of this 'bug' / 'feature'. Probably should get some input
from Bogdan as to whether it's something to be used or avoided.
Regards,
Norm
Geoffrey Mina wrote:
> Either a 'bug' or a 'feature'... whatever you want to ca
Either a 'bug' or a 'feature'... whatever you want to call it :) It
worked as I _expected_, but not as intended.
thanks,
Geoff
On Wed, Feb 11, 2009 at 6:18 PM, Iñaki Baz Castillo wrote:
> El Jueves, 12 de Febrero de 2009, escribió:
>> I have addressed the ds_select_domain() usage issue after re
El Jueves, 12 de Febrero de 2009, escribió:
> I have addressed the ds_select_domain() usage issue after re-reading
> the documentation upon your guidance. The last config I sent over is
> using ds_next_dst() in the failure_route.
>
> I am not sure why OpenSIPS didn't have a startup error, but it
>
I have taken your initial suggestions and mofied by deployment. You
were a great help in assisting me to see some loop holes in my
configuration.
The updated configuration, based on your suggestions is attached.
There is also a debug=10 trace attached for both a scenario where my
asterisk server
El Miércoles, 11 de Febrero de 2009, Geoffrey Mina escribió:
> Thanks for the input. I did test the 408 scenario by disabling one of
> the asterisk systems. I was connected properly and the one end point
> was removed from the pool. I will run some more tests and analyze the
> debug info more ca
Brett,
The 408 in this scenario is generated by OpenSIPS internally. As
discussed previously, these systems are simply acting as IVR end
points. They ALWAYS answer the call. It is not dependant on anything
upstream or outside of the Asterisk server. If OpenSIPS generates an
internal timeout...
El Miércoles, 11 de Febrero de 2009, Brett Nemeroff escribió:
> That 408 doesn't make sense to me..
>
> If you got a 408 back, the gateway you are sending to WORKS, but IT'S
> destination is failing.
That's incorrect. RFC 3261 states clearly that, in case a request has no reply
(after the corres
Geoffrey Mina wrote:
> I generally don't like to presume that individuals want to
> help me Pro Bono
But we do it all day.
When you ask the question in $300 terms, you make it a $300 issue.
When you ask a question, you make it into a compelling challenge for
those who love to help others in t
That 408 doesn't make sense to me..
If you got a 408 back, the gateway you are sending to WORKS, but IT'S
destination is failing.
say for example you send to a gateway and that gateway sends to 10 different
providers.. if this call routes to provider A, which doesn't respond, then
your gateway may
Thanks for the input. I did test the 408 scenario by disabling one of
the asterisk systems. I was connected properly and the one end point
was removed from the pool. I will run some more tests and analyze the
debug info more carefully to see exactly what path is being taken.
In regards to aster
El Miércoles, 11 de Febrero de 2009, Geoffrey Mina escribió:
> ## If we are in-dialog loose_route() should return true and we should
> ## end up here. I am not sure the subsequent check of has_totag() is
> ## necessary, but I could be wrong.
Yes, you should do it. In your actual config, if a
Hi, if OpenSIPS routes a request via UDP to SERVER_IP:SERVER_PORT and receives
a ICMP error:
ICMP SERVER_IP udp port SERVER_PORT unreachable
then OpenSIPS remains retransmiting the request (until transaction timer
expires and so). After that it generates a 408 response.
I wonder why. Some fi
I don't understand the following:
---
## Handles relay of INVITE messages
## with round-robin load balancing
###
Thanks for your clarification. I do see where you are coming from.
Here is my config with some better comments. I have also attached the
file in case the email client mangles the formatting.
thanks,
Geoff
#
# OpenSIPS configuration
# by Geoff Mina
#
# Please refer to reference http://www.op
El Miércoles, 11 de Febrero de 2009, Geoffrey Mina escribió:
> It's nice to see there are actually some folks out there who can
> (almost) see where I am coming from. The general response of the
> community is quite surprising. I was expecting much different
> responses based on the level of supp
John,
Attached is my config file. I generally don't like to presume that
individuals want to help me Pro Bono, which is why I offered the
bounty.
Thanks,
Geoff
On Wed, Feb 11, 2009 at 4:33 PM, John Rose wrote:
> If those items are not involved it is definitely less complicated.
>
> Maybe you sh
El Miércoles, 11 de Febrero de 2009, Mark Sayer escribió:
> I'm afraid I have to echo Geoff's response to this. It's fascinating
> to see so many people telling him that this community isn't going to
> help him. It would seem that if you can't or are unwilling to help
> for whatever reason, then ju
If those items are not involved it is definitely less complicated.
Maybe you should post your script and seek comments? What's the big deal
it's only 150 lines...
John
> -Original Message-
> From: Geoffrey Mina [mailto:geoffreym...@gmail.com]
> Sent: Wednesday, February 11, 2009 1:49 P
It's nice to see there are actually some folks out there who can
(almost) see where I am coming from. The general response of the
community is quite surprising. I was expecting much different
responses based on the level of support specific questions receive
when submitted.
My system is working
Geoff,
The old OpenSER/Kamailio site has a "business" mailing list that is
dedicated to this type of topic. Seems like a good idea to create a
similar mailing list for the OpenSIPS folks.
I have to agree with Adrian's recent comments in regard to narrowing
down the scope of your problem/reque
I'm afraid I have to echo Geoff's response to this. It's fascinating
to see so many people telling him that this community isn't going to
help him. It would seem that if you can't or are unwilling to help
for whatever reason, then just press delete. His request, while
perhaps a bit naive, was h
You will likely obtain more help from this mailing lists if you narrow
down the scope of your questions and ask for help about the specific
problems you encounter. For example locate your problem, describe it,
provide a useful trace.
You will be surprised how effective this can be. Just a d
There is no NAT
There is no Firewall (running local IPTables)
There are no 'users'
There are no 'phones'
There is a single upstream Carrier
There ARE a plurality of asterisk based IVR systems which are
processing the media. All configured identically, all on the same
network.
I have been doing SI
Often people and companies underestimate the complexities of a SIP proxy
installation. There are many variables particularly ones with carriers,
phones, NAT's, firewalls... Many assume it is just configuration. A lot of
it is but unless you have worked it in depth you won't know.
Here is a free "
Well, I wouldn't really be looking for solutions to any problems
found. I would simply be looking for someone to identify possible
problems in my config. Once an 'issue' has been identified, I will
work to address it myself. My config works great in a bubble, so I
can't be that far off.
Basical
2009/2/11 Geoffrey Mina :
> Thanks for your reply, but I tend to disagree. I have spent many
> hours programming my OpenSIPs deployment as well as processing test
> calls. It is a VERY simple deployment, which is why I think that 2
> hours will be more than enough. The factors are:
>
> 1 - We ha
Geoffrey,
I am trying to help you fine tune your expectations. Anyone can
briefly tell you what is wrong as they spot some issue. But a good
consultant would tell you what you have missed too and how to solve
problems you did not foresee yet. So by asking the wrong question and
setting up
If it is simply a matter of financials, I would be willing to discuss
that matter with any interested party off-line.
thanks.
On Wed, Feb 11, 2009 at 12:25 PM, Alex Balashov
wrote:
> Iñaki Baz Castillo wrote:
>> 2009/2/11 Adrian Georgescu :
>>> Geoffrey,
>>> In my experience nobody on this maili
Please let us all know if you get the blessing at least we know we are
wrong.
Adrian
On Feb 11, 2009, at 6:25 PM, Geoffrey Mina wrote:
Thanks for your reply, but I tend to disagree. I have spent many
hours programming my OpenSIPs deployment as well as processing test
calls. It is a VERY si
Thanks for your reply, but I tend to disagree. I have spent many
hours programming my OpenSIPs deployment as well as processing test
calls. It is a VERY simple deployment, which is why I think that 2
hours will be more than enough. The factors are:
1 - We have one SIP provider
2 - We are ONLY h
Iñaki Baz Castillo wrote:
> 2009/2/11 Adrian Georgescu :
>> Geoffrey,
>> In my experience nobody on this mailing list with enough knowledge in this
>> matter will be able to help fix all your possible miss-configurations as it
>> can span fixing the whole universe depending on an infinit matrix
>>
2009/2/11 Adrian Georgescu :
> Geoffrey,
> In my experience nobody on this mailing list with enough knowledge in this
> matter will be able to help fix all your possible miss-configurations as it
> can span fixing the whole universe depending on an infinit matrix
> of possibilities.
> A 2 hour 'ble
Geoffrey,
In my experience nobody on this mailing list with enough knowledge in
this matter will be able to help fix all your possible miss-
configurations as it can span fixing the whole universe depending on
an infinit matrix of possibilities.
A 2 hour 'blessing' is most likely exactly w
Hello,
I am looking for anyone who would consider themselves an 'expert' in
the field of OpenSIPS. My company is launching an OpenSIPS deployment
to front-end all the SIP traffic entering our network. I would like
to have someone experienced look over my config to give it the
proverbial 'blessing
I will post on the list when this is available...
Thanks and regards,
Bogdan
ASHWINI NAIDU wrote:
> Thank you for considering the request.
>
> On Wed, Feb 11, 2009 at 4:21 PM, Bogdan-Andrei Iancu
> mailto:bog...@voice-system.ro>> wrote:
>
> I do agree...I already did it once, 3 years ago, on
Hi Brett,
To make both happy (generating docs from XML and also allowing users to
comment), we create special web pages for each module where to place
comments for...
Just an idea...
REgards,
Bogdan
Brett Nemeroff wrote:
> As a user speaking here, I think the docs should be wikified so people
Hi Matti,
please submit patches to the readme files that comes with the module.
Thanks and regards,
Bogdan
Matti Zemack wrote:
> Hi all,
>
>
> Is it possible to help out with minor edits of documentation? E.g.
> Examples 1.4 & 1.5 on
> http://www.opensips.org/html/docs/modules/devel/userblackli
As a user speaking here, I think the docs should be wikified so people like
me can add their experiences.
I know a lot of this is SGMLd on the backend, so I don't know how to
properly address it.
-Brett
On Wed, Feb 11, 2009 at 9:07 AM, Matti Zemack
wrote:
> Hi all,
>
>
> Is it possible to help
Hi all,
Is it possible to help out with minor edits of documentation? E.g. Examples
1.4 & 1.5 on
http://www.opensips.org/html/docs/modules/devel/userblacklist.html are
slightly minorly faulty.
And as a newbie going through documentation, I find these tiny faults here
and there now and then...
R
Thank you for considering the request.
On Wed, Feb 11, 2009 at 4:21 PM, Bogdan-Andrei Iancu wrote:
> I do agree...I already did it once, 3 years ago, on the OpenSER
> time...there was an APT repo for openserI think it will not be
> difficult to do it again.
>
> Regards,
> Bogdan
>
> Alex Bal
Yes, I have already seen that link, but I would build something more flexible
and updated with the modules of the OpenSIPS 1.5.0 versions.
Thanks.
Matteo Marzuola
>Hi Matteo
>Like this: http://www.sipwise.at/index.php/products?start=3 ?
>Sebastian
> -Original Message-
> From: users
So, I understand it work ok now, right?
Regards,
Bogdan
Mauro Davi' wrote:
> Thanks very much
>
>MD
> -Messaggio originale-
> Da: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro]
> Inviato: mercoledì 11 febbraio 2009 13:35
> A: Mauro Davi'
> Cc: users@lists.opensips.org
> Ogge
Hi Mauro,
The problem seams to be on UAS.
UAS received in contact:
Contact:
But is sends out:
Contact:
probably in the UAS cfg you do some fix_nated_contact() in onreply_route
Regards,
Bogdan
Mauro Davi' wrote:
> Hi Bogdan,
>
> In attach there are the scripts files.
>
> The lo
Hi Carlo,
On 10 Feb 2009, at 18:47, Carlo Dimaggio wrote:
>
> Il giorno 09/feb/09, alle ore 20:39, Ruud Klaver ha scritto:
>
>> Hi Carlo,
>>
>> Thanks for the extensive log information. For some reason your
>> OpenSIPS does not send the re-INVITE and the resulting 200 OK to
>> the dispatcher.
Hi Bogdan,
In attach there are the scripts files.
The load balancer route the INVITE request message to the sip server if it
received from an UAC otherwise it sent the request to the appropriate client
using a t_relay function.
All the subsequent message are routed in the loose_routed branch (
Dear Bogdan,
Thanks a lot...It's working fine
Regards
On Wed, Feb 11, 2009 at 12:13 PM, Bogdan-Andrei Iancu <
bog...@voice-system.ro> wrote:
> Hello michel,
>
> nothing is done automatically, so do you handle the OPTION requests in your
> script ?
>
> Regards,
> Bogdan
>
> michel freiha wrote:
I do agree...I already did it once, 3 years ago, on the OpenSER
time...there was an APT repo for openserI think it will not be
difficult to do it again.
Regards,
Bogdan
Alex Balashov wrote:
> It is something AG Projects helpfully does for all their stuff; well,
> MediaProxy and CDRtool for
It is something AG Projects helpfully does for all their stuff; well,
MediaProxy and CDRtool for sure. Really comes in handy. I highly
recommend.
On Wed, 11 Feb 2009 12:45:51 +0200, Bogdan-Andrei Iancu
wrote:
> Hi,
>
> I will consider setting up such a repository.
>
> Thanks and regards,
>
Hi,
I will consider setting up such a repository.
Thanks and regards,
Bogdan
ASHWINI NAIDU wrote:
> hi,
>
> For downloading deb packages this url is used. But i wanted to
> install through apt-get also. for that how can i give the repository
> address. the repository address specified in
Well, the idea is as follows - if you receive an OPTIONS point to your
server and without username, it means it needs to be handled by the
proxy and not by a user.
So, do something like this in the begining of the script:
if (is_method("OPTIONS") && uri==mysqelf && $rU==NULL ) {
s
Hello Bogdan,
No I'm not doing anything to OPTIONS packets...Any idea about what should i
do please?
Regards
On Wed, Feb 11, 2009 at 12:13 PM, Bogdan-Andrei Iancu <
bog...@voice-system.ro> wrote:
> Hello michel,
>
> nothing is done automatically, so do you handle the OPTION requests in your
> s
Hello michel,
nothing is done automatically, so do you handle the OPTION requests in
your script ?
Regards,
Bogdan
michel freiha wrote:
> Dear All,
>
> The registration is OK but the problem is that OpenSIPs does not reply
> to OPTIONS packets as you can see in the SIP trace on
> http://paste
Victor Pascual Ávila wrote:
> Bogdan,
>
> On Wed, Feb 11, 2009 at 10:27 AM, Bogdan-Andrei Iancu
> wrote:
>
>> Hi Victor,
>>
>> I think this "limitation" is part of the mechanism :).
>>
>> it is the same as for secure sip and TLS - if you get on the path a node
>> with not TLS support, the call
Bogdan,
On Wed, Feb 11, 2009 at 10:27 AM, Bogdan-Andrei Iancu
wrote:
> Hi Victor,
>
> I think this "limitation" is part of the mechanism :).
>
> it is the same as for secure sip and TLS - if you get on the path a node
> with not TLS support, the call will fail. In this case, if a hop does not
> u
Hi Adrian,
This is the part i like about SIP identity:
- it is more efficient than TLS
- it is protocol independent. With TLS you have a lot of burn with
protocol switching if you want to get some security between 2 nodes.
Regards,
Bogdan
Adrian Georgescu wrote:
> Beyond being plain int
Hi Matteo
Like this: http://www.sipwise.at/index.php/products?start=3 ?
Sebastian
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of
> mmarzu...@interfree.it
> Sent: Tuesday, 10. February 2009 18:47
> To: users@lists.
Hi Victor,
I think this "limitation" is part of the mechanism :).
it is the same as for secure sip and TLS - if you get on the path a node
with not TLS support, the call will fail. In this case, if a hop does
not understand SIP identity and changes the message, the call will be
denied.
Regard
Dear All,
The registration is OK but the problem is that OpenSIPs does not reply to
OPTIONS packets as you can see in the SIP trace on
http://pastebin.com/d20338f0d
Please check and let me know
Regards
On Wed, Feb 11, 2009 at 1:10 AM, michel freiha wrote:
> Dear All,
>
> I'm trying to regist
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