I hope this does not sound silly but what driver are you using to connect
to mssql? Is it just odbc? I am not sure opensips will support mssql.
On Nov 7, 2013 3:07 PM, "bluerain" wrote:
> Anybody tried to use the dialog module with sql server 2008 r2 back end?
>
> Opensips will start ok if I use
r wrote:
>>
>> Nothing works here. I guess the format of specifying this should be
>> changed where the username and password can be allowed to be put in single
>> quotes which sounds most appropriate for now.
>> Like you mentioned, I could change the DB password but no
OK then for sure URL encode the @ symbol. Changing password would be ideal
but not everyone has the luxury
On Oct 23, 2013 6:13 AM, "Schneur Rosenberg"
wrote:
> Didn't work for me at the time, maybe it has been fixed since
> On Oct 23, 2013 1:10 PM, "David J"
Or just do user:'p@ssword' in quotes and it should work without changing
anything
On Oct 23, 2013 6:05 AM, "Schneur Rosenberg"
wrote:
> I had this problem in the past and all I did was change my password, if
> you need to keep this password then add another MySQL user and give it
> correct permis
Maybe 2600hz
On Aug 19, 2011 11:22 AM, "Bogdan-Andrei Iancu" wrote:
> Hi Pete,
>
> I recall that; haven;t received any contact yet and do not remember
> exactly the name /company of the speaker .something with Whistle ??
>
> Maybe we can dig the ClueCon presentations from the first day.
>
> Re
Load balancing is the same in both.
What I was saying is that in some networks you may find the need to use
both. I was not suggesting that you need to do that. Each has a slightly
different subset of modules outside the that each developer made after the
split. So potentially you would have a case
Nick,
All are used; SIP-Router is really just kamailio; But in terms of
OpenSIP's vs Kamailio they are virtually the same software.
As far as performance goes; I think they are about even. You will notice
some differences for example OpenSIPs has a B2B module that you wont
find in Kamailio;
There is an opensips admin panel; Its php based; It is probably much
better supported.
I am not so sure the sermyadmin is well supported anymore;
On 8/4/11 9:12 PM, aelix systems wrote:
Hello,
I am first time user on this list.
When I try to create a "User Register" using OpenSIPS (ver: 1.6.4
11:01:07 AM bogdan_vs: Hi all
SylkServer [~sylkse...@node10.dns-hosting.info] entered the room.
(11:01:15 AM)
bogdan_vs has changed the topic to: - OpenSIPS community meeting in
progress (11:01:28 AM)
11:01:34 AM saghul: hi bogdan_vs
11:01:38 AM aidanna: howdy bogdan_vs
11:01:45 AM SylkServer:
You dont have to recompile;
Look in the docs; there is a Server header you can set;
I am not in front of it;
On 2/7/11 7:42 PM, Maciej Bylica wrote:
Hello,
Does anyone knows how to change the server header content the proxy
presents itself.
For answering the call i have:
SIP/2.0 100
I am noticing a strange behavior on the phones.
They are connecting over TCP; It seems like every 3 minutes they quickly
lose registration and then re-register.
I am not sure if it is because the phone is sending a register every 3
minutes and then opensips is closing the old socket and openi
I was wondering if it is possible to modify the SDP information a client
is sending?
I see it sends some invalid chars; If I could somehow look for that line
in the SDP and delete it would be great.
Thanks.
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I saw a new feature for detecting on hold;
Where could I see a small example of this?
Thanks
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; version='1.0'
xmlns:stream='http://etherx.jabber.org/streams'>]
On 12/13/10 7:17 AM, Anton Zagorskiy wrote:
Make logging to a file and set debug=9.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-tel
I am trying to connect to my ejabber server and getting stream errors;
I am not sure how to connect this module; I set all the parameters as
follows;
modparam("xmpp", "backend", "component")
#modparam("xmpp", "xmpp_domain", "domain.com")
modparam("xmpp", "xmpp_host","domain.com")
modparam("xmp
Ok this is a really pointless discussion; Please use Asterisk or
FreeSWITCH forum for these things. This is not a debate forum.
Thanks to everyone for thei wonderful feedback;
On 12/10/10 10:31 AM, Laszlo wrote:
Hmm, it's like Ferrari owners talking about which one is better:
Volkswagen or T
em.
>
> The only way you can make transfer work reliably is behind the same PBX.
>
> Adrian
>
>
> On Sep 15, 2010, at 6:02 PM, David J. wrote:
>
>> Seems like when I try to transfer a call from one user to the next, it
>> does not do anything, so I am guessing we
Seems like when I try to transfer a call from one user to the next, it
does not do anything, so I am guessing we have to handle the REFER message?
What is the best practice for handling REFER messages?
Thanks.
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alias module. Also, I'd like to determine if the incoming call is from
my PSTN gateway and give different aliases than if the call was a SIP
URI call.
Brett Woollum
br...@woollum.com
- Original Message -
From: "David J."
To: "OpenSIPS users mailling list"
Sen
Hi Brett,
The common practice is to use the alias module for inbound routing.
You can look at the docs for its usage, but essentially you can map
DID's to local users.
On 9/14/10 3:18 AM, Brett Woollum wrote:
Hello!
I have an OpenSIPS 1.6.3 installation that is working well. I have
subs
Yaniv,
Thanks so much for your reply, I am going to play around with your
suggestion.
On 9/5/10 11:47 AM, Yaniv Vaknin wrote:
> David,
> If I understand your question correctly, the answer is yes.
> The phone will need to subscribe first in order to get the NOTIFY.
>
> Example:
> Lets presum
:26 AM, Saúl Ibarra Corretgé wrote:
> On 09/05/2010 05:35 AM, David J. wrote:
>> Does anyone know if there is a built in way to send a NOTIFY?
>> For example if we have an external messaging source, and we want to
>> generate a NOTIFY to the UA, what is the best way to do
Does anyone know if there is a built in way to send a NOTIFY?
For example if we have an external messaging source, and we want to
generate a NOTIFY to the UA, what is the best way to do this?
Thanks in advance for any guidance.
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nd forward it to the media
server set up in that users or domains avp.
On Sep 3, 2010 2:30 PM, "David J." <mailto:da...@styleflare.com>> wrote:
Is it possible to store the users media server in an AVP and then
relay to it if the user is unavailable?
can you do a t_relay($avp(
Is it possible to store the users media server in an AVP and then
relay to it if the user is unavailable?
can you do a t_relay($avp(s:media_server));?
Or do you just have to set a pseudo var to the media server?
Thanks.
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Use
Asterisk responds.
But I guess I am wrong in my assumption.
Thanks.
David.
On 8/30/10 10:30 AM, Anca Vamanu wrote:
On 08/30/2010 05:01 PM, David J. wrote:
I am not exactly sure how this module works.
If I have Asterisk on the Same Box on Port 5070? Then is it just a
matter of
Open sips 1
I have just fixed this check.
> If you do not set the 'force_dialog' param in uac module, you must set
> the 'append_fromtag' parameter in rr module.
>
> Regards,
> Razvan
>
> On 08/31/2010 01:15 PM, David J. wrote:
>> Razvan,
>>
do not set the 'force_dialog' param in uac module, you must set
> the 'append_fromtag' parameter in rr module.
>
> Regards,
> Razvan
>
> On 08/31/2010 01:15 PM, David J. wrote:
>> Razvan,
>>
>> Sorry about the trouble; here is the casue.
&
double check you are using the correct configuration
> file.
>
> Regards,
> Razvan
>
> On 08/30/2010 07:44 PM, David J. wrote:
>> Razvan,
>>
>> I have set beug to 6 but the logs are just as verbose. It seems to
>> output the same information.
>&g
ply the new log.
>
> Regards,
>
> On 08/30/2010 01:21 AM, David J. wrote:
>> I saw a post today regarding setting callerid to Anonymous, I was
>> interested in testing the code; so I loaded the UAC module in and
>> restarted opensips;
>>
>> Not much inf
I am not exactly sure how this module works.
If I have Asterisk on the Same Box on Port 5070? Then is it just a
matter of
Open sips 10.10.10.10:5060
Asterisk 10.10.10.10:5070
Changing this line?
modparam("presence", "server_address", "sip:10.10.10.10:5060")
modparam("presence", "server_ad
I saw a post today regarding setting callerid to Anonymous, I was
interested in testing the code; so I loaded the UAC module in and
restarted opensips;
Not much information why it crashed...
The only thing I added to my existing opensips.cfg is include "uac.so";
Then restarted. (If I remove
Any idea how to Install Frontier from Cpan?
I am trying to run osipsconsole but get the following messages.
Can't locate Frontier/RPC2.pm in @INC (@INC contains: /etc/perl
/usr/local/lib/perl/5.10.0 /usr/local/share/perl/5.10.0 /usr/lib/perl5
/usr/share/perl5 /usr/lib/perl/5.10 /usr/share/per
e heads up.
David.
On 8/16/10 6:01 AM, Bogdan-Andrei Iancu wrote:
> Hi David,
>
> Are you sure you correctly provisioned the aliases ? could you check the
> content of the "dbaliases" table ?
>
> Regards,
> Bogdan
>
> David J. wrote:
>>I am playing a
username/domain.
- Jeff
On Aug 13, 2010, at 8:08 AM, David J. wrote:
Thx...
Any idea what the purpose of the field is?
On 8/13/10 6:39 AM, Pasan Meemaduma wrote:
Hi David,
UUID field is not mandatory. you can just insert other fields
without UUID.
It could use as a reference
I am playing around with the LIVE VM, I tried adding a DID alias to
the database, but it seems when I send an INVITE to the Alias I get "No
pstn calls." back. Shouldn't the script know to use usrloc rather than
dialplan if the alias is found?
Thanks.
__
|
*From:* David J.
*To:* OpenSIPS users mailling list
*Sent:* Friday, August 13, 2010 15:57:28
*Subject:* [OpenSIPS-Users] UUID value in usr_preferences
List,
what value should I be using in the UUID field in the
List,
what value should I be using in the UUID field in the usr_preferences table?
I was trying to manually insert into the table, and got stumbled by this
field, the rest seem to make sense.
Thx.
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h
Turns out vmware is not free for Mac. Guess I have 30 days to play
before trial runs out.
Maybe we can make a livecd instead not vmware based?
On 8/3/10 11:46 PM, Bogdan-Andrei Iancu wrote:
> Hello all,
>
> The first OpenSIPS Virtual Machine is now available - the VM is a free
> show case of
Bogdan,
Looks great thanks again.
Will have to get VMware to play.
Thx.
On 8/3/10 11:46 PM, Bogdan-Andrei Iancu wrote:
> Hello all,
>
> The first OpenSIPS Virtual Machine is now available - the VM is a free
> show case of a simple SIP provider setup - a ready-to-run run SIP
> platform with
Try another machine, with Debian - What kernel are you running?
I have stable on debian, centos, and Fedora core 8
Hope that helps.
On 8/2/10 7:26 AM, Yaniv Vaknin wrote:
> Are you sure ?
> It happen on both clean Red-hat server and Gentoo server...
>
> Yaniv
Yaniv,
Try re-installing your OS!
On 8/2/10 7:20 AM, Yaniv Vaknin wrote:
> Hi,
> Just wanted to update, I've upgrade the server to version 1.6.3, the server
> still crashes almost every hour.
> Something new that I've noticed, now I get 2 or 3 core file when the server
> crashes.
>
> Is ther
It seems like for quite some time alias_db does not support domains, but
is listed to use domains in the default script.
# - multi-module params -
/* uncomment the following line if you want to enable multi-domain support
in the modules (dafault off) */
#modparam("alias_db|auth_db|usr
Let me ask a foolish question then,
what do we do for logging?
Meaning what is the equivalent?
Thanks.
On 7/11/10 3:53 AM, Bogdan-Andrei Iancu wrote:
> Hi David,
>
> As the xlog module was removed, after doing an svn update, just remove
> "modules/xlog"
>
> R
make[1]: Entering directory `/usr/src/opensips.trunk/modules/xlog'
make[1]: *** No targets specified and no makefile found. Stop.
make[1]: Leaving directory `/u
I tried this on 2 different machines, ran make clean first as well as
make proper.
Thanks.
__
Bogdan,
Not a problem, I was just wondering about it...
Like I said it probably makes no difference.
On 7/6/10 4:16 PM, Bogdan-Andrei Iancu wrote:
> My bad!
>
> It seams the trunk still have in Makefile.defs the 1.6.1 version :PI
> will fix it.
>
> Regards,
> Bogd
994 (1.6 branch at it compiles with
> -DVERSION='"1.6.2-notls"'
>
> Try to do in your sip_server checkout dir:
> make proper
> svn update
> make all
>
> Regards,
> Bogdan
>
> David J. wrote:
>
>> I have a SVN t
I have a SVN trunk copy running r6990
Why when I build do I see
-DVERSION='"1.6.1-notls"
This trunk was created back by v 1.6.1, however , it has been since
updated to latest (r6990 as of writing).
Thanks in advance for any clarification.
___
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Dovid,
How do you do that using AMI?
On 6/23/10 4:39 PM, Dovid Bender wrote:
> - Original Message -
> From: "Daniel Knoll"
> To:
> Sent: Sunday, June 20, 2010 08:57
> Subject: [OpenSIPS-Users] Load balancing Conference System
>
>
>
>> Hello Group,
>> I have an Conferencing System (A
On Fri, May 21, 2010 at 11:00 AM, David J. <mailto:da...@styleflare.com>> wrote:
Daniel,
I am calling fix_nated_register() on REGISTER and on INVITE
fix_nated_contact()
On 5/21/10 1:46 PM, Daniel Goepp wrote:
Ah yes, are you also doing NAT detection and fixing co
Daniel,
I am calling fix_nated_register() on REGISTER and on INVITE
fix_nated_contact()
On 5/21/10 1:46 PM, Daniel Goepp wrote:
Ah yes, are you also doing NAT detection and fixing contact?
-dg
On Fri, May 21, 2010 at 10:40 AM, David J. <mailto:da...@styleflare.com>> wrote:
you are.
-dg
On Fri, May 21, 2010 at 9:05 AM, David J. <mailto:da...@styleflare.com>> wrote:
Daniel,
You said you got opensips/rtp-proxy working on EC2, when testing
it, I see the VIA headers have the EC2 internal ip and the BYE
messages not getting routed correctly, a
Daniel,
You said you got opensips/rtp-proxy working on EC2, when testing it, I
see the VIA headers have the EC2 internal ip and the BYE messages not
getting routed correctly, any advice on how to solve that?
Thanks
On 5/21/10 12:02 PM, Daniel Goepp wrote:
You need to be careful though becau
Stefan,
I had a similar question, we don't play ads to the caller, but we do
locate the callee while the caller is waiting or "on hold", so we could
use the B2BUA functionality.
Is there an example available of this functionality in the SEMS examples?
(Sorry to post a SEMS question on the Open
Sorry, The way I recommend doing this was assuming the user on the
Asterisk box needed to be publicly reachable from anywhere.
I think that approach makes sense when using DID's and inbound routing
that does need authentication.
On 5/4/10 12:55 PM, Olle E. Johansson wrote:
> 4 maj 2010 kl. 18
Check in the SIP.conf where you send all unauthenticated calls.
On 5/4/10 11:45 AM, wüber wrote:
> The problem seems to be not only in the extensions.conf file, but also in the
> sip.conf file.
> I still get this forbidden message!
>
___
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You need to add a route in your extensions.conf in the context where you
send all un-authenticated calls.
Maybe its your default context?
[default]
exten = 1001,1,Dial(SIP/1001,10,tTr);
On 5/4/10 9:02 AM, wüber wrote:
> Hi Bogdan,
>
> connecting Opensips with Asterisk I can see that if a cl
Paul,
Please re-read the documentation again.
I've posted my global.inc in another mail. I'd appreciate some more
experienced eyes on this.
Thanks.
On 4/27/10 10:05 PM, Paul Wise wrote:
On Tue, 2010-04-27 at 10:41 +0200, Adrian Georgescu wrote:
I have no clue how you arrive at this
n the next days.
>
> Regards,
> Bogdan
>
> David J. wrote:
>
>> I dont mean to keep asking, but is there an upload from the last webinar?
>>
>>
>>
>>
>> ___
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>> Use
I dont mean to keep asking, but is there an upload from the last webinar?
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I was looking at the registrar documentation and not sure if this is
in-consistent documentation.
/flags/ (optional)- string of the following flags:
/'cnn' (max Contacts)/ - this flag can be used to limit the number of
contacts for this AOR (Address of Record) in the user location database.
V
Any idea about when last weeks webinar will be posted?
Thanks.
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Just curious, Whats the sense of this?
Christian Vo wrote:
> Trying to manually install SerMyAdmin (step by step , since install
> script uses apt-get) on my OpenSUSE 11 setup..
> Which already has OpenSIPs-CP installed…
> I’m noticing that the sql script has several conflicts… anyone try
> this
Matt,
I am for sure probably wrong, but I think you would need Asterisk or
Variant to Determine that it is a Fax Call,
I dont think UAC's send T38 information without negotiating with the
other side who request that it is capable, then it brings you to Jeff's
answer.
See above.
Matthew S. Cr
ex: opensipsctl db show subscriber
Jan Rozhon wrote:
> Definitely sureit's the only one I have.
>
> Dne 5.3.2010 16:44, David J. napsal(a):
>
>> And your sure your checking the same database your script is pointing to?
>>
>>
>>
>> Jan Rozhon
Yikes! Can you run any of the other opensipsctl commands on that database?
Or do they fail as well?
Jan Rozhon wrote:
> Definitely sureit's the only one I have.
>
> Dne 5.3.2010 16:44, David J. napsal(a):
>
>> And your sure your checking the same database you
And your sure your checking the same database your script is pointing to?
Jan Rozhon wrote:
> No, user in database - checked twice. Problem is, that I cannot add any
> user
>
>
> Dne 5.3.2010 16:39, David J. napsal(a):
>
>> Where is the error?
>>
>> C
Where is the error?
Check your database and make sure you dont have an entry for that user.
Jan Rozhon wrote:
> Hi list,
>
> I am trying to set up and run opensips with berkeley module. After
> configuration, and creating of database I get this error:
>
> chieftec:~# opensipsctl add 1234 1234
>
Brett,
From a user perspective then its not so difficult to learn a few extra
commands.
Most users I would imagine want to download the code and compile.
Although, I agree that SVN is by far more adopted.
I think going forward, companies like GitHub and BitBucket offer great
code hosting opti
nge this you will have to update your tables
>
> David J. wrote:
>
>> Sounds good, Do we need to update the DB Schema or if the column is
>> missing it fails gracefully?
>>
>> Thanks.
>>
>>
>> Andrei Dragus wrote:
>>
>>
>
Sounds good, Do we need to update the DB Schema or if the column is
missing it fails gracefully?
Thanks.
Andrei Dragus wrote:
> Hi all,
>
> A new feature has been added to the drouting module that allows the
> probing of gateways in case they become unreachable.
>
> Typical usages:
Where is the best place to use the append_rpid_hf()?
It seems I add it once, but subsequent messages dont include the header.
Perhaps I have to add it for each request?
Thanks.
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what is responsible for using the external ip for the record route?
Right now cancels are not being routed properly.
It seems from the SIP trace that we are using the natted ip rather than
the public.
However the box does not have a public ip, we are port forwarding from
our router to our opens
RTP Proxy wont help either, It is also just a media relay.
wüber wrote:
> Perhaps you can try RTPproxy ... I'm not so sure but I guess it should
> support transcoding. Regards
>
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MediaProxy is a separate module for Media Relay also has nothing to do
with transcoding.
Transcoding is a conversion process from one codec to another, maybe you
can try using asterisk if it supports the codecs your are trying to
transcode.
OpenSIPS is a sip proxy, media proxy is a media relay
I am not the expert, but I dont think OpenSIPS can be used as a transcoder.
It acts only as a signaling proxy, media is not passed through opensips,
just the SDP information.
But I am not the expert.
Live School wrote:
> Hi,
>
> is it possible to use openSIPS as transcoder only ?
> if yes,
Pardon my ignorance, but what exactly is the problem?
Your customer does not need to register to you, however, then you need
to have a static ip address to route to.
and vice versa. When a customer sends an INVITE to your opensips, you
need to authenticate by IP address.
The sipXecs does not ne
Changing the mask does resolve the problem.
Thank you.
Bogdan-Andrei Iancu wrote:
> The mask is not correct - it must be less than 32 , the max len of an
> IPv4 address.
>
> Regards,
> Bogdan
>
> David J. wrote:
>
Will try to update,
However, it was added through opensipsctl address add
I didnt manually insert the data.
Bogdan-Andrei Iancu wrote:
> The mask is not correct - it must be less than 32 , the max len of an
> IPv4 address.
>
> Regards,
> Bogdan
>
> David J. wrote:
>
ase post here
> the content of the address table (do a select * ).
>
> Regards,
> Bogdan
>
> David J. wrote:
>
>> MySQL version 5.0.51a
>>
>>
>>
>>
>> Bogdan-Andrei Iancu wrote:
>>
>>
>>> Hi David,
>>
MySQL version 5.0.51a
Bogdan-Andrei Iancu wrote:
> Hi David,
>
> what db backend are you using for permissions module?
>
> Regards,
> Bogdan
>
> David J. wrote:
>
>> I am running SVN Trunk and getting the following Error.
>>
>> /sbin/opensips[
I am running SVN Trunk and getting the following Error.
/sbin/opensips[1923]: ERROR:permissions:reload_address_table: database
problem
/sbin/opensips[1923]: CRITICAL:permissions:init_address: reload of
address table failed
/sbin/opensips[1923]: ERROR:permissions:mod_init: failed to initialize
t
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