I had a similar problem happen behind a certain router it was the routers
sip alg, try to do a wireshark behind the router if it has proper invite
try disabling sip alg, if not possible change router
On Nov 7, 2012 8:16 PM, "Bogdan-Andrei Iancu" wrote:
> Hi,
>
> See in the logs:
>
> Nov 7 08:58
Are they both using same database? I use master master replication and it
works fine for me.
On Jan 28, 2013 12:36 PM, "microx" wrote:
> Hi all,
>
> In my scenario, I have two proxy servers PA and PB and a number of RTP
> proxies. Basically, PA is active and PB is stand-by.
> For each RTP proxy,
Hi I'm trying to change the ip part of the FROM header, is there a
variable that I can use that has the value of the IP that will be used
to send this packet?
Here is what I'm trying to do
uac_replace_from("sip:$fU@70.97.189.150");
but I want the ip address 70.97.189.150 to be replaced by a variab
Thanks, sound right i will try it soon.
On Tue, Mar 5, 2013 at 8:16 AM, Nick Altmann wrote:
> You may want to use $Ri variable. It contains opensips IP address where
> request was received.
>
> --
> Nick
>
>
> 2013/3/5 Schneur Rosenberg
>>
>> Hi I'm
I have a similar setup and I use the full URI for incoming calls, so
lets say the OpenSIPS server is at sip1.mycarrier.com and I want to
send the call to a sip user called 101 then I send the call to
1...@sip1.mycarrier.com
On Sun, Mar 10, 2013 at 4:04 AM, Nick Khamis wrote:
> Hello Everyone,
>
>
Thats why I do a avp_db_query on the location table on each invite to
check if the caller exists is the db, and if found system knows its a
outgoing call, and if not found then it knows its a incoming call,
there is a performance penalty on doing the queries but I had no
choice.
If you only get in
Hi, I'm trying to pull the active calls from the dialog table to output on
a client portal, I would like to know if I can set some kind of flag in my
script that will change a field in the db table, so I can do a query on
that field.
___
Users mailing lis
DB record also; you'll want
> to test that one if you think it might be useful for you.
>
>
> - Jeff
>
>
>
> On Tue, Oct 22, 2013 at 1:12 PM, Schneur Rosenberg <
> rosenberg11...@gmail.com> wrote:
>
>> Hi, I'm trying to pull the active calls fro
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
permissions and use that in your script
On Oct 23, 2013 1:01 PM, "Jayesh Nambiar" wrote:
> Yes, still doesn't work.
>
> --- Jayesh
>
>
> On Wed, O
Didn't work for me at the time, maybe it has been fixed since
On Oct 23, 2013 1:10 PM, "David J" wrote:
> 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:
&g
Install ncurses
On Nov 8, 2013 12:16 PM, "Vishnu Vardhan" wrote:
> Hi,
>
> When i am trying to configure Opensips in Centos 6.4 i am getting this
> error messgae in the process of mysql DB creation pls see the follow
> message and pls help me to get out of this.
>
> make[1]: Entering directory `/
If OpenSIPS is set to use mysql then yes its a reason not to start
On Feb 8, 2014 11:53 PM, "Freddi Guerrero" wrote:
> Thanks for responding and for the help. I am checking on this as well,
> would this be a reason for opensips service not starting?
>
>
> -Original Message-
> From: users
It would be nice if you give us the problem and the solution.
On Mar 25, 2014 1:49 PM, "dpa" wrote:
> It seems I found the problem myself.
>
> Thank you
>
>
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *dpa
> *Sent:* Tuesday, March 25, 20
Hi Bogdan and the OpenSIPS community
I currently use OpenSIPS for loadbalancing and registrations, all the rest
of the call handling is handled by a cluster of Asterisk servers, I
currently have 3 Opensips servers 2 are located in same DC and are in a
active/passive Master/Slave mode via Hertbeat
1) Install ngrep
2) run "ngrep -q -W byline port 5060 > dump.txt"
After the call press + c to stop the ngrep and thes dump should be
in the file dump.txt, assuming that there is no other traffic it should
contain only the dialog he wants
On Apr 25, 2014 1:35 AM, wrote:
> What command should I
Try sipsak
On Apr 25, 2014 11:43 AM, "Maksim Solovjov"
wrote:
> Hello,
>
> I am using OpenSIPS server and pjsip library for a VoIP implementation.
> The thing is, that sometimes my web server ( not the sip server, but a
> web server designed for the application ) needs to send an instant
> messag
You missed the fifo
On Apr 27, 2014 3:11 AM, wrote:
> Help says it's opensips domain add.
> This is what I get:
>
> # opensipsctl domain add 10.10.10.3
> INFO: execute '/sbin/opensipsctl domain reload' to synchronize cache and
> database
> [root@lion opensips]# opensipsctl domain reload
> 500 co
tion working?
> Did you try to add a user? It worked?
>
>
>
> Il 28/04/2014 11.47, Schneur Rosenberg ha scritto:
>
> You missed the fifo
> On Apr 27, 2014 3:11 AM, wrote:
>
>> Help says it's opensips domain add.
>> This is what I get:
>>
>
sorry mispelled try "opensipsctl fifo domain_reload"
On Tue, Apr 29, 2014 at 1:30 AM, Schneur Rosenberg wrote:
> opensipsctl fifo domain reload
> On Apr 28, 2014 11:31 PM, wrote:
>
>> No, nothing working, still the same error.
>> And what does it mean that I m
vailable
>
> Thank you!
>
> On 04/28/2014 06:35 PM, Schneur Rosenberg wrote:
>
> sorry mispelled try "opensipsctl fifo domain_reload"
>
>
> On Tue, Apr 29, 2014 at 1:30 AM, Schneur Rosenberg <
> rosenberg11...@gmail.com> wrote:
>
>> opensipsctl
The way I did it, was to create a table of the ip's and which group it
should use, every call does a db query, it may put a strain on the db if
there is lots of calls, let me know if you have a better solution
On May 29, 2014 10:28 AM, "Muhammad Naseer Bhatti"
wrote:
>
> Hi, I am using load balan
I'm new to opensips, I currently have a asterisk server with over 300
users and at times it crashes from too many things going on at the
same time, therefore I would like to set up a opensips box to handle
registrations and load balancing for our asterisk servers, here is my
plan let me know if thi
I'm pretty new to opensips, I'm having a interesting problem, I use my
opensips for loadbalancing purposes I'm trying to place a call, and
from My linksys phone everything works fine, call comes into opensips
and opensips sends it to my asterisk system and call goes through
properly, from other pho
ensips.cfg does the exact same thing for linksys and
> aastra phones I can't see it being an opensips issue. That's just a guess
> since I don't have anything to go on.
>
> On Wed, Sep 21, 2011 at 4:25 PM, Schneur Rosenberg
> wrote:
>>
>> I'm pretty new
e answer to a different IP. Also enable the debug
> log on the asterisk console to spot any error / warning messages or sip
> retransmissions.
>
> Hope this would help.
>
> Regards,
> -vma
> .
>
> On Sep 21, 2011, at 11:43 PM, Schneur Rosenberg wrote:
>
>>
If the packet would of reached asterisk then you might of been right,
problem is a ngrep trace does not show a single packet reaching it.
On Thu, Sep 22, 2011 at 1:09 AM, Brett Nemeroff wrote:
> On Wed, Sep 21, 2011 at 5:03 PM, Vallimamod ABDULLAH
> wrote:
>>
>> Hi Schneur,
>>
>> What do you mea
oes filtering
> or mangling in some way…
> Try to trace the sip packet on every hop between the 2 servers to see how far
> it goes.
>
> Regards,
> - vma
> .
>
> On Sep 22, 2011, at 12:07 AM, Schneur Rosenberg wrote:
>
>> The packet does not reach asterisk, I did a ng
We have a Opensips server that is used to load balance a few asterisk
servers, the opensips also handles registration, but asterisk handles
everything else, everything works fine but I don't know how to get MWI
indicator to work, I tried rewritehostport for the SUBSCRIBE but it
did not work, can an
t_reply("500", "Server internal error");
> }
> exit;
> }
> }
> }
>
>
> Dani
> On 10/27/11 17:34, Schneur Rosenberg wrote:
>>
>> We have a Opensips server that is used to load balance a few asterisk
>>
I'm trying to use load balancing, but I have a question I set the
probing mode on 2, now my question is will opensips automatically
disable the route if it does not probe or I need to do it manually
with lb_disable() ?
___
Users mailing list
Users@lists.
Hi
I'm not yet experienced enough in opensips, so please bear with me.
I set up a opensips to loadbalance 2 asterisk servers, in the failure
route I placed
if (t_check_status("(408)") && t_local_replied("all"))
{
lb_disable();
# Try to load balan
ode-1-bug-td4245085.html#a5715729
>
> Put this before your first IF statement and then try again
>
> xlog("L_INFO", "-- BEFORE LB IF Statement: Call [$rm] ru[$ru] fu[$fu]
> si[$si] \n");
>
>
>
> On , Schneur Rosenberg wrote:
>> Hi
>>
>>
//www.opensips.org/html/docs/modules/devel/tm.html#id250308>
>
> Regards,
>
> Vlad Paiu
> OpenSIPS Developer
>
>
> On 11/08/2011 12:13 AM, Schneur Rosenberg wrote:
>
>> Ok it was working it just took a very long time until it timed out.
>>
>> thank you
&
I'm using opensips as a loadbalancer for 2 asterisk servers, almost
every day the system stops accepting new calls and when I run
"opensipsctl fifo lb_list" it says "Resource:: pstn max=50 load=50"
for both servers, even though there is no calls on either system, how
can I know whats causing the sy
How is your modparam("dialog", "db_mode", ?) set up?
>
> Just wondering if you have some stall dialogs and the LB module still thinks
> there are active calls.
>
> Just guessing.
>
>
>
> On , Schneur Rosenberg wrote:
>> I'm using opensips as a l
dialogs how many LB resources are currently being used?
>
>
>
> On , Schneur Rosenberg wrote:
>> I will have to try that next time it happens, the problem is I hope
>>
>> there wont be a next time, I need it fixed before next time :-)
>>
>> th
, Nov 15, 2011 at 12:49 AM, Schneur Rosenberg
wrote:
> 8
>
> On Tue, Nov 15, 2011 at 12:47 AM, wrote:
>> So if you show 177 dialogs but you know for a fact that there are really
>> only 2 active calls then you have a lot of stale dialogs. I believe you are
>> probably no
then the asterisk boxes are kept
> out of the equation.
>
>
>
>
>
> On , Schneur Rosenberg wrote:
>> I think I see the problem but I don't know how to fix it, my system
>>
>> works as follows, opensips is used as a registrar and as a load
>
In my case this is not relevant, because I'm calling the other phone
through a DID and the did needs to go to asterisk to decide what to do
with it, it can send it to a IVR which can later send it to Opensips
etc. in any case I need to know why asterisk is not sending the BYE to
the phone, and why
.116:1047
SIP/2.0 200 OK.
U asterisk2IP:5060 -> opensipsIP:5060
BYE sip:solhome7@93.172.0.116:5060;nat=yes SIP/2.0.
.
U opensipsIP:5060 -> asteriskIP:5060
SIP/2.0 404 Not here.
On Tue, Nov 15, 2011 at 2:19 AM, wrote:
> Could you provide a sip trace of a call from INVITE to BYE? Als
else
{
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
On Tue, Nov 15, 2011 at 3:00
I might be missing a record route, would that cause this problem?
On Tue, Nov 15, 2011 at 3:16 AM, Schneur Rosenberg
wrote:
> Here is the code sending the 404 not here, I dont understand why
> if(loose_route()) does not return true, is this the way its supposed
> to be?
>
>
lhome7@93.172.0.116:5060;nat=yes SIP/2.0.
>
> Does OpenSIPS know of a user named solhome7@93.172.0.116? Since that is all
> that is in the SIP message that is all I have to go by. I also see that
> there are devices called solhome7, solhome3 and solhome5
>
>
> On Mon, Nov 14,
OpenSIPS know of a user named solhome7@93.172.0.116? Since that is all
> that is in the SIP message that is all I have to go by. I also see that
> there are devices called solhome7, solhome3 and solhome5
>
>
> On Mon, Nov 14, 2011 at 7:00 PM, Schneur Rosenberg
> wrote:
>>
&
I just did a test, and sure enough the it fixed the dialog problem too.
On Tue, Nov 15, 2011 at 3:31 AM, Schneur Rosenberg
wrote:
> Yes opensips knows of all the users that asterisk knows, they share
> the same database, I think my problems was a missing record_route(),
> thanks g
I never really understood how the session expires works and whats the
difference between UAC or UAS, who is supposed to send the reinvite?
etc, if anyone can provide me with a simple explanation I would really
appreciate.
I'm getting a call from a DID which has Expires: 240. Min-SE: 1200 in
the si
Thanks Joel
On Nov 15, 2011 8:28 PM, "joel.oliveira"
wrote:
> On a side note it's important to know the difference between Min-SE,
> Expires
> and Session-Expires headers.
>
> A quick, and maybe naive point of view, is:
> . Min-SE : minimum Session-Expires value that should be use while
> negotia
; would just ask him directly?? :)
>
> If I was you I would reproduce the issue and do a sip trace of the whole
> call and see whats going on.
>
>
>
> On , Schneur Rosenberg wrote:
> > I never really understood how the session expires works and whats the
> >
> >
I'm using Opensips as a Load balancer and as a registrar, so basically
all phones are registered to the Opensips, all Incoming calls hit the
opensips server which forwards the call to asterisk with load
balancing, asterisk decides what to do with the call ie IVR voicemail
etc and if the call needs
Bogdan there is too little info about this online, can you please help
me a bit more with this, how do I write the if statement, and how do I
set a variable for the first call, and how do I retrieve which server
was used for the first call.
On Wed, Nov 23, 2011 at 9:46 PM, Schneur Rosenberg
wrote
Did u do record_route() on initial invite?
On Dec 13, 2011 8:02 AM, "Nick" wrote:
> Hello
>
> I used ngrep .
> U 220.130.6.180:55260 -> 192.168.20.118:5060
> BYE sip:81@220.130.6.180:17882 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.20.153:55260;branch=**z9hG4bK1489712528;rport.
> From: ;tag=**1735203887
I'm using opensips on a computer with 2 ip addresses one steady one
and one is a floating ip address provided by heartbeat, when heartbeat
is on and I have 2 ip addresses opensips takes a very long time to
start and I get a error in the messages file, the error is
opensips: WARNING:core:fix_socke
c 15, 2011 at 2:52 PM, Schneur Rosenberg
> wrote:
>> I'm using opensips on a computer with 2 ip addresses one steady one
>> and one is a floating ip address provided by heartbeat, when heartbeat
>> is on and I have 2 ip addresses opensips takes a very long time to
>>
I have multiple domain names going to the same server, right now if I
don't have the domain field in the subscriber table set to the domain
I'm using to register it wont authenticate, because is_from_local()
will be false, how can I change that, that as long as the domain
points to my ip address i
I have multiple domain names going to the same server, right now if I
don't have the domain field in the subscriber table set to the domain
I'm using to register it wont authenticate, because is_from_local()
will be false, how can I change that, that as long as the domain
points to my ip address i
yes
On Mon, Dec 19, 2011 at 8:55 AM, Sammy Govind wrote:
> Not sure, just a thought- have you added those domains in domain table in
> opensips DB ?!!
>
> On Mon, Dec 19, 2011 at 11:41 AM, Schneur Rosenberg
> wrote:
>>
>> I have multiple domain names going to the
.
in my debug I get the following 2 lines
Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]:
DBG:db_mysql:db_mysql_convert_rows: no rows returned from the query
Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]:
DBG:auth_db:get_ha1: no result for user 'gra...@sipsvr1.mydomain.com'
ant install why don't you want to
> input the domain info in the subscriber table?
>
> On Dec 19, 2011 11:15 AM, "Schneur Rosenberg"
> wrote:
>>
>> here is my problem, my server has multiple domain names, different
>> customers register with a different do
info in the subscriber table?
>
> On Dec 19, 2011 11:15 AM, "Schneur Rosenberg"
> wrote:
>>
>> here is my problem, my server has multiple domain names, different
>> customers register with a different domain name (we combined a few
>> servers into one, therefore s
My opensips server is not behind nat and is used as a load balancer
and as a registrar, but all RTP passes through my asterisk servers
which is not behind nat, do I need a proxy like rtpproxy? or is it
enough that everything runs through asterisk which is on the open
internet?
my
>
> On Wed, Dec 21, 2011 at 5:01 PM, Schneur Rosenberg <
> rosenberg11...@gmail.com> wrote:
>
>> My opensips server is not behind nat and is used as a load balancer
>> and as a registrar, but all RTP passes through my asterisk servers
>> which is not behind nat, do I n
Is there a way to reboot phones, similar to asterisk "sip notify" command?
___
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Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
The Contact header is missing the closing bracket, it happens to all
phones at this customers location, that's why I believe its a broken
ALG in the router, I dont have access to this router, I tried
registering to Asterisk and Asterisk finds the error, but says trying
anyway and everything runs fi
missing bracket and try to register this way.
On Wed, Jan 4, 2012 at 4:07 PM, Brett Nemeroff wrote:
>
>
> On Wed, Jan 4, 2012 at 7:16 AM, Saul Ibarra Corretge
> wrote:
>>
>>
>> On Jan 4, 2012, at 12:21 PM, Schneur Rosenberg wrote:
>>
>> > The Cont
Hi, I have a openSIPS server setup to do registration and load
balancing between 2 Asterisk servers, the Asterisk servers do
everything besides registration and they are load balanced by the
openSIPS servers, incoming calls hit the openSIPS server which sends
it to the Asterisk server and if it nee
sips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg
> Sent: Monday, January 30, 2012 15:07
> To: OpenSIPS users mailling list
> Subject: [OpenSIPS-Users] No audio on some routers with PAP2T
>
> Hi, I have a openSIPS server setup to do registration
Hi I have a Opensips server that handles registration and
loadbalancing, it load balances a few asterisk servers, I was having
issues with transfers that if each leg is on a different server it
would hangup, so Bogdan suggested I use dialogs and when opensips sees
a call on a extension, next time i
erisk still involved in call?
>>
>> I'm asking as I'm not so familiar on how the parking is done with Asterisk.
>>
>> Regards,
>> Bogdan
>>
>> On 04/02/2012 05:09 PM, Schneur Rosenberg wrote:
>>> Hi I have a Opensips server that handles re
My phones are registered to opensips, the Voicemail is handled by
asterisk, Asterisk has a list of all sip devices, but it has no idea
if phones are connected or not, I would like to get the MWI from
asterisk sent over to the opensips, so opensips can send it to the
phone.
Is there anyway to have
>> regularly.
>>
>> Best Regards,
>> - vma
>> .
>>
>>
>>
>> On May 14, 2012, at 9:02 PM, Schneur Rosenberg wrote:
>>
>>> My phones are registered to opensips, the Voicemail is handled by
>>> asterisk, Asterisk has a list of
notify with sipsak
> (http://www.voip-info.org/wiki/view/Asterisk+Realtime+MWI+Hacks) ?
>
> This hack does not handle the reboot case. You can eventually setup a cron
> job that checks for new voicemails and sends the corresponding notify
> regularly.
>
> Best Regards,
> - v
I use opensips to send calls via Asterisk, I share the sip usernames
and passwords from opensips with Asterisk, and thats how Asterisk
knows what context, caller id etc to use.
everything works fine, the only issue is when doing a Blind Transfer
OpenSIPS sends a REFER to asterisk, but for some rea
setup. We
> would need more information.
>
>
> On , Schneur Rosenberg wrote:
>> I use opensips to send calls via Asterisk, I share the sip usernames
>>
>> and passwords from opensips with Asterisk, and thats how Asterisk
>>
>> knows what context, caller id
ce again this totally depends on your OpenSIPS and Asterisk setup. We
>> would need more information.
>>
>>
>> On , Schneur Rosenberg wrote:
>>> I use opensips to send calls via Asterisk, I share the sip usernames
>>>
>>> and passwords from opensips
I'm trying to manipulate MWI with sipsak, problem is that I get a
SIP/2.0 483 Too Many Hops, I'm probably sending the packet wrong, here
is the packet and response.
U 64.69.27.120:57036 -> 75.98.168.213:5060
NOTIFY sip:grattcis...@sip.mytcm.com SIP/2.0.
Via: SIP/2.0/UDP 64.69.27.120:57036;branch=z
rg/html/docs/modules/1.4.x/maxfwd.html#id227234
>
> Before the sl_send_reply("483","Too Many Hops"); in your code do an xlog and
> print out $retcode to see what its value is.
>
>
>
>
>
>
> On , Schneur Rosenberg wrote:
>> I'm trying to
our opensips until the Max-Forward reaches 0 ?
> If so, opensips doesn't recognize the request as local so it tries to forward
> it (to himself.)
> Check if the domain sip.mytcm.com is actually declared in the domain table.
>
> Best Regards,
> - vma
> .
>
>
>
&g
Thanks Vallimamod Abdullah I used sipsak and I rewrote the script
because I use ODBC storage for my voicemail, and the script is meant
for file storage, and it works like a charm.
On Tue, May 15, 2012 at 3:04 AM, Schneur Rosenberg
wrote:
> I didn't try sipsak , but I will try, I will p
I'm using OpenSIPS to load balance multiple Asterisk servers, all
phones are registered to OpenSIPS, and Asterisk shares the subscriber
table, every INVITE gets sent to asterisk and when Asterisk sees the
invite it recognizes the user and sends call accordingly, (call plan,
caller id etc).
Everyth
Nothing is stopping you from creating new tables
On Sun, Jun 17, 2012 at 9:55 AM, prasad kelkar wrote:
> hello,
> I want to create new table "rtp" with two coloums in opensips database.
> please help
>
>
> ___
> Users mailing list
> Users@lists.opensips
I'm using opensips as a registrar server and as a loadbalancer, all
phones are registered to opensips and all incoming and outgoing calls
go to Asterisk boxes via load balancing, therefore I have 3 kinds of
calls going to opensips,
1) outgoing calls coming from one of the phones Registered to opens
.x/permissions.html#id293503
>
> Look at the "check_source_address" and or "get_source_group". Either of
> these can compare the source IP of the originator to a known list. From
> there, you can perform script logic based on where the request came from.
>
> H
;
> if ($rU=~"^\+?[0-9]{3,18}") {
> # request-uri is for a PSTN number, send the message to whatever
> route(1)
> }
>
> Basically you need to find a difference between the call attributes and
> examine that, it can be the src_ip, ruri pattern, etc.
>
> Regards,
&g
ocs/modules/1.8.x/registrar.html#id293162
>
> Regards,
> Ali Pey
>
>
> On Wed, Jul 25, 2012 at 4:23 PM, Schneur Rosenberg
> wrote:
>>
>> I already did something similar look at snippet bellow so any call
>> coming from a IP thats registered to our server wil
ining usernames'
> patterns can be faster.
>
> You can also use the registered function instead of a db query:
>
> if (registered("location","$fu")) {
> xlog("caller is registered\n");
> }
>
> http://www.opensips.org/html/docs/modules/1.8.x/
ss table and keep track of who is allowed and block any other
> requests. There won't be any script change or reload required. A new pbx
> would require a new ip address in the table and a reload command.
>
> Regards,
> Ali Pey
>
> On Mon, Jul 30, 2012 at 7:39 PM, Schneur R
I would like to send calls to a user that is registered to opensips,
the user is a PBX and I would like to send a extension like its coming
from a pri, basically its sending a URI to a registered peer, I cant
send a URI, because the ip address is not static, in asterisk i would
do dial(sip/pbxpeern
wrote:
> Hi Schneur,
>
> Yes, there is a very easy way. Check out the lookup function is the
> Registrar module:
>
> http://www.opensips.org/html/docs/modules/1.8.x/registrar.html#id292636
>
> Regards,
> Ali Pey
>
> On Thu, Aug 9, 2012 at 12:45 PM, Schneur Rosenber
Just to let you know I got it to work with the following few lines of code
if($hdr(PBX-exten)!=NULL)
{
$rU=$hdr(PBX-exten);
uac_replace_to("$hdr(PBX-exten)","");
}
On Thu, Aug 9, 2012 at 9:03 PM, Schneur Rosenberg
There is a pretty good book out there to learn sip, it won't teach you
asterisk or OpenSIPS just sip, once u really understand sip OpenSIPS will
be a breeze, knowing c type language basics will help too, the name of the
book is "SIP Understanding the Session Initiation Protocol" third edition
by Al
I'm having a interesting scenario, every once in a while my OpenSIPS
location table begins losing its contents, I lose like 100 at a time,
then it might regain some and then loses again another 100 or so.
I have 2 servers one active and one standby via Heartbeat, heartbeat
shuts and turns on opens
It happened again just out of the nowhere, please can someone please
suggest what can cause it, I'm going crazy already.
On Sun, Aug 19, 2012 at 12:48 PM, Schneur Rosenberg
wrote:
> I'm having a interesting scenario, every once in a while my OpenSIPS
> location table begins los
e time on both servers and make sure the date
> and time are in sync.
>
> Regards,
> Ali Pey
>
>
> On Mon, Aug 20, 2012 at 2:00 PM, Schneur Rosenberg
> wrote:
>>
>> It happened again just out of the nowhere, please can someone please
>> suggest what can cause
I'm not sure what info u need from OpenSIPS, do u need to get info from it
or change behaviors but OpenSIPS is database driven so if you can get your
app to read/write from the database you might be able to accomplish what u
need
On Aug 23, 2012 5:32 AM, "Carlos Cruz" wrote:
> Thanks for your inp
Shalom Shimon u need to make sure that you compiled all needed modules.
On Sep 2, 2012 1:03 PM, "Shimon Mishal" wrote:
> Hi
>
> I followed your instruction and loaded the missing "sipmsgops" module but
> now I get different errors:
>
> Could someone send me the correct config file for 1.8.1
>
> E
Shimon do u have a script that worked and you're trying to make it work
with 1.8? Or are u trying to use opensips for the first time? OpenSIPS will
not work out of the box u will need to write the script according to your
specifications, and its not magic where u install it and it will work.
On Se
I have this interesting scenario, caller sends call to our OpenSIPS
who actsd as a loadbalancer which sends the call to a gateway for
termination and the gateway sets Session-Expires: 1800;refresher=uas,
the caller sends call the call through port 1090 and the rport in the
Via shows rport=1090 and
You can use avp_db_query
http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail";
target="_blank">https://ipmcdn.avast.com/images/icons/icon-envelope-tick-green-avg-v1.png";
alt="" width="46" height="29" style="width: 46px; h
ttp://www.opensips.org/events/Summit-2018Amsterdam
>
>
> On 02/12/2018 05:05 PM, Schneur Rosenberg wrote:
>>
>> I have this interesting scenario, caller sends call to our OpenSIPS
>> who actsd as a loadbalancer which sends the call to a gateway for
>> termination and t
he fix on the 200 OK and to get a routable URI there.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> OpenSIPS Summit 2018
> http://www.opensips.org/events/Summit-2018Amsterdam
>
> On 02/13/2018 12:16 PM,
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