gt; that.
>
> Try OpenSIPS 2.2 or 2.3
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
> http://www.opensips.org/events/Summit-2017Amsterdam.html
>
>> On
sterdam
> http://www.opensips.org/events/Summit-2017Amsterdam.html
>
>> On 04/27/2017 03:43 AM, Satish Patel wrote:
>> Yes, whenever fix_nated_sdp() fiction run it produce that error which I
>> mentioned in my previous email. Every single time.
>>
>> Sent f
/Summit-2017Amsterdam.html
>
>> On 04/26/2017 08:44 PM, Satish Patel wrote:
>> Here is my payload again we have custom application which is using SER
>> so some of them are custom values, This is the payload after i apply
>> fix_nated_sdp() function.
>>
>>
>
ps-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
> http://www.opensips.org/events/Summit-2017Amsterdam.html
>
>
> On 04/25/2017 10:35 PM, Satish Patel wrote:
>>
>> We have some custome Voice solution and in-house media server so right
>> now i
sense would there be in substituting the source of the SIP
> message in there?
>
> On Mon, Apr 24, 2017 at 11:23:30PM -0400, Satish Patel wrote:
>
>> I meant "origin public address of client" if c line isn't public then
>> media never work.
>>
>> c=IN IP4 192.168.1.
ess", do you mean the external source
> address and port of the SIP message, or the incoming RTP stream?
>
> On Mon, Apr 24, 2017 at 11:00:40PM -0400, Satish Patel wrote:
>
>> In my INVITE/SDP i am seeing sometime rfc1918 address which i want fix
>> and replace it with
In my INVITE/SDP i am seeing sometime rfc1918 address which i want fix
and replace it with origin public address. ex
I am seeing following info in INVITE
v=0.
o=amsip 0 0 IN IP4 192.168.1.8.
s= .
c=IN IP4 192.168.1.8.
t=0 0.
m=audio 22530 RTP/AVP 127 111 0 101.
g the Public internet, he's the one
> that should take care of any NAT handling.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
>
>
> On 04/16/2017 03:48 AM, Satish Patel wrote:
>>
>> Any help here?
>>
&g
Any help here?
On Sat, Apr 15, 2017 at 10:04 AM, Satish Patel <satish@gmail.com> wrote:
> currently we have following design 1 dispatcher and 3 SIP Proxy and
> every server on Public IP. Dispatcher using as a load-balancer to
> distribute load on proxy.
>
> My disp
currently we have following design 1 dispatcher and 3 SIP Proxy and
every server on Public IP. Dispatcher using as a load-balancer to
distribute load on proxy.
My dispatcher is very simple and stateless (I am not using record_route() too)
Everything working fine, but now we decided to move alls
il.com> wrote:
> This might help you.
> http://www.opensips.org/html/docs/modules/2.2.x/rr.html#id293864
>
> On 31 March 2017 at 06:31, Satish Patel <satish@gmail.com> wrote:
>>
>> is there a way i can re-write record-route port number?
>>
>> On Wed, Mar 29, 20
is presumed.
>
> On March 29, 2017 6:27:30 PM EDT, Satish Patel <satish@gmail.com> wrote:
>>what is the use of port number in record-route?
>>
>>I am having major issue with that look like we are running sip server
>>on different port to protect ourself from sip scanner
what is the use of port number in record-route?
I am having major issue with that look like we are running sip server
on different port to protect ourself from sip scanner we are using
non-standard port like 6060/7070 multiple port on single server so it
will failover to other port if firewall
n Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
>
>
> On 03/26/2017 08:16 PM, Satish Patel wrote:
>>
>> any suggestion?
>>
>> On Tue, Mar 21, 2017 at 6:48 PM, Satish Patel <satish@gmail.com>
>> wrote:
>>>
>>> This is littl
any suggestion?
On Tue, Mar 21, 2017 at 6:48 PM, Satish Patel <satish@gmail.com> wrote:
> This is little tricky question, we are developing softphone and we put
> logic in phone it will try to connect 5060 if it's blocked by some
> country then it will try 5061 if that is bloc
This is little tricky question, we are developing softphone and we put
logic in phone it will try to connect 5060 if it's blocked by some
country then it will try 5061 if that is block then try 5062
Now on OpenSIPS we are listening on all 3 ports 5060, 5061 and 5062.
Now problem is here INVITE
g/About/PerformanceTests
> to see results for different types of configurations. However, do keep in
> mind that those results may be done on older versions of OpenSIPS and you
> may want to stress test your setup separately to know what are your
> capabilities.
>
> Regards,
> Sa
t; relative to the business logic itself. For example doing alot of DB queries,
> engaging various other modules etc these things really define how light or
> heavy your system is going to be.
>
> Regards,
> Sammy
>
>
>> On Sun, Mar 6, 2016 at 10:36 A
Any thought on it???
On Fri, Mar 4, 2016 at 1:30 PM, Satish Patel <satish@gmail.com> wrote:
> We have dispatcher and we are using very simple code block like following
>
> if (method=="REGISTER" || method=="INVITE" ) {
> ds_select_dst("1",
We have dispatcher and we are using very simple code block like following
if (method=="REGISTER" || method=="INVITE" ) {
ds_select_dst("1", "2");
t_relay();
}
Does t_relay will keep all transaction in memory? and what will be the
performance issue? we have ~200k cps calls.. what
ng system decides where the reply should be
> sent to. And in your case, the operating system simply chooses a different
> interface. So it seems this is the normal behavior, there's nothing wrong.
> If you really want to use the same interface for replies, you should use the
> force_se
mhome=1
listen=udp:10.0.0.1:6060 udp:10.0.0.1:5060 udp:192.168.100.1:6060
udp:192.168.100.1:5060
>From client when i send REGISTER to 5060 then server sending reply
back using port 6060, it should send reply back client using 5060
right???
If i use mhome=0 everything works!
I have following scenario
[client]-Public-IP--[dispatcher]--LAN-IP--[proxy]
Dispatcher has multi home interface, public and private, when client send
request to dispatcher public Interface then dispatcher should use private LAN
IP to send that request to Proxy can dispatcher do that?
I am following this document:
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1
My sipML5 client successfully register but somehow its not calling each
other. I check AOS and it looks strange, where is the received: header to
contact client?
AOR:: 1...@sip.example.com
);
exit;
}
On Wed, Jun 24, 2015 at 9:24 AM, Eric Tamme e...@uphreak.com wrote:
just t_relay the request to your other server... OpenSIPS wont
automatically challenge anything
On 06/24/2015 07:22 AM, Satish Patel wrote:
All,
I have special requirement which is little odd, I want
All,
I have special requirement which is little odd, I want to use WebRTC with
Opensips but all REGISTER process will done by other SIP server,
Example:
[UA][WebRTC-Opensips]---[Asterisk/Freeswitch]
UA will use WebRTC of Opensips but opensips forward all
if that didn’t pose too great a performance issue? I haven’t
tried it, just thinking outloud.
On Jun 2, 2015, at 10:45 AM, Satish Patel satish@gmail.com wrote:
We have opensips 2.1, is there anyway we can enable disable siptrace
without restart opensips? because it will be very helpful
We have opensips 2.1, is there anyway we can enable disable siptrace
without restart opensips? because it will be very helpful.. we don't want
to keep it on for no reason..
route {
...
...
sip_trace();
}
___
Users mailing
nevermind, i found answer.
opensipsctl fifo sip_trace off
opensipsctl fifo sip_trace on
On Tue, Jun 2, 2015 at 11:58 AM, Satish Patel satish@gmail.com wrote:
But siptrace() has no connection with cache store (memory, etc)... it is
just raw sip messages store in database or send to Homer
we have old opensips 1.7.2 but running with default options, we want to
increase shared memory so do we need to re-compile it?
As per this document
http://www.opensips.org/Documentation/TroubleShooting-IncreaseMem
Re-compile only for Private memory or for it apply to shared memory too?
Sorry guys!!!
On Sat, May 9, 2015 at 9:05 PM, Satish Patel satish@gmail.com wrote:
Hi,
We have integrate Openstack with vCenter VMware and when i create instance
on Openstack GUI it created successfully on VMware but it throwing error on
Openstack. Look like openstack trying 3 time
Hi,
We have integrate Openstack with vCenter VMware and when i create instance
on Openstack GUI it created successfully on VMware but it throwing error on
Openstack. Look like openstack trying 3 time to create VM even VM is
already running on VMware. I don't know what is wrong here, why openstack
Hi,
I have installed 2.1 but i didn't understand use of Async mode. I was
reading article http://www.opensips.org/Documentation/Script-Async-2-1
But don't understand how and where i can use in my script because in my
script i am doing some SQL operation but i don't know how it can fit and
how
Developerhttp://www.opensips-solutions.com
On 30.04.2015 23:34, Satish Patel wrote:
mysql select * from userblacklist;
++--+++---+
| id | username | domain | prefix | whitelist |
++--+++---+
| 1 | 1001
module to check against valid ANIs and reply
with a 403.
On Thu, Apr 30, 2015 at 12:09 PM, Satish Patel satish@gmail.com
wrote:
Question: what happen if client send call using random string in
RPID/PAI/From header (non-standard ANIs) How to verify ANIs for those
headers? is there a blacklist
with
a 403.
On Thu, Apr 30, 2015 at 12:09 PM, Satish Patel satish@gmail.com
wrote:
Question: what happen if client send call using random string in
RPID/PAI/From header (non-standard ANIs) How to verify ANIs for those
headers? is there a blacklist or any method which scan ANIs
mysql select * from userblacklist;
++--+++---+
| id | username | domain | prefix | whitelist |
++--+++---+
| 1 | 1001 ||| 0 |
| 2 | 9198362323 ||| 0 |
i tried to using is_uri_user_e164 () but it will allow anything with '+'
sign, like if i set +1001 its allowing...
I want to match E.164 or ANI string match logic so it should be something
like 16463272823 44635364894 like those number..
i don't want send random string to my SIP provide so
nevermind, i found issue it was double quote string so use
*$(fn{s.select,1,\})
to get rid on it*
On Thu, Apr 30, 2015 at 4:34 PM, Satish Patel satish@gmail.com wrote:
mysql select * from userblacklist;
++--+++---+
| id | username | domain
Question: what happen if client send call using random string in
RPID/PAI/From header (non-standard ANIs) How to verify ANIs for those
headers? is there a blacklist or any method which scan ANIs for source and
block them?
___
Users mailing list
I want to overwrite FROM: header using RPID in opensips. but i don't know
where to put code and how to use that rpid feature in subscriber table
Does any one has sample code which i can try on my config because i don't
know how to call rpid and modify FROM address in standard ANI format
We are using IP base authentication so how do i load rpid value from
subscriber table? or do i need to do SQL query function to load?
is there any built in function in opensips to manually load rpid from
subscriber table if we don't want to use Registration method?
wrote:
status can be either Enabled, Disabled, Probing so what you see in the web
interface is I guess that it is not disabled. Correct me if I'm wrong.
On Tue, Apr 21, 2015 at 6:44 PM, Satish Patel satish@gmail.com
wrote:
In command line it is showing
root@dispatcher1:/var/www
wrote:
MySQL has a option --password to provide the password on command line.
postgres however does not. but here is some refs how you can accomplish
what you would like.
http://www.postgresql.org/docs/9.4/static/libpq-pgpass.html
On Tue, Apr 21, 2015 at 4:26 PM, Satish Patel satish
I just did for 1.11 branch, if you want i can do same for 2.1 latest branch
On Wed, Apr 22, 2015 at 2:32 PM, Satish Patel satish@gmail.com wrote:
Hey Liviu,
I have fixed script here is the pull request
https://github.com/OpenSIPS/opensips/pull/476
On Wed, Apr 22, 2015 at 8:27 AM, Liviu
I am installing opensips 1.11.3 with postgress DB but i don't know what is
going on here
root@dopensips:/etc/opensips# opensipsdbctl create
INFO: creating database opensips ...
Password for user postgres:
Password for user postgres:
Password for user postgres:
NOTICE: CREATE TABLE / UNIQUE will
Does dispatcher module use internal cache? Because we have many group and i
don't want opensips hit everytime SQL table to read IP address.
Does opensips load dispatcher table in memory to not hit SQL for each call?
___
Users mailing list
.
On Tue, Apr 21, 2015 at 1:44 PM, Satish Patel satish@gmail.com
wrote:
I am installing opensips 1.11.3 with postgress DB but i don't know what
is going on here
root@dopensips:/etc/opensips# opensipsdbctl create
INFO: creating database opensips ...
Password for user postgres:
Password
In command line it is showing
root@dispatcher1:/var/www# opensipsctl dispatcher dump
SET:: 1
URI:: sip:173.XXX.XXX.181 state=Probing
But on Opensips-cp web interface it is showing
DB state: Active
Look like it is not syncing with opensips properly.
what would be the issue?
I got confused in your diagram so i just wanted to clean, anyway i think
your diagram should be like following
[UA]-[Opensips]--[SIP Provider]
| |
| |
| |
:: value
that's displayed in the ul show output, setting it as $du, while the actual
Request-URI of the message will contain the private Contact that the client
registered.
Best Regards,
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 14.04.2015 18:09, Satish Patel wrote
Hi,
I have following User registred over public IP but that client doesn't
support STUN so contact info showing private IP 192.168.1.6
lookup function default extract Contact:: sip:1001@192.168.1.6:27098
Is there a way i can extract Received:: sip:173.XX.XX.215:27098 so i
can create new URI
seed = 2219233510
rfd = 4
__FUNCTION__ = main
(gdb)
On Mon, Apr 13, 2015 at 11:30 AM, Satish Patel satish@gmail.com wrote:
Thanks Jeff for reply,
Opensips crash with error child process 6645 exited by a signal 11
I have generated core dump and following is my gdb
particular environment.
Only after all these things are done will opensipsctl produce any output.
- Jeff
On Sun, Apr 12, 2015 at 3:53 PM, Satish Patel satish@gmail.com
wrote:
Bump!! Please help
--
Sent from my iPhone
On Apr 10, 2015, at 7:56 AM, Satish Patel satish@gmail.com
Bump!! Please help
--
Sent from my iPhone
On Apr 10, 2015, at 7:56 AM, Satish Patel satish@gmail.com wrote:
Any thought? Why that command killing my opensips? I think it's a BUG.
On production it will be dangerous if it kill service with random command.
--
Sent from my iPhone
*From:* users-boun...@lists.opensips.org [mailto:
users-boun...@lists.opensips.org] *On Behalf Of *Satish Patel
*Sent:* Thursday, April 09, 2015 10:13 AM
*To:* OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] URGENT: contact header changed between
opensips and client
Any thought
Any thought guys? Why contact address change in transit? is it normal?
On Wed, Apr 8, 2015 at 12:12 PM, Satish Patel satish@gmail.com wrote:
Very interesting thing going on between opensips and my client [UA]
Opensips sending following contact header to Client but on client side i
check
...@lists.opensips.org]
*On Behalf Of *Satish Patel
*Sent:* Thursday, April 09, 2015 10:13 AM
*To:* OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] URGENT: contact header changed between
opensips and client
Any thought guys? Why contact address change in transit? is it normal
traces clearly indicate the message being sent
correctly by OpenSIPS and it is being altered in a way that could only
occur in the UA’s private network.
Ben Newlin
*From:* users-boun...@lists.opensips.org [mailto:
users-boun...@lists.opensips.org] *On Behalf Of *Satish Patel
*Sent
Very interesting thing going on between opensips and my client [UA]
Opensips sending following contact header to Client but on client side i
check it received different contact header, who is changing it?
[UA]-[Opensips]--[FS]
UA - 173.xx.xx.xx.215
Opensips - 182.xx.xx.164:5060
FS -
I have following R-URI, want to remove sip: and anything after ;
sip:12345678@176.74.234.222:31156;rinstance=19e78a48990c0005
I want following from above string
12345678@176.74.234.222:31156
___
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Users@lists.opensips.org
Never mind, i figure out
if ($avp(route_info) == 'NULL')
On Tue, Apr 7, 2015 at 11:07 AM, Satish Patel satish@gmail.com wrote:
I have following SQL query
avp_db_query(SELECT billing_customer,route_info FROM inbound WHERE
(did='$tU'),$avp(bparty);$avp(route_info));
I want to check
It is URGENT!!
can some one help? This is very strange issue and i am stuck here :(
loose_route() sending ACK/BYE itself instead of next hope :(
I have removed all entries from domain table but no luck :(
On Thu, Mar 26, 2015 at 12:09 AM, Satish Patel satish@gmail.com wrote:
Hi
://www.opensips-solutions.com
On 07.04.2015 19:32, Satish Patel wrote:
It is URGENT!!
can some one help? This is very strange issue and i am stuck here :(
loose_route() sending ACK/BYE itself instead of next hope :(
I have removed all entries from domain table but no luck :(
On Thu, Mar 26
I am using opensips 2.1 and trying to use get_dialog_info
I do have dialog module loaded. Did you remove that function in 2.1?
loadmodule dialog
___
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Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Hi,
senario:
[UA]-[Opensips]-[Freeswitch]
UA sending correct ACK to freeswitch but Opensips loose_route() sending it
to itself and it break dialog, If use fix_dialog_route() then it works, I
don't have any IP address added in domain table also.
How do i check whether
Wow! so every person will get T-shirt who reported bug or one person among
all bug reporter?
On Tue, Mar 24, 2015 at 1:06 PM, Răzvan Crainea raz...@opensips.org wrote:
Hi, All!
Hurry up, we already have three bugs reported[1] :).
[1] http://www.opensips.org/Community/BugHuntContest
Default SIP port 5060 UDP also you need media port call RTP
--
Sent from my iPhone
On Mar 19, 2015, at 7:29 AM, Mattia Adducchio m.adducc...@progel.net wrote:
Hello Everyone,
I'm trying to setup my personal sip server. In this moment it works only in
my network, but now I want to open
://www.opensips.org/Documentation/Script-CoreFunctions-2-1#toc48
Best Regards,
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 18.03.2015 22:47, Satish Patel wrote:
I know you guys are super busy in OpenSIPS 2.1 release, but any suggestion
on above issue?
On Wed, Mar 18
://www.opensips-solutions.com
On 19.03.2015 06:17, Satish Patel wrote:
I have add extra zone column in subscriber table,
+--+-+
| username | zone |
+--+-+
|1001 |1|
|1002 |2|
+--+-+
In dispatcher table
Thanks! for quick answer!!
On Thu, Mar 19, 2015 at 12:41 PM, Vlad Paiu vladp...@opensips.org wrote:
Hello,
It will do fail-over.
Best Regards,
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 19.03.2015 18:39, Satish Patel wrote:
Thanks Vlad,
Superb! so it will do
I know you guys are super busy in OpenSIPS 2.1 release, but any suggestion
on above issue?
On Wed, Mar 18, 2015 at 12:17 AM, Satish Patel satish@gmail.com wrote:
I am getting following error in log, I can understand my contact: and
Route: values mismatching here. why it is happening
I have two Freeswitch in dispatcher table:
+---+-+--+
| setid | destination | description |
+---+-+--+
| 1 | sip:fs1.example.com | Freeswitch-1 |
| 2 | sip:fs2.example.com | Freeswitch-2 |
I have add extra zone column in subscriber table,
+--+-+
| username | zone |
+--+-+
|1001 |1|
|1002 |2|
+--+-+
In dispatcher table I have following two Freeswitch in two groups.
I am getting following error in log, I can understand my contact: and
Route: values mismatching here. why it is happening? is there a way to get
rid on this error?
Following is scenario. Only getting error in BYE message.
[UA][OpenSIP]---[Freeswitch]-[Opensip]-[SIP
wrote:
I do not see the 500 from opensips in this log.
On 03/17/2015 11:07 AM, Satish Patel wrote:
Here is the debug 4 logs http://pastebin.com/CdPxFrNp
173.48.111.111 - UA
188.79.242.164 - OpenSIPs
On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme e...@uphreak.com wrote
Even after disabled siptrace it is happening. no luck :(
On Tue, Mar 17, 2015 at 1:20 PM, Eric Tamme e...@uphreak.com wrote:
Turn of your sip tracing and see if the issue occurs. Its running some
sl_callbacks (which i assume are realated to siptrace).
On 03/17/2015 11:19 AM, Satish Patel
the error.
On 03/17/2015 12:10 PM, Satish Patel wrote:
Sorry forgot to post link
http://lists.opensips.org/pipermail/users/2012-August/022705.html
also interesting thing, I am not seeing xlog in opensips.log, why?
if ( 0 ) setflag(TCP_PERSISTENT);
if (!save(location
))
xlog(L_ERR, Saving contact failed - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n);
sl_reply_error();
exit;
}
On Tue, Mar 17, 2015 at 1:27 PM, Satish Patel satish@gmail.com wrote:
Even after disabled siptrace it is happening
Here is the debug 4 logs http://pastebin.com/CdPxFrNp
173.48.111.111 - UA
188.79.242.164 - OpenSIPs
On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme e...@uphreak.com wrote:
This is a ladder diagram, not a sip trace. A ladder diagram is not useful
in this case.
Turn your debug up to 4,
RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n);
sl_reply_error();
exit;
}
On Tue, Mar 17, 2015 at 2:09 PM, Satish Patel satish@gmail.com wrote:
I have check on book example and it doesn't have any brace also. just
wonder!
Look at this link, someone
())
##{
## sl_send_reply(403,Forbidden auth ID);
## exit;
##}
if (!save(location))
sl_reply_error();
exit;
}
}
On Tue, Mar 17, 2015 at 1:48 PM, Satish Patel satish@gmail.com wrote:
Eric,
I found what was the issue, I sent you REGISTER method snippet before, if
you look at it, If remove/comment out
suprised your script runs at all
On 03/17/2015 11:48 AM, Satish Patel wrote:
Eric,
I found what was the issue, I sent you REGISTER method snippet before,
if you look at it, If remove/comment out sl_reply_error(); line in
following code, it stopped sending 500 Error. Very interesting.. Do
I have set debug level 9 but still not seeing any 500 in logs
debug=9
log_stderror=no
log_facility=LOG_LOCAL7
On Tue, Mar 17, 2015 at 1:11 PM, Eric Tamme e...@uphreak.com wrote:
I do not see the 500 from opensips in this log.
On 03/17/2015 11:07 AM, Satish Patel wrote:
Here is the debug
Following is trace, OpenSIPs sending 500 to UA ( SIP Phone), Here is the
pastebin. http://pastebin.com/UPhNVSGZ
[SIP-Phone][OpenSIP Proxy]
11:33:21.331372 │ REGISTER │
▒ │ ── │
▒
Guys,
any suggestion? it is Opensips 2.1.x Master branch. Is it a bug?
On Fri, Mar 13, 2015 at 4:03 PM, Satish Patel satish@gmail.com wrote:
Hello,
Why we are getting 500 error in REGISTER time? Even i got register
successfully, is it normal?
01:26:01.078025 │ REGISTER
(Golam Sarwar)
PGP Key Fingerprint : D3A1 EED0 5BA0 72D3 A011 75CB 8EA6 7D99 F433 E92D
PGP Key Download URL: http://bit.ly/gsbabil-pgp-key
On Fri, Mar 13, 2015 at 1:03 PM, Satish Patel satish@gmail.com
wrote:
Hello,
Why we are getting 500 error in REGISTER time? Even i got register
at 3:16 PM, Satish Patel satish@gmail.com wrote:
On download page, http://www.opensips.org/Downloads/Downloads, currently
we have following.
GIT clone of development stable version 2.1.1 (MASTER):
Does on march 18th release will be 2.1.2 ?
On Thu, Mar 12, 2015 at 11:26 AM, Bogdan-Andrei
On download page, http://www.opensips.org/Downloads/Downloads, currently we
have following.
GIT clone of development stable version 2.1.1 (MASTER):
Does on march 18th release will be 2.1.2 ?
On Thu, Mar 12, 2015 at 11:26 AM, Bogdan-Andrei Iancu bog...@opensips.org
wrote:
The D day for
ERROR:siptrace:pipport2su: bad protocol
On Tue, Mar 10, 2015 at 12:12 PM, Satish Patel satish@gmail.com wrote:
I am configuring siptrace with homer server and getting following error,
ERROR:siptrace:pipport2su: bad protocol
ERROR:siptrace:pipport2su: bad protocol
ERROR:siptrace:pipport2su: bad
I am configuring siptrace with homer server and getting following error,
ERROR:siptrace:pipport2su: bad protocol
ERROR:siptrace:pipport2su: bad protocol
ERROR:siptrace:pipport2su: bad protocol
Razvan fix following patch but still getting above error in log
,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already using it
since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM
Sorry It was branch
My iPhone is over smart :(
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On Mar 9, 2015, at 9:12 AM, Satish Patel satish@gmail.com wrote:
Superb, definitely going to give a try, I have a silly question. Can I apply
that patch manually on my beach because if I try new master then fires
Thanks Razvan,
It is working great!! you guys are awesome!
On Mon, Mar 9, 2015 at 9:43 AM, Satish Patel satish@gmail.com wrote:
Sorry It was branch
My iPhone is over smart :(
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On Mar 9, 2015, at 9:12 AM, Satish Patel satish@gmail.com wrote:
Superb
, sigio_rt, select.
git revision: b3beb20
main.c compiled on 11:44:56 Dec 31 2014 with gcc 4.4.7
On Mon, Mar 9, 2015 at 10:39 AM, Satish Patel satish@gmail.com wrote:
Thanks Razvan,
It is working great!! you guys are awesome!
On Mon, Mar 9, 2015 at 9:43 AM, Satish Patel satish
cloning the entire Master.
Best regards,
Răzvan Crainea
OpenSIPS Solutionswww.opensips-solutions.com
On 03/09/2015 05:43 PM, Satish Patel wrote:
Hey Razvan,
Can i take following patch and directly apply to my existing install
branch instead of downloading new Master?
https://github.com
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting following
error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com wrote
We have upgraded 1.12.x to 2.1.x and its been 5 month no issue so far,
everything works! i am waiting for 2.1.x stable release so i can push it
out but i would say so far its good and stable.
Just question to Liviu, How do i use latest 2.1.x feature? currently i am
using 1.x config. but i would
Bogdan,
I am running 2.1.x and so far great, I had issue with sipteace with homer which
I already reported. So please look into it before release.
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On Mar 8, 2015, at 7:02 PM, Terrance Devor ter.de...@gmail.com wrote:
Good news,
What is rtpengine support. Will
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