branch, without 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 ins
, 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 wrote:
> Thanks Razvan,
>
> It is working great!! you guys are awesome!
>
>
> On Mon, Mar 9, 2015 at 9:43 AM, Satish Patel wrote:
>
&g
Thanks Razvan,
It is working great!! you guys are awesome!
On Mon, Mar 9, 2015 at 9:43 AM, Satish Patel wrote:
> Sorry It was "branch"
>
> My iPhone is over smart :(
>
> --
> Sent from my iPhone
>
> On Mar 9, 2015, at 9:12 AM, Satish Patel wrote:
>
> S
Sorry It was "branch"
My iPhone is over smart :(
--
Sent from my iPhone
> On Mar 9, 2015, at 9:12 AM, Satish Patel 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 th
Best regards,
> 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 i
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.
--
Sent from my iPhone
> On Mar 8, 2015, at 7:02 PM, Terrance Devor wrote:
>
> Good news,
>
> What is rtpengine support. Will the proxy manage
se 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 Pa
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 wrote:
> sorry for push but it wired error!
>
> I have configure siptrace to send packet to &q
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
lo
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
I have two Freeswitch in dispatcher, everything works great but i have
notice in sip trace if FS1 receive 404 SIP code then it sending it to next
FS2, i think it should stop there instead of forwarding next FS2
Following is my config
Dispatcher
loadmodule "dispatcher.so"
modparam("dispatcher
I have setup 2.1.x opensips and configure Homer on other box which is
running on Kamailio
somehow my Opensips siptrace not sending packet to sipcapture server, both
are on same LAN. what i am doing wrong?
I ran tcpdump on capture server but get nothing.
### Capture Server
modparam(
, Satish Patel wrote:
> ignore last email, it was my variable issue. I got it work now :)
>
> Thanks you very much!
>
> On Thu, Mar 5, 2015 at 12:15 PM, Satish Patel
> wrote:
>
>> I am not seeing my custom variable in MI output also i got this error in
>> logs
&g
ignore last email, it was my variable issue. I got it work now :)
Thanks you very much!
On Thu, Mar 5, 2015 at 12:15 PM, Satish Patel wrote:
> I am not seeing my custom variable in MI output also i got this error in
> logs
>
> ERROR:core:do_assign: setting PV failed
>
> On T
I am not seeing my custom variable in MI output also i got this error in
logs
ERROR:core:do_assign: setting PV failed
On Thu, Mar 5, 2015 at 12:04 PM, Liviu Chircu wrote:
> Yes. You should see that value in the dlg_list_ctx MI command.
>
>
> On 05.03.2015 18:56, Satish Patel wr
val will fail
> (check for "ERROR:core:do_assign: setting PV failed")
>
> The code you posted is for sequential request handling. Normally, the
> dialog should have been created by the time this block is reached.
>
>
> On 05.03.2015 18:39, Satish Patel wrote:
>
&g
tomer_name) = $var(name);
>
> [1] : http://www.opensips.org/html/docs/modules/2.1.x/dialog.html#id297182
>
> Best regards,
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 05.03.2015 18:27, Satish Patel wrote:
>
> Hello,
>
> How do
Hello,
How do i add customer info in opensipsctl fifo dlg_list_ctx output?
I want to add custom field ( customer name) in dlg_list_ctx output
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ls in
> OpenSIPS. Take a look at the pstn.cfg file included in the examples
> directory of the source. You'll see the proxy_authorize() function around
> line 96. That, with some module and database configuration, will get on
> the right path.
>
>
> - Jeff
>
>
> On Thu,
I am using opensips 1.11 but i have seen wired issue, How i can check auth
before process INVITE packet?
I have following code, I have seen if i send only INVITE packet using SIPP
it is processing that call, I want it check AUTH before processing INVITE
packet how can we do that?
# To FreeSWITCH
Guys! please advice me!
On Fri, Feb 13, 2015 at 10:07 AM, Satish Patel wrote:
> I have question about how to stop INVITE coming from unknown source or not
> subscribed user.
>
> I have opensips front end proxy and Freeswitch PSTN
>
> But recently i have seeing some calls c
I have question about how to stop INVITE coming from unknown source or not
subscribed user.
I have opensips front end proxy and Freeswitch PSTN
But recently i have seeing some calls coming from unknown source and method
is INVITE so it is sending direct INVITE to opensips and opensips
forwarding
a friendly IP.
>
>
> On Wednesday, December 31, 2014, Satish Patel
> wrote:
>
>> How it will help if i want to allow only IP auth for specific user but
>> not registration auth? How your logic deal with User level?
>>
>>
>> On Wed, Dec 31, 2014
_method("REGISTER"))
> {
> if (t_newtran()) {
> save("location");
> }
>
> exit;
> }
>
> On Wed, Dec 31, 2014 at 10:22 AM, Satish Patel
> wrote:
>
>> Hi,
>>
>> We have many users us
Hi,
We have many users using both registration method and IP auth method to
send calls but i wants if they use IP Auth method then we can disable
registration method ( just prevention from hacking attack).
I believe registration is only required for incoming calls to find user
location, right? Ho
Just curious when opensips 2.2.x stable version will release?
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> UserB ---> ds_select_dst("$var(group)", "4")
>
> where set $var(group)=$rU
>
> then
>
> $var(group)=$var(group){s.substr,0,1})
>
> I think this work
>
> Regards
>
> El 19/11/2014 12:10, Satish Patel escribió:
>>
>>
xtract first value from $rU pseudo variable (group 1 is 1, group 2
> is 2) and use this group on the ds_select_dst(set, alg [, max_results])
> function.
>
> Regards
>
>
> El 19/11/2014 11:31, Satish Patel escribió:
>
> Hi,
>
> We have running opensips 1.12 with disp
Hi,
We have running opensips 1.12 with dispatcher, we want to route call to
dispatcher bases on specific users. Is there any way we can implement that
scenario?
Ex:
UserA ---> dispatcherA
UserB ---> dispatcherB
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I am using opensips 1.12.x version from git, I have multiple gateways and
everything works great with uac_auth()
but recently when i add new SIP gateway which is asterisk based. I am
getting 403 Auth error (even my password is correct)
Opensips >INVITE>Asterisk
Opensips <-- 401 Unauthor
Guys please help me
Look like SIP gateway is asterisk base and it is sending 401 instead of 407.
How UAC handle 401? Do I need to change any hdr?
--
Sent from my iPhone
> On Nov 13, 2014, at 11:56 AM, Satish Patel wrote:
>
> We have opensip configured with UAC auth to register SI
We have opensip configured with UAC auth to register SIP provide. I have
configured two provider and both got registered but very interesting thing
happened.
Provider "A" sending me 407 challenge for authentication - Working
Provider "B" sending me 401 for authentication - its failed to auth
wha
Hi,
I just put new SIP provider info in registrant table for UAC auth and it is
showing
state:: REGISTERED_STATE
but my calls failing and in ngrep i check somehow remote SIP providing not
sending me 407 auth challenge packet, is it possible?
1. INVITE SIP Provider
2. SIP Provider -- 100 T
".
>
> Try with that flags and let me know.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 20.09.2014 15:18, Satish Patel wrote:
>
> Hey any clue? I'm using 1.12 version.
>
> --
> Sent f
Great! I figured out, it was group module.
--
Sent from my iPhone
> On Oct 14, 2014, at 1:17 PM, Adrian Georgescu wrote:
>
>
>> On 14 Oct 2014, at 13:02, Satish Patel wrote:
>>
>> I am reading this document to implement Quota
>> http://cdrtool.ag-project
; statistically speaking for postpaid customers. The documentation explains
> the modus operandi in more detail.
>
> Adrian
>
> On 12 Oct 2014, at 11:14, Satish Patel wrote:
>
> Thanks!! I think you got my point, we have very high density call ratio
> that is why prep
would be call center or high density call customer, how i can use
quota in that scenario?
On Sun, Oct 12, 2014 at 9:38 AM, Adrian Georgescu
wrote:
>
> On 12 Oct 2014, at 09:48, Satish Patel wrote:
>
> I have run sipp test and it only able to handle 30 calls and later all
&
ested don't use prepaid because of limitation and performance,
and suggested use Postpaid or Quota system.. is that true?
On Thu, Oct 9, 2014 at 3:46 PM, wrote:
> Yes, it is capable.
>
> On 08 Oct 2014, at 15:42, Satish Patel wrote:
>
> > Hi,
> >
> > Just want
Hi,
Just want to know does CDRTool prepaid capable of handling couple hundreds
of concurrent calls? I heard it can handle only 2/3 concurrent calls per
account? what is the solution if we want to host big prepaid system with
thousands of users?
___
User
Bogdan, any update?
--
Sent from my iPhone
> On Sep 22, 2014, at 3:43 PM, Satish Patel wrote:
>
> Where are you? We need you :) lol
>
> --
> Sent from my iPhone
>
>> On Sep 20, 2014, at 8:18 AM, Satish Patel wrote:
>>
>> Hey any clue? I'm us
Where is the trunk git URL to download latest 1.12.x? does it ready for
production?
On Thu, Sep 25, 2014 at 2:39 PM, Ovidiu Sas wrote:
> Are we ready to deprecate the mi_xmlrpc module now (for 1.12)?
>
> -ovidiu
>
> On Fri, Mar 21, 2014 at 11:24 AM, Bogdan-Andrei Iancu
> wrote:
> > Hello all,
Where are you? We need you :) lol
--
Sent from my iPhone
> On Sep 20, 2014, at 8:18 AM, Satish Patel wrote:
>
> Hey any clue? I'm using 1.12 version.
>
> --
> Sent from my iPhone
>
>> On Sep 19, 2014, at 4:29 AM, Bogdan-Andrei Iancu wrote:
>>
We are looking for IP auth but with accounting so I are planing to use tech
prefix so customer will send call with some prefix and we will use it identify
customer and bill that according
I'm planing to use permission module and its DB table has "pattern" column I
don't know what that pattern
ogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>> On 17.09.2014 19:04, Satish Patel wrote:
>> I just trying to print $avp(271) $avp(272) and $avp(273)
>>
>> I am getting following output, why dst_avp is null ? and cnt_avp s
> http://www.opensips-solutions.com
> On 17.09.2014 19:04, Satish Patel wrote:
>> I just trying to print $avp(271) $avp(272) and $avp(273)
>>
>> I am getting following output, why dst_avp is null ? and cnt_avp should be 2
>> right?
>>
>> dst_avp
ts protocols, use
> "channel-update pear.php.net" to update
> Did not download optional dependencies: pear/DB, pear/Mail, use --alldeps
> to download automatically
> Skipping package "pear/Log", already installed as version 1.12.8
> No valid packages found
> instal
.1.6-44.el5_10
> php-pear-1.4.9-8.el5
>
> Thanks & Regards
> *Sandeep Sharma*
> *IMImobile *Plot 770, Rd. 44 Jubilee Hills, Hyderabad - 500033.
> *T *+91 9912244250 - Ext: 251
> *www.imimobile.com <http://www.imimobile.com>*
>
> *From:* Satish Patel
>
Make sure you have install all components also don't forget to install
php-mysql driver.
Check apache logs definitely you will see something there.
Sent from my iPhone
On Sep 18, 2014, at 6:08 AM, "Sandeep Sharma" wrote:
> Hi Liviu,
>
> Small progress in installing and configuring control
You have to install manually also you have to install MySQL and apache.
Sent from my iPhone
On Sep 18, 2014, at 3:36 AM, "Sandeep Sharma" wrote:
> Hi,
>
> Coming to opensips control panel installation & configuration do I need to
> install below package manually or else does it come by defa
I am doing following operation in opensips script and i want this
information in memcached because every single call hitting MySQL for this
information
avp_db_query("SELECT username FROM registrant WHERE
(registrar='$var(x)')","$avp(user)");
avp_db_query("SELECT password FROM regis
o the first destination.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 17.09.2014 14:09, Satish Patel wrote:
>
>> Confirmed probing/inactive thing is working,
>>
>> Now problem is failov
elect_dst(), you can see how many other
> gw are prepared to used in case of failover.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 16.09.2014 21:02, Satish Patel wrote:
>> After doing couple of
After doing couple of TEST look like its marking "Probing" for failed
gateway but not auto failover to next gateway, i meant call get disconnect
and i need to re-initiate call then all call goes to second active
gateway..
I believe it should first mark gateway "Probing" and then fall-back to
secon
http://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc47
>
> Also try an avp_print() after ds_select_dst() to see what data is kept
> into transaction.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
&g
ion via "ds_next_dst". Firs mark the used one as
> probing and then use the next one.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 16.09.2014 07:59, Satish Patel wrote:
>
> following is my config,
following is my config, I have two Freeswitch, if i stop one of freeswitch
and call it won't failover itself. but if again i call if send call to
other FS and if again i call it send to failed one but not auto failover..
But after my prob detect it is dead then it change status from Active to
Pro
I want to disable "db_check_from" function but want to make sure Opensips
is secure enough.
Reference email:
http://lists.opensips.org/pipermail/users/2012-June/022057.html
Bogdan-Andrei saying "If you disable the function, any SIP user will
be able to use any valid
auth credentials."
I have dis
On Mon, Sep 15, 2014 at 1:22 PM, Satish Patel wrote:
>> Holy crap!! it just got registered, look like i didn't wait enough :( sory
>> my bad... it seems it works!!
>>
>> AOR:: sip:73757...@sip.example.com:5060 expires=300
>>state:: REGISTERED_STATE
currently we have following config, and IP is hard coded, we have many
src_ip so how do i put them in MySQL database? i know i can use avp_db but
how they will stored in memory and opensip read them from memory instead of
disk everytime.
also how do i reload them without restarting opensips servic
2014
registrar:: sip:sip.example.com
binding:: sip:73757...@sip.example.com
dst_IP:: IPv4 ip=xxx.xxx.xxx.xxx
On Mon, Sep 15, 2014 at 1:02 PM, Satish Patel wrote:
> I have totally removed "binding_params" from table but still seeing same
> error, wh
p them, then you need to prefix them with';'
>
> Regards,
> Ovidiu Sas
>
> On Mon, Sep 15, 2014 at 10:19 AM, Satish Patel
> wrote:
> > Here is my registrant dump output
> >
> > AOR:: sip:73757...@sip.example.com:5060 expires=300
> >
t header that you built, it doesn't seem right:
> transport=UDP;expires=300
> You need a ';' after '>'. It should look like this:
> ;transport=UDP;expires=300
>
> Fix your config and try again.
>
> Regards,
> Ovidiu Sas
>
> On Mon, Sep 15,
Any thought? it works with other SIP clients but not Opensips UAC :(
On Fri, Sep 12, 2014 at 9:16 PM, Satish Patel wrote:
> But if i configure same account on my SIP phone it works! why it is
> misbehaving with Opensips?
>
> SIP provide will argue if it works with SIP phone then it
:" + $rd + ":" + $rp;
>
> [1]: http://www.opensips.org/Documentation/Script-CoreVar-1-12
>
> Best regards,
> Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
> On 12.09.2014 23:48, Satish Patel wrote:
>> Following is $ru a
; Regards,
> Ovidiu Sas
> On Sep 12, 2014 4:49 PM, "Satish Patel" wrote:
>
>> What is the solution? do i need to tell my SIP provide or i should do
>> something at Opensips side?
>>
>> On Fri, Sep 12, 2014 at 4:47 PM, Ovidiu Sas
>> wrote:
>>
&
; < :5060>transport=UDP>;q=1;expires=300;received="sip:182.xx.xx.xx.xx:5060"
>
> See the double '<'.
>
> Regards,
> Ovidiu Sas
> On Sep 12, 2014 4:37 PM, "Satish Patel" wrote:
>
>> Here is the trace, so where is the problem?
>&
Following is $ru and i want to extract following sip:sipprovider.com:5060
sip:123456...@sipprovider.com:5060
to
sip:sipprovider.com:5060
I am using following regex but its not working, does following make sense?
$var(z) = $ru;
$var(z) = "s/[^:@]*@//";
xlog("My regex $ru\n");
eld with
> zero or more values containing address bindings.
>
> The error that you have is related to the Contact header in the reply.
> Check the reply received from the registrar.
>
> Regards,
> Ovidiu Sas
>
>
> On Fri, Sep 12, 2014 at 3:03 PM, Satish Pate
In logs i am getting this error
ERROR:uac_registrant:run_reg_tm_cback: failed to parse Contact body
AOR:: sip:9...@sip.example.com:5060 expires=300
state:: INTERNAL_ERROR_STATE
last_register_sent:: Sat Sep 13 00:30:28 2014
registration_t_out:: Sat Sep 13 00:35:28 2014
ttrs field (in dr_gateways) - when
> that GW is used, attrs will become available so you can use them.
>
> Regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 11.09.2014 18:28, Satish Patel wrote:
>> Currently i have
Currently i have UAC_AUTH working with single gateway and configuration
look like following, How do i configure multiple gateway trunk account?
what would be the best way to make it happen, also i am worried about
"uac_replace_from" address, in multiple gateway account won't be same so i
how i map
Thanks for replay, anyone did that before? Any example or sample script will
help
Sent from my iPhone
On Sep 10, 2014, at 12:17 PM, Juha Heinanen wrote:
> Satish Patel writes:
>
>> I heard somewhere LCR can do routing based on call rate and call
>> price, does it true?
I heard somewhere LCR can do routing based on call rate and call price, does it
true? I haven't seen any config or doc which does call rate using LCR. It only
does routing base on prefix scan.
Am I missing something here?
Sent from my iPhone
___
User
I have a question, I want to modify R-URI host portion for example
Sip: 1...@abc.com
Change to
Sip: 1...@foo.com
But after doing that it break my routing logic, so just want to know did it
possible to chnage host portion of R-URI?
Sent from my iPhone
I have setup DR on opensips and just added only single gateway to test my
routing but i am getting following error.
/opt/opensips/sbin/opensipsctl fifo dr_gw_status
ID:: 1 IP=65.xxx.xxx.xxx:5065 State=Inactive
INFO:drouting:do_routing: All the gateways are disabled
do_routing: No rules matching
wrote:
> In this case, DR module is what you need.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 04.09.2014 19:31, Satish Patel wrote:
>
> ALL,
>
> I heard opensips not going to maintain carrierrou
ALL,
I heard opensips not going to maintain carrierroute? should i use
Drouting? I need strong reason to no go with carrierroute.. our main goal
is LCR function but we need robust application which handle thousands of
concurrent calls.
___
Users mailing
I need to configure carrierroute with my opensips but I didn't find any config
example or any kind of detail document, I saw there is a module documents but I
really need something good.
Also what is the different between Drouting and carrierroute?
Sent from my iPhone
how to enable RTPproxy for a
> call.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 02.09.2014 19:31, Satish Patel wrote:
>
> Do you have solution or documents to do that?
>
>
> On Tue, Sep 2, 2
,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 01.09.2014 21:34, Satish Patel wrote:
>
> I have following setup, UA register to Opensips and opensips send call
> to FS (freeswitch) and again freeswitch send call back to open
s mediaproxy or rtpproxy) in order to hide also the RTP side.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 02.09.2014 05:58, Satish Patel wrote:
>
> But topology hiding not hiding 100% info, I can see SDP RTP
Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 28.08.2014 23:47, Satish Patel wrote:
>
> I am looking for top hiding and i tried topoloy_hiding() but it doesn't
> handling BYE mesg so i am planing to go with B2B. I have few question
>
> 1. D
I have following setup, UA register to Opensips and opensips send call to
FS (freeswitch) and again freeswitch send call back to opensips and then
call get outside routed. in following Senior freeswitch sending 482 Loop
detect error, How do i achieve following scenario?
[UA]--[Opensips]--[
gt; Best Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 29.08.2014 08:11, Satish Patel wrote:
>
> Very interesting thing happened, If i am authentication trunk using
> uac_auth() function then it is not handling BYE from callee, but if i use
allow=all
nat=yes
On Thu, Aug 28, 2014 at 8:40 AM, Vlad Paiu wrote:
> Hello,
>
> Please privately send me again the SIP trace for the call.
>
>
> Best Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 28.08.2014 15:23, Satish P
I am looking for top hiding and i tried topoloy_hiding() but it doesn't
handling BYE mesg so i am planing to go with B2B. I have few question
1. Does B2B work like Proxy?
2. Does B2B support NAT SIP client?
Or should i install Opensips proxy and B2B opensips on same box and
interconnect them?
___
e call to record_route() from your script, or move
> topology_hiding() after the record_route() function call.
>
> Best Regards,
> Vlad Paiu
> OpenSIPS Developer
> http://www.opensips-solutions.com
> On 27.08.2014 20:25, Satish Patel wrote:
>> Hi Vlad,
>>
>> I
e's no need to call record_route() at all, so
> please remove the call to record_route() from your script, or move
> topology_hiding() after the record_route() function call.
>
> Best Regards,
> Vlad Paiu
> OpenSIPS Developer
> http://www.opensips-solutions.com
> On
they show up ? at request ? initial or sequential ?
>
> Regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 26.08.2014 23:33, Satish Patel wrote:
>> I am using 1.12 Opensips and just playing with B2B top hi
PM, Satish Patel wrote:
> I have put topology_hiding() function at following place in script but its
> not hiding VIA header following is my senerio
>
> [UA]>[Opensips]---[Asterisk/SIP gateway]
>
> I want to hind my UA IP address so Asterisk doesn't see the
ly what's going on ( leave plain topology hiding in place,
> please remove your hacks with the Route headers ).
>
> Best Regards,
>
> Vlad Paiu
> OpenSIPS Developer
> http://www.opensips-solutions.com
>
> On 27.08.2014 14:08, Satish Patel wrote:
>> I
opology_hiding()
> and when routing sequential requests, and also please pastebin a full SIP
> trace showing the traffic for such a dialog.
>
> Best Regards,
> Vlad Paiu
> OpenSIPS Developer
> http://www.opensips-solutions.com
> On 26.08.2014 15:38, Satish Patel wrote:
>&
I have post many question on topology hiding any not get any reply back from
people and developers now I don't have any option except some goofy hack
When I use topology hiding it removes Route: and because of that callee doesn't
able to send BYE back to opensips.
I want use insert_hf to inje
I'm on same boat, I really want to use topology hiding but it's not working and
missing BYE because its deleting route: in sip dialogs.
There is not any good document out there so for now we are going with
freeswitch.
Sent from my iPhone
On Aug 26, 2014, at 4:59 AM, Eugene Prokopiev wrote:
uests processing as usual
}
}
On Tue, Aug 26, 2014 at 8:38 AM, Satish Patel wrote:
> I have tried your logic and it works but it is not handling BYE message,
> after caller hang up phone, caller not receiving BYE and caller phone is
> still in connected state not getting hung up.
>
I am using 1.12 Opensips and just playing with B2B top hiding and i am
getting following error
ERROR:b2b_logic:create_top_hiding_entities: failed to create new b2b server
instance
ERROR:b2b_logic:create_top_hiding_entities: failed to create new b2b server
instance
b2b_reply (B2B.346.3114641)
ERROR
per
> http://www.opensips-solutions.com
> On 26.08.2014 06:48, Satish Patel wrote:
>> I have put topology_hiding() function at following place in script but its
>> not hiding VIA header following is my senerio
>>
>> [UA]>[Opensips]---[Asterisk/SI
I have put topology_hiding() function at following place in script but its
not hiding VIA header following is my senerio
[UA]>[Opensips]---[Asterisk/SIP gateway]
I want to hind my UA IP address so Asterisk doesn't see them, currently my
asterisk can see what IP address UA coming f
Just followup email, I have use avp variable to set username/password/realm
and it works!
On Sun, Aug 24, 2014 at 4:24 PM, Satish Patel wrote:
>
> Hi,
>
> my Opensips (UAC) registered to PSTN gateway and now i am trying to call
> using my SIPphone which is register to opensip bu
isk:mypassw0rd#$")
On Mon, Aug 25, 2014 at 10:59 AM, Satish Patel wrote:
> Perfect!!! just resync code from repo and look like it compile
> successfully!!
>
> I am going to give it a shot and update you soon!
>
>
> On Mon, Aug 25, 2014 at 10:36 AM, Vlad Paiu wrote:
>
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