OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
> http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 03/23/2017 09:19 AM, John Nash wrote:
>
> I am using drouting and recently tried to use gateway a
I am using drouting and recently tried to use gateway attribute. I call ...
do_routing("$avp(int_grp_id)","WF","$avp(gw_whitelist)" ,
"$avp(rules_attributes)","$avp(gw_attributes)"))
After this call I can see $avp(gw_attributes) is populated frp, attr column
of dr_gateways table.
but when i
I can get C line using {sdp.line} but how can I separate "Connection
address" and store in some variable? csv transformation is a cool option
but that requires coma separated string.
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ain?
>
> [1] https://gist.github.com/razvancrainea/03a43bfa8b554a7ca89f2740a3c54c96
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/10/2017 10:02 AM, John Nash wrote:
>
> Dear Razvan,
>
> I think I found one issue in do_
Dear Razvan,
I think I found one issue in do_routing function if i pass gw_whitelist
then only i see this drop of memory. If I skip this parameter private
memory does not increase with every call.
Regards
Manoj
On Thu, Mar 9, 2017 at 7:29 PM, John Nash <john.nash...@gmail.com> wrote:
sips-solutions.com
>
> On 03/09/2017 03:27 PM, John Nash wrote:
>
> OK..May i send you my script privately?
>
> On Thu, Mar 9, 2017 at 6:13 PM, Răzvan Crainea <raz...@opensips.org>
> wrote:
>
>> Hi, John!
>>
>> No, I nothing is suspicious. Definitely not f
.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/09/2017 01:50 PM, John Nash wrote:
>
> Do you see anything suspicious in the latest mem dump?
>
> On Wed, Mar 8, 2017 at 7:20 PM, John Nash <john.nash...@gmail.com&
Do you see anything suspicious in the latest mem dump?
On Wed, Mar 8, 2017 at 7:20 PM, John Nash <john.nash...@gmail.com> wrote:
> One more useful info. I disabled drouting functions and just rewrote RURI
> to hardcoded address keeping rest of the functions same and I do no
One more useful info. I disabled drouting functions and just rewrote RURI
to hardcoded address keeping rest of the functions same and I do not see
drop in private memory of that process.
On Wed, Mar 8, 2017 at 4:40 PM, John Nash <john.nash...@gmail.com> wrote:
> OK Here is the dum
at 4:07 PM, Răzvan Crainea <raz...@opensips.org> wrote:
> No, you should not kill any process. Simply send a SIGUSR1 to the process
> you suspect.
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/08/2017 12:28 PM, John Nash wrote:
>
Sorry...Should I kill only the process where i see memory leak?
On Wed, Mar 8, 2017 at 3:41 PM, Răzvan Crainea <raz...@opensips.org> wrote:
> use only memdump set to 1.
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/08/2017 12:11 PM, J
mp).
> So please try to send as many calls as possilble, and if this issue still
> persists, make a pkg memory dump when the server is in idle mode and send
> it over.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/08/2017 11:2
any suggestion for me?..should i try to crash opensips by sending many
calls?
On Tue, Mar 7, 2017 at 4:54 PM, John Nash <john.nash...@gmail.com> wrote:
> version: opensips 2.1.5 (x86_64/linux)
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> DBG_QM_MA
I am using t_on_failure("external_failure"); t_on_reply("external_reply");
before calling do_routing function. I expected failure replies to go
to failure_route[external_failure] only but failure replies also going
to onreply_route[external_reply] along with failure_route[external_failure]
Is
please ignore i had selected wrong option to clean.
On Tue, Mar 7, 2017 at 9:09 PM, John Nash <john.nash...@gmail.com> wrote:
> I was trying to compile on fresh linux cent OS. But when i run make
> menuconfig i find that I am not able to see "Configure Compile Flags "
&
I was trying to compile on fresh linux cent OS. But when i run make
menuconfig i find that I am not able to see "Configure Compile Flags "
seems disabled. Other options I can go inside like exclude modules etc
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an Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/07/2017 12:23 PM, John Nash wrote:
>
> Please note when i call do_routing in such a way that its unable to find
> any rules matching and reject call i do not see free memory drop. But if it
> finds a route, sends c
Please note when i call do_routing in such a way that its unable to find
any rules matching and reject call i do not see free memory drop. But if it
finds a route, sends call to that gateway memory drops with each attempt.
On Tue, Mar 7, 2017 at 3:17 PM, John Nash <john.nash...@gmail.com>
Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/07/2017 11:32 AM, John Nash wrote:
>
> when I check stats after a call attempt pkmem:7-free_size:: 3304280
>
> In this entry with every call I see a drop of 1000 bytes around and this
&
t show any leak.
> Are you sure you are having a private memory leak and not a shared
> memory leak?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/06/2017 08:09 PM, John Nash wrote:
>
> here is another trace
> http://p
here is another trace
http://pastebin.com/9Ge2NEVQ
I see lot of alloc request but no free.
On Mon, Mar 6, 2017 at 6:57 PM, John Nash <john.nash...@gmail.com> wrote:
> Ok will try that. Is it possible that wrong usage of drouting may cause
> this to happen instead of actual l
I don't see any leaks. Perhaps some of those
> fragments increase over time. Can you make a memory dump after the server
> runs some time, like after it gets 100 messages?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/0
> out what is "eating" memory.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/06/2017 02:39 PM, John Nash wrote:
>
> with every call attempt it decreases. I tried some changes by rejecting
> invite before drou
rivate IP that leaks. Next, is
> the memory stabilizing in time? Or it is continously decreasing?
> Yes, that's how you should make the dump.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/06/2017 10:57 AM, John Nash wrote:
>
>
process is the one you are seeing the leak into?
> You can find out using the 'opensipsctl ps' command.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 03/06/2017 09:55 AM, John Nash wrote:
>
> I am using OpenSIPS (2.1.5 (x86_64/l
I am using OpenSIPS (2.1.5 (x86_64/linux)) in production. I observed
private memory is decreasing constantly for one process mainly and
ultimately leading to memory errors and crash.
To debug this issue I prepared a test server and compiled opensips as per
I am using opensips version 2.1 and using dispatcher module. In fact i have
been using it for a while without any problem. But today suddenly all my
gateways started showing status as "probing". I took wireshark trace and
can see that opensips is pinging gateways (freeswitch) and getting reply as
Yes fingerprints are different in Invite and session progress.
On Fri, Jun 24, 2016 at 3:00 AM, sevpal <sev...@aol.com> wrote:
> Take a look at the “fingerprint:” line.
>
> *From:* John Nash <john.nash...@gmail.com>
> *Sent:* Thursday, June 23, 2016 3:42 PM
> *To:* O
DTLS/SRTP cipher 128 to 128 and 256 to 256.
>
> You can print the request body ($rb) on the INVITE with “application/sdp”
> and visually compare the exchange, do this on offer and answer.
>
> *From:* John Nash <john.nash...@gmail.com>
> *Sent:* Thursday, June 23, 2016 3:42 P
y thing that looks bad is that you are retransmitting the ACK which
> FS either ... doesnt like, or is never getting, because it keeps
> retransmitting the 200, which is why you get a 481 when you send BYE.
>
> -Eric
>
>
> On 06/23/2016 01:24 PM, John Nash wrote:
>
lieve revealing ip
>> addresses etc. is any problem - to put my money where my mouth is here is a
>> gist link to an unaltered SIP trace on my server :)
>>
>> https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52
>>
>> -Eric
>>
>>
>> On 0
>>
>> the offer to freeswitch would be:
>>
>> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
>> replace-origin ICE=remove";
>>
>>
>> and the answer back up to the browswer would be:
>>
>> $var(rtpengine_flags) =
back up to the browser.
>
> the offer to freeswitch would be:
>
> $var(rtpengine_flags) = "RTP/AVP replace-session-connection
> replace-origin ICE=remove";
>
>
> and the answer back up to the browswer would be:
>
> $var(rtpengine_flags) = "UDP/TLS
I am following http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2
and trying to test a call
sipml5 --->Opensips + rtpengine > SIP end point (Freeswitch)
But I do not have any audio on both sides. I see this error at rtpengine
log "SRTP output wanted, but no crypto
e functionality was there .
> https://github.com/RestComm/restcomm-android-sdk
> On Jun 22, 2016 5:34 AM, "John Nash" <john.nash...@gmail.com> wrote:
>
>> My objective is to make a native webrtc application which can use SIP
>> over wss for signalling and for m
com> wrote:
> John,
>
> You can utilize sipjs and jssip on account that they utilize sip over
> websocket. Take into consideration that chrome will only allow getusermedia
> if you are using wss and https .
> On Jun 22, 2016 3:07 AM, "John Nash" <john.nash...@gma
Apart from sipml5 is there any native webrtc client also which I can
explore to work with opensips?
The examples I find for webrtc native seem to be using jingle protocol but
in case to make it work with opensips, It has to use SIP/SDP at client end
right?..Any examples?
Git version (2.2) is OK. May be in tar download latest files are not there
no biggi.
On Tue, Jun 21, 2016 at 10:08 PM, John Nash <john.nash...@gmail.com> wrote:
> I downloaded opensips 2.2 stable tar file and upgraded my existing
> opensips.cfg to use wss as per doc
I downloaded opensips 2.2 stable tar file and upgraded my existing
opensips.cfg to use wss as per document
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2
but when I start opensips I get error "cannot handle protocol " right
at the line where I have listener wss:127.0.0.1:443
Do I
I tried this function to delete "NSE" codec from SDP but it doesnt seem to
be working. Any consideration to make it work?...I am using rtpengine
module but calling it after calling codec_delete but this codec still
passes to other end.
___
Users mailing
in the same
> way. IF they have a Route hdr , it may be because they do pre-loaded route
> (the Route points to your SIP server) to be sure the REGISTER gets to the
> registrar server.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-soluti
I am dealing with In-dialog requests using
--
if (has_totag() && (is_domain_local("$rd") || $Ri== "127.0.0.1") &&
is_method("INVITE|ACK|BYE|UPDATE"))
{
# sequential request within a
new algorithm,
> SBT[1], which is very accurate and custamizable.
>
> [1]
> http://www.opensips.org/html/docs/modules/2.2.x/ratelimit.html#id293435
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 06/02/2016 08:49 AM, J
Also how can I decide which Rate limit algorithm should I choose ? Like RED
or TAILDROP or NETWORK
On Thu, Jun 2, 2016 at 9:37 AM, John Nash <john.nash...@gmail.com> wrote:
> I am using opensips(2,1) + freeswitch. At opensips doing auth and
> drouting. Now i plan to test rate limit
I am using opensips(2,1) + freeswitch. At opensips doing auth and drouting.
Now i plan to test rate limit but should I be checking CPS at opensips or
at freeswitch?...as Rate limit uses timers would it be more appropriate to
check at freeswitch?
___
g/Documentation/Script-CoreFunctions-2-2#toc16
> [2] http://www.opensips.org/Documentation/Script-CoreVar-2-2#toc43
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 30.05.2016 11:33, John Nash wrote:
>
> On my linux box
On my linux box we have multiple public IP addressess
1.1.1.1
2.2.2.2
3.3.3.3
I am listening on two of them as
udp:1.1.1.1:5060
udp:2.2.2.2:5060
I have mhomed=1 in my config. I am also using drouting module. What I
expect is when an Invite comes to 1.1.1.1:5060 . drouting should send
outgoing
p(my_IP)+":5060";
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 29.05.2016 16:20, John Nash wrote:
>
> Is it possible to use avp or any other vraiable as argument
> to force_send_socket ?
Is it possible to use avp or any other vraiable as argument
to force_send_socket ?
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Thank you Pete.
On Thu, May 19, 2016 at 2:44 PM, Pete Kelly <pke...@gmail.com> wrote:
> I think this may be the video https://www.youtube.com/watch?v=3XYcQQCWylw
>
> On 17 May 2016 at 20:41, John Nash <john.nash...@gmail.com> wrote:
>
>> I saw
>> http:/
ixed:
>
> https://github.com/OpenSIPS/opensips/commit/4b0fca533cd7be4a45c1381c78f2b37aaba6152b
>
> Please update from GIT and let me know if you still have the problem.
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-soluti
I saw
http://www.opensips.org/pub/events/2012-08-07_ClueCon_Chicago/VLAD_PAIU-OpenSIPS-Securing_SIP_Networks.pdf
.
I would love to watch the video session of this, is there any place I can
get the video? Tried searching google but did not find.
Regards
John
in the logs ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 12.05.2016 08:46, John Nash wrote:
>
> Actually crash happened shortly after we uploaded 11000 codes but looks
> like it is not related to drouting. I
012Please help us make OpenSIPS better by reporting it at
https://github.com/OpenSIPS/opensips/issues#012
In log file I see following messages time to time
ERROR:core:pv_get_contact_body: failed to parse contact hdr
On Wed, May 11, 2016 at 11:29 PM, John Nash <john.nash...@gmail.com> wrote:
I have been using drouting module with just 200 entries from 8 months
yesterday we had need of adding around 11000 entries in rules table but
after that opensips started to crash. I am currently using -m 2048 -M 1024
isn't it enough memory?
How can I anticipate memory usage?
John
ps.org>
wrote:
> Hi John,
>
> either dig into M4 secrets , either simply change your password to avoid
> the hash char :)
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 05.05.2016 18:20, John
I am trying to generate .cfg file with the help of m4. I have following
line in my opensips.cfg.m4
DB_USER:DB_PASS@DB_IP/DB_NAME
In defines.m4 I have corresponding values.
The issue is my DB_PASS contains "#" as one of the character and because of
that any word after DB_PASS is not being
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 29.04.2016 19:26, John Nash wrote:
>
> OK. Any way to use Xlog to print DBG messages even if script has debug
> level as ERR ?
>
> On Fri, Apr 29, 2016 at 9:2
Is there any way to log messages (Custom messages and SIP trace) from
script for a given parameter say IP or ruri.
A crude way can be to store say user in local cache and match with the user
in script and log else pass but ..
1- I am not sure if any other smart way to do it
2- How can I dunp SIP
eloperhttp://www.opensips-solutions.com
>
> On 19.04.2016 17:49, John Nash wrote:
>
> Ok got it thanks. I also noticed that transactions cancelled because of
> fr_inv_timeout , CDR records as "Request timeout". It is quite confusion,
> shouldnt it be "Request Termin
I am using fr_inv_timer and fr_timer and logging failed transactions, but
in both cases I get request timeout. Can I control this somehow so that I
log "Time out" only in case fr_timer expires and record something else in
case fr_inv_timer?
___
Users
Extend the coverage of the preocessing context and TM context over the
> cancel_branch() function (in the timeout handler) so the TH callbacks can
> reach back the dialog and do the TH related changes.
> Reported by Julian Santer on mailing list.
>
> Kind regards,
> Julian
t;> Hi Julian,
>>>>
>>>> I will have to test this and come back to you.
>>>>
>>>> Regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>> OpenSIPS Founder and Developer
>>>> http://www.opensips-solutions.com
I am not able to find the commit can you please point me?
On Wed, Jan 13, 2016 at 9:07 AM, John Nash <john.nash...@gmail.com> wrote:
> OK thank you I will try to find and patch.
>
> On Tue, Jan 12, 2016 at 10:58 PM, Răzvan Crainea <raz...@opensips.org>
> wrote:
>
&g
ystem to delete the
> stale key/value from the cachedb store.
>
> This may or may not address your concern, but I hope it helps.
>
> Jarrod
>
> > On Jan 12, 2016, at 1:54 PM, John Nash <john.nash...@gmail.com> wrote:
> >
> > I am using local cache db module in
OK thank you I will try to find and patch.
On Tue, Jan 12, 2016 at 10:58 PM, Răzvan Crainea <raz...@opensips.org>
wrote:
> Hi, John!
>
> This issue was fixed in newer versions of OpenSIPS, and the fix will be
> part of OpenSIPS 2.1.2.
>
> Best regards,
> Răzvan
>
&g
I am using OpenSIPS (2.1.1 (x86_64/linux)) with dialogs module and topology
hiding modules.
I am not saving dialogs in database.
After I run it for few hours and stop traffic i see hundreds of dialogs
using fifo command which wont be deleted from memory until I restart. I see
dialogs like ..
I am using local cache db module in order to keep user id and password in
memory and now plan to keep some other data in memory too and that I want
to keep in some centralized cache.
It will be like Opensips1, Opensips2 <-> Cache server <->
Postgresql
I also need to periodically
I am using Create Dialog with "Pp" as parameter but I have one doubt, what
if the either side of Opensips is another proxy or SBC which has following
code at the start (Before loose route or topology hiding check)
if(((is_method("NOTIFY") && $hdr(Event) =~ "keep-alive") ||
is_method("OPTIONS")))
I get following warning in log
Dec 1 08:40:14 [14516] WARNING:dialog:dlg_onroute: tight matching failed
for BYE with callid='[%!<
to}'/36, ftag='9r60aB1Q77tUj'/13, ttag='PROXY1448958761787to'/20 and
direction=0
Dec 1 08:40:14 [14516] WARNING:dialog:dlg_onroute: dialog identification
elements
I have couple of things i need your valuable inputs I have already seen
some articles and slides but some questions remain...
1- AVP db queries do we need to escape parameters or its taken care of by
module internally.
2- How can I secure opensipsctl and mi_datagram as that is gateway to my
ndrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 17.11.2015 08:34, John Nash wrote:
>
> I am using Drouting module and quite happy with it. I have some new
> requirement where I need to be able to strip some digits from "incoming"
>
I am using Drouting module and quite happy with it. I have some new
requirement where I need to be able to strip some digits from "incoming"
number and add some prefix at dr_rule level. I know I can do that at
dr_gateway wise but as per my new requirement I need to use same gateway
for routing for
Anyone has thoughts on this?.If i use record routing instead of topology
hiding would it help?.
On Wed, Jul 15, 2015 at 10:47 AM, John Nash <john.nash...@gmail.com> wrote:
> Dear Vlad,
>
> Do you need any more information? Like debug log or complete wireshark
> pcap?
>
>
you can
> route the call to actual IP of user2 via lookup("location").
>
> To translate between the 2 number and User2 (before the lookup),
> use the aliases module.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-so
I think I can use local_cache and store caller id as key and sip user as
value. Right?
On Thu, Sep 24, 2015 at 8:26 AM, John Nash <john.nash...@gmail.com> wrote:
> OK..I also looked at alias module. I am really fascinated by cache any way
> to use alias db with cache?
>
> On
I am trying to test a use case in which I need to send call to a registered
user based on its caller ID.
User1
Username = 1001
Carrier id = 11
User2
Username = 1002
Carrier id = 2
Both are registered to my opensips. Now if User1 calls number 2 , I
want opensips to try to
On 12.08.2015 09:52, John Nash wrote:
I am not sure if its some bug or my mistake.
I am using topology hiding module (opensips 2.1 version) and I have
noticed that Call-id in Cancel message is different than Invite sent to
gateway.
Invite is sent to gateway and we get session progress
I am not sure if its some bug or my mistake.
I am using topology hiding module (opensips 2.1 version) and I have noticed
that Call-id in Cancel message is different than Invite sent to
gateway.
Invite is sent to gateway and we get session progress but call is not
picked up, as per fr_timer
Dear Vlad,
Do you need any more information? Like debug log or complete wireshark pcap?
John
On Wed, Jun 24, 2015 at 10:06 PM, John Nash john.nash...@gmail.com wrote:
I am using opensips 2.1 with topology_hiding module. I have an issue only
with one SIP endpoint. This endpoint sends Ack
Hello Frank,
I saw your message
http://lists.opensips.org/pipermail/users/2015-March/031296.html Did you
get any head or tail of this issue?..I also face the same situation where
in wireshark I see perfect messages but opensips log shows unable to parse
and shows junk characters
John
On Thu,
It seems like media proxy flag related issue to me. My guess is from first
gateway which fails you get session progress and then it rejects and you
get another session progress from second gateway. If you play around with
mediaproxy flags (at the time of session progress and 200 OK) it can be
than the 200 OK received (It only replaced .0 with
.2). I think this is happening because I am also using drouting module and
UAC which use branches.
I can post trace also if someone wants to have a look.
On Wed, Jun 24, 2015 at 10:06 PM, John Nash john.nash...@gmail.com
wrote:
I am using
-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 15.06.2015 22:08, John Nash wrote:
Hello Bogdan,
Thank you I will just ignore them. I have one more related issue. I am
using uac_replace_from in auto mode along with topology_hiding. In a case
when UA sends
Is it required for SIP message to have a to-tag in order to be matched with
dialog using topology_hiding_match or match_dialog?
In one situation ACK message from UA does not have to-tag but From-tag,
call-id and all headers seems to belong to ongoing dialog? Is there a way
to match such request
received (It only replaced .0 with .2). I think
this is happening because I am also using drouting module and UAC which use
branches.
I can post trace also if someone wants to have a look.
On Wed, Jun 24, 2015 at 10:06 PM, John Nash john.nash...@gmail.com wrote:
I am using opensips 2.1
I am using opensips 2.1 with topology_hiding module. I have an issue only
with one SIP endpoint. This endpoint sends Ack message (after 200 OK to
Invite) without any to tag because of that it is not matching with In
dialog request section.
Can a UA send ACK without to tag?...If yes any way I can
I have modified my proxy config to support topology_hiding function of
dialog module. But I see lot of dialog related errors like ..
ERROR:dialog:push_reply_in_dialog: [487] reply in dlg state [2]: missing
TAG param in TO hdr
ERROR:dialog:w_validate_dialog: null dialog
I am just wondering if my
of receiving in Early state a reply without tag
param in TO header - something like that is bogus.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 15.06.2015 12:32, John Nash wrote:
I have modified my proxy config to support
I had started testing 1.X series taken from github master branch couple of
months ago (It shows version as Server:: OpenSIPS (1.12.0dev-notls
(x86_64/linux))
Now I need to install it in one of the production server and before I do
that I want to update to the latest version of 1.X series. Which
-solutions.com
On 08.03.2015 17:59, John Nash wrote:
I had started testing 1.X series taken from github master branch couple
of months ago (It shows version as Server:: OpenSIPS (1.12.0dev-notls
(x86_64/linux))
Now I need to install it in one of the production server and before I do
that I want
As per documentation pike module can be implemented manual as well as
automatic. The way I understand it manual mode will not monitor (Not even
queue) packets for which pike_check_req() is not called and it gives
performance advantage as we can skip this call for trusted IPs.
First of all is my
I have used opensips+rtpproxy for years for simple scenarios but now I am
trying to use it with serial forking. My flow is
UA---Invite ---Opensips
1 Branch
---Media Server
I tested drouting module and it is very good but when I try to change value
of one custom header (Header was added in initial Invite), I see it added
twice.
--Initial Invite---(Header added)Sent to dest1
---Failure comes from dest1--- Sent to dest2 (Using
use_next_gw)---Remove
I am testing a setup where opensips sending call to freeswitch and if call
is rejected by freswitch a custom header X-internal-hangup. In opensips
failure_route I am trying to check it using is_present_hf() function but it
never reaches inside conditions. In wireshark I see this header.
Crainea
OpenSIPS Solutionswww.opensips-solutions.com
On 01/15/2015 12:40 PM, John Nash wrote:
I am testing a setup where opensips sending call to freeswitch and if
call is rejected by freswitch a custom header X-internal-hangup. In
opensips failure_route I am trying to check it using
and Developerhttp://www.opensips-solutions.com
On 12.01.2015 11:55, John Nash wrote:
OK..I think doing accounting in exec makes perfect sense.
On Mon, Jan 12, 2015 at 2:36 PM, Bogdan-Andrei Iancu bog...@opensips.org
wrote:
Hi John,
Indeed, depending on the nature of the query, some answers can
,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 11.01.2015 15:48, John Nash wrote:
Hello Bogdan,
Thank you. Cache features are really good and I am using for Register and
Invite auth but I need to run a query to find out allowed duration for a
call
I am testing one setup where opensips drouting module sends call to
Freeswitch and I encountered one situation ...
UA sends Invite to opensips, opensips uses drouting module and sends Invite
to Freeswitch , callee rejects the call and opensips sends ACK to
freeswitch and sends second invite
/dialog.html#id294001)
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 09.01.2015 20:19, John Nash wrote:
I have used opensips for load balancing and some border proxy+
NAT+rtpproxy in past and am quite happy with it. Recently I decided to add
DB
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