Group,
I've been looking for SIP client that can send PIDF-LO information within
the SIP INVITE body. But so far I haven't found it yet.
I'd like to try the emergency module as described in this tutorial. In
Section 4.2 (Scenario 1, step no.2) it describes a SIP client sends INVITE
with PIDF-LO i
Hi Jonas,
Thanks for the additional information. Indeed, what you say makes sense,
but before making a call (and it is not a SIP call :) ), I need a fresh
mind tomorrow morning - to go again through the scenario, through the
RFC and all the logic aspects here.
Best regards,
Bogdan-Andrei Ia
Hi Jason,
That command is not available in 1.7.x
Consider placing some xlog()'s in your script and to enable debug level
4 to get an idea of what is going on with your processing.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 28.09.2015 21:
Great!
Thanks.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org em nome
de Max Mühlbronner
Enviado: segunda-feira, 28 de setembro de 2015 12:44
Para: users@lists.opensips.org
Assu
Sorry, copy&paste mistake.
if(is_present_hf("X-Timeout")){
$DLG_timeout = $(hdr(P-Source-IP));
On 28.09.2015 17:40, Max Mühlbronner wrote:
Hi,
seems to be a simple solution, without overhead/database/...
if(is_present_hf("P-Source-IP")){
$DLG_timeout = $(hdr(P-Source-IP
Hi,
seems to be a simple solution, without overhead/database/...
if(is_present_hf("P-Source-IP")){
$DLG_timeout = $(hdr(P-Source-IP));
}else{
$DLG_timeout = 3600;
}
But you could also save the information into a e.g. mysql/... database
and pull it from the db. (check out avpop
Hi Bogdan.
I have just finished studying about some parts of OpenSIPS documentation,
mainly about the modules.
I saw that it is possible to take control of call duration using Call
Control,which is too complex to my simple needings as my projetct doesn't need
a charging system, or using DLG
Hi,
Sorry, I thought I did buy no, the request-uri will NOT be
"sip:public_ip:5061", if it was it would not find the correct
connection since it is not stored under that key. Due to
"fix_nated_contact", as part of the REGISTER flow, the Contact of that
REGISTER will now be "sip:public_ip:public_po
That "3" you can load it from DB (via avp_db_query()) or you can get it
from user profile (load_credentials) or it can be the result of a
translation (dp_translate based on SIP domain).
It is up to you where you get the value from - the important part is
that you can pass it via a variable to
Hi, Tito!
So you can detect the event, but you do not see any information attached
to it?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 09/22/2015 11:09 PM, Tito Cumpen wrote:
Group,
I am noticing issues with 2.2 dev in reference to sending params when
rai
Hi, Tito!
I am not sure I understand your scenario. So you first send a message to
a client (is he a WS client?) and then, if that fails, you failover to a
different server(AS server, not WS)?
Can you post OpenSIPS debugging logs on pastebin? You can email them
personally if you don't want to
Use a variable then :)
$avp(grp) = 3;
do_routing("$avp(grp)");
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 28.09.2015 15:31, Sasmita Panda wrote:
Hi Andrei,
Yes , I know the do_routing() reflects the routing groups . But in
my case i
Hi Hamid,
I see you to Ctrl-C the "trapping" ; normally it should terminate by
itself, after the backtrace is taken from all processes.
Try to do "opensipsctl fifo ps" get the PID of a worker process and try
to attached with gdb to it, to see what is doing.
BTW, to avoid confusions, the stu
Hi,
The db_cachedb module is only compatible to the cachedb_mongo.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 10.09.2015 21:27, Sasmita Panda wrote:
Hi All ,
I am trying to compile openisps-1.11 with db_cachedb and
cachedb_redis mo
Hi Matt,
That is not actually an error, is OpenSIPS complaining that is not able
to open a TCP connection towards the wanted destination (one of yours
UACs) - see the "Connection refused" error log.
Why ? if there is no limitation on the network level (NAT, firewall,
etc), then it means the
OK, thank you !
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 28.09.2015 14:58, Pete Kelly wrote:
Done https://github.com/OpenSIPS/opensips/issues/656
On 28 September 2015 at 12:32, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:
That is
Hi Surya,
This question is more appropriate for the devel mailing list :).
The "turns" is a mechanism to force OpenSIPS to process the PUBLISH
requests in the some order as they were received. OpenSIPS is a
multi-processes application and requests received in a certain order
from the network
Hi Aron,
No, we do not have, but it is an interesting idea, so please open a
feature request on the GITHUB tracker.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 11.09.2015 00:56, Podrigal, Aron wrote:
Hi,
Is there a way to get the line nu
Done https://github.com/OpenSIPS/opensips/issues/656
On 28 September 2015 at 12:32, Bogdan-Andrei Iancu
wrote:
> That is true Pete,
>
> Could you open a bug report on the GITHUB tracker please ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.co
Hi Pete,
I assume you do rtpproxy_answer() for the 200 OK on B leg, right ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 23.09.2015 11:44, Pete Kelly wrote:
I am using rtpproxy with parallel fork and noticed some interesting
behaviour (
Hi Bogdan-Andrei.
I will take a look in the first link now. I have already read about dlg_end_dlg
and I'm using it with success, following another hint from you.
Thank you very much!
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
_
That is true Pete,
Could you open a bug report on the GITHUB tracker please ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 28.09.2015 14:27, Pete Kelly wrote:
On 28 September 2015 at 12:23, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>>
On 28 September 2015 at 12:23, Bogdan-Andrei Iancu
wrote:
> Hi Pete,
>
> You mean that ideally the username part of Contact URI should be preserve
> through TH ?
>
Yes, and also the domain, which would need to be preserved if the UAS
wanted to 302 to a completely different URI?
>
> Regards,
>
Hi Pete,
You mean that ideally the username part of Contact URI should be
preserve through TH ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 23.09.2015 11:40, Pete Kelly wrote:
This is something I have noticed for a while now and came acros
Hi Nabeel,
I see you have a SIP msg with IPV6 in RURI:
sip:+44798494@2a04:4a41:218:c8d2::2383:20a5%4:39572;transport=tls
In this case, the IPv6 address must be between brackets, like:
sip:+44798494@[2a04:4a41:218:c8d2::2383:20a5%4]:39572;transport=tls
Regards,
Bogdan-Andrei Iancu
OpenS
Hi Kneeoh,
Do you have a backtrace for the crash ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 14.09.2015 17:30, Kneeoh wrote:
Hi I've been using the couchbase module for cachedb for a while now
with no issues. Recently I started tracking a
Hi Sasmita,
The param of do_routing() has to reflct the routing group you want to
use (the groupid in the dr_rules table) .
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 28.09.2015 13:26, Sasmita Panda wrote:
Hi All ,
I mean to say , w
Hi Jonas,
One question (which remained unanswered): when the call comes back to
opensips (from main server to Alice, let's say), what it the the RURI?
Is "sip:public_ip:5061" (as per contact in REGISTER) ? If you use
fix_nated_contact() at REGISTER time, does the RURI in INVITE changes to
"si
Hi Alcindo,
Not sure what are you asking here, but if you need help, you need to ask
it for a particular issue (to report something that does not work) and
not in a general way. General reviewing for scripts is out of the scope
for the people here.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Foun
Hi All ,
I mean to say , when there is more that one group and for each group
there is different gateways then how I will set the do_routing parameter
inside my script ?
If I have a single group then I am setting it as do_routing("1") . But
for different groups this logic wont work . I do
Hi,
I'm getting the followinh error with IPv6 when attempting to make a call.
Is it related to the SIP client or OpenSIPS?
09-28 03:02:33.564 18214-21831/com.sipdomain I/IntegratedSipProvider﹕
message:
SIP/2.0 500 Internal Error
Via: SIP/2.0/TLS
[2a04:4a41:218:c8d2::2383:20a5]:34248;rece
Hi Rodrigo,
The timeout for a call can be set (in a per-call manner) when the call
starts (during re-INVITE) or during any sequential requests (ACK,
re-INVITE), etc. See DLG_timeout variable:
http://www.opensips.org/html/docs/modules/1.11.x/dialog.html#timeout-pvar-id
If you want to tear down
Sasmita,
The parameter from "is_from_gw(n)" must be aligned with the "type" you
have for the gws, if you want to check comes from a particular set of
gateways (with a certain type).
If you want to check against all gws (any type), simply do
"is_from_gw()" with no parameter.
Regards,
Bogda
Hi Bogdan,
Sorry if it sounds stupid. I just want to ask more about your following line.
"If you get those errors means your OpenSIPS has no single process able to do
any kind of processing."
No I don't have any traffic on that server.
[root@aws-dev-ec2-sipserver tmp]# opensipsctl trapINFO: Trap
Thierry Luo:
我对 sip 各 rfc protocol 都很熟的。 但只做兼职工作。适合独立完成项目。
不知您的工作需求是什么？
George Wu
在 2015-09-28 14:22:34，"Thierry Luo" 写道：
很报歉通过这个邮件列表打扰大家。
我们是一个创业团队，刚刚拿到投资，正在组建核心团队，现需要一位在OpenSIPS方面有经验的C/C++开发工程师加盟，待遇优厚，期权可谈，职位在北京。请有意者速与luoyongh...@nane.cn联系。谢谢！
打扰见谅！
罗
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