Re: [OpenSIPS-Users] (no subject)

[Alexey Kazantsev via Users]

Hey Liviu,
 
thank you for the informative answer!
 
By the way, I configured a full-sharing usrloc cluster
and it seems to be what I need.
 
---
BR, Alexey
http://alexeyka.zantsev.com/
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Re: [OpenSIPS-Users] Example of configuration "Full Sharing" Topology with NoSQL

[Alexey Kazantsev via Users]

Hey Artiom,
 
the main difference in call-flow between federated and full-sharing
architecture is as follows:
 
 
Federated cluster:
  
 ++++ 
 |||| 
 | Osips1 |--> | Osips2 |--->UAC_2
 ++++ 
  ^   
  |   
  |   
  |   
   UAC_1  
  
  
  
  
  
Full-sharing cluster:  
  
+++-+ 
||| | 
|Osips1  ||Osips2   | 
++-\  +-+ 
 ^  --\   
 | ---\   
 | --\
  UAC_1   -> UAC_2
 
 
 
 
So, as already written in the documention, the full-sharing
architecture requires either an edge SBC or a lack of any
network/IP routing problems between UACs and all OpenSIPS nodes.
 
---
BR, Alexey
http://alexeyka.zantsev.com/
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[OpenSIPS-Users] change_reply_status - dropping SDP from 183?

[Monideth Pen]

Hi,

I am able to map 183 to 180 using the change_reply_status() function.

However, I would also like to drop SDP if it is present in the 183. How
could I achieve this?

Thank you.
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[OpenSIPS-Users] Opensips crash while connecting mariadb

[RAVI bhatt]

Hello all,
i am using opensips 3.0.1 with mariadb 10.4.8 and i am facing issue  in
starting opensips. As opensips crash with below given message:
ct 15 03:27:17 [22849] DBG:core:find_mod_export: found  in
module db_mysql [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:db_bind_mod: using db bind api for db_mysql
Oct 15 03:27:17 [22849] DBG:core:db_do_init: connection 0x7fba0e093510 not
found in pool
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno:
2000

i have attached logs and back  trace from core  so please provide your
suggestions.
please note credentials given in db_url are working while using from msyql
client

Thanks in advance.
Ravindra Bhatt
Oct 15 03:27:17 [22849] INFO:core:evi_publish_event: Registered event 

Oct 15 03:27:17 [22849] DBG:core:find_cmd_export_t: found  in module 
tm [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:find_cmd_export_t: found  in module 
rr [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:find_mod_export: found  in module 
db_mysql [/usr/local/lib64/opensips/modules/]
Oct 15 03:27:17 [22849] DBG:core:db_bind_mod: using db bind api for db_mysql
Oct 15 03:27:17 [22849] DBG:core:db_do_init: connection 0x7fba0e093510 not 
found in pool
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000
Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: 2000

Got ERROR: "InnoDB: Operating system error number 11 in a file operation." 
errno: 2000
Got ERROR: "InnoDB: Error number 11 means 'Resource temporarily unavailable'" 
errno: 2000
Got ERROR: "InnoDB: Cannot open datafile '/var/lib/mysql/ibdata1'" errno: 2000
Got ERROR: "InnoDB: Could not open or create the system tablespace. If you 
tried to add new data files to the system tablespace, and it failed here, you 
should now edit innodb_data_file_path in my.cnf back to what it was, and remove 
the new ibdata files InnoDB created in this failed attempt. InnoDB only wrote 
those files full of zeros, but did not yet use them in any way. But be careful: 
do not remove old data files which contain your precious data!" errno: 2000
Got ERROR: "InnoDB: Plugin initialization aborted with error Cannot open a 
file" errno: 2000
Got ERROR: "Plugin 'InnoDB' init function returned error." errno: 2000
Got ERROR: "Plugin 'InnoDB' registration as a STORAGE ENGINE failed." errno: 
2000
Got ERROR: "unknown: Can't lock aria control file 
'/var/lib/mysql/aria_log_control' for exclusive use, error: 11. Will retry for 
30 seconds" errno: 2000



^C^CGot ERROR: "unknown: Got error 'Could not get an exclusive lock; file is 
probably in use by another process' when trying to use aria control file 
'/var/lib/mysql/aria_log_control'" errno: 2000
Got ERROR: "Plugin 'Aria' init function returned error." errno: 2000
Got ERROR: "Plugin 'Aria' registration as a STORAGE ENGINE failed." errno: 2000
Got ERROR: "Unknown/unsupported storage engine: InnoDB" errno: 2000
 DBG:db_mysql:db_mysql_connect: opening connection: 
mysql://:@localhost/cc_master
CRITICAL:core:sig_usr: segfault in attendant (starter) process!
DBG:core:restore_segv_handler: restoring SIGSEGV handler...
 DBG:core:restore_segv_handler: successfully restored system SIGSEGV handler
^CSegmentation fault (core dumped)

(gdb) bt full
#0  intern_plugin_lock (lex=0x0, state_mask=14, rc=0x0)
at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_plugin.cc:948
pi = 0x0
#1  plugin_thdvar_init (thd=0x28af578) at 
/usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_plugin.cc:3155
old_table_plugin = 0x0
old_tmp_table_plugin = 0x0
old_enforced_table_plugin = 0x0
#2  0x7fc7d58722b1 in THD::init (this=this@entry=0x28af578)
at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_class.cc:1177
No locals.
#3  0x7fc7d587310a in THD::THD (this=0x28af578, id=, 
is_wsrep_applier=)
at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_class.cc:798
tmp = 
#4  0x7fc7d57f2bdc in create_embedded_thd 
(client_flag=client_flag@entry=-2143837683)
at /usr/src/debug/MariaDB-10.3.18/src_0/libmysqld/lib_sql.cc:685
thd = 0x7fff689c20b0
#5  0x7fc7d57fa1e4 in mysql_real_connect (mysql=0x7fc7da7190b0, 
host=, user=, 
passwd=, db=0x7fc7da717620 "cc_master", port=port@entry=0, 
unix_socket=unix_socket@entry=0x0, 
client_flag=2151129613, client_flag@entry=2147549184)
at /usr/src/debug/MariaDB-10.3.18/src_0/libmysqld/libmysqld.c:179
name_buff = "\200\314|\326\307\177\000\000\060!\234h\377\177\000\000 
!\234h\377\177\000\000\060!\234h

Re: [OpenSIPS-Users] OpenSIPs as Registration server in front of Asterisk

[Todd Routhier]

Yes the end point phones are behind NAT but reach is behind a different
NAT. Typically one or two phones at each location. NATs are different at
each location so I'm sure this is why some work and some don't.

None of them use STUN but this is because I never had to use STUN for any
of these same runs points when registered directly to Asterisk.


On Wed, Oct 16, 2019, 2:07 AM Răzvan Crainea  wrote:

> Hi, Todd!
>
> Can you provide a pcap of one of the calls that are not working?
> Also, are these clients behind NAT? Do they use STUN?
>
> Best regards,
> Răzvan
>
> On 10/15/19 9:01 PM, Todd Routhier wrote:
> > Problem: Calls from PSTN provider > Asterisk > OpenSIPs > SIP Endpoint
> > have intermittent audio issues. See below for details.
> >
> > I am a long time Asterisk user but extremely new to OpenSIPs.
> >
> > We are in the process of a migration from an older Asterisk server to a
> > newer version along with some other changes.
> >
> > First order of business is for us to offload all registrations from our
> > current 1.8.x Asterisk server to OpenSIPs 2.4.6.
> >
> > We have a setup that seems to be mostly working but intermittent audio
> > issues are what we are trying to eliminate.
> >
> > When I say intermittent, audio seems to work for a particular end
> > point in certain situations or it doesn't. For example, we have some end
> > points which have no audio at all such as my personal soft-phone. I
> > can't get audio on any of three different soft-phones on my laptop, no
> > audio in either direction. But, I have a Grandstream phone on the same
> > LAN which works perfectly every time, on every call.
> >
> > I have other end points which are Grandstream phones with perfectly
> > working audio in both directions, all the time, consistently.
> >
> > I have other Grandstream end points which work for the same type of call
> > every time, with audio in both directions but the same phone has no
> > audio on slightly different types of calls (hard to explain what I mean
> > by "types of calls").
> >
> > Ideally, we would not care about this working with Asterisk 1.8.x since
> > we are moving away from it but it's important for it to work as part of
> > our transition/migration.
> >
> > I had horrible audio issues at first were it was hardly working at all
> > or one way audio consistently. I fixed this by setting nat=yes in the
> > sip.conf for the context pointing to the OpenSIPs server. I couldn't
> > understand why this fixed it since the OpenSIPs server and the Asterisk
> > server both have static IP's and are NOT behind any NAT of any sort.
> > Only the end points registered to OpenSIPs are behind end points.
> >
> > Still I noticed that Asterisk was trying to send calls to the LAN IP of
> > the end points, so I tested nat=yes and it fixed most of the audio
> > issues with only the issues outlined above remaining.
> >
> > My next steps are to see if I have good audio if I push calls to the
> > newer Asterisk server then to the end points registered to the OpenSIPs
> > server. Even if that works, it does not solve my current need to make
> > this work with Asterisk 1.8.x at least until the migration is complete.
> >
> > Thanks in advance for any assistance with this.
> >
> > Regards,
> >
> > Todd
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
>http://www.opensips-solutions.com
>
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[OpenSIPS-Users] Double record route with TLS

[Callum Guy]

Hi All,

Using 3.0.1

I'm dealing with a call scenario where a UA is dialling out using TLS and
travelling between two geographic sites, both of which use NAT. TLS is only
required between the UA and the initial registrar, once the initial
requests hit the network it is forwarded using cleartext UDP. Instances
listen on the internal address only

The problem I am facing is that double_rr is not using my defined
advertised address of the public IP, instead it replaces one of the
addresses and not the other causing a routing breakdown later in the
dialog. I'm using set_advertised_address(SITEA-PUB-IP)

To illustrate, the following headers are presented at site B:

INVITE sip:01234567890@192.168.153.226 SIP/2.0
Record-Route:

Record-Route:


For correct routing I need both of those to present the public address. In
a typical NAT traversal scenario I have reply routes which re-write the
public and private addresses accordingly and I can certainly intercept
these INVITE messages and do that however I wondered if there was a cleaner
way. Am I missing a trick?

I have attempted to make changes using a site A branch route to rewrite the
RR however the header is not yet formed. I also attempted to use
record_route_preset() however this did not have the desired effect.

Any ideas would be very much appreciated!

Callum

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Re: [OpenSIPS-Users] change_reply_status - dropping SDP from 183?

[Fabian Gast]

You can use something like

loadmodule "sipmsgops.so"

onreply_route[myreply] {

if (t_check_status("183")) {
change_reply_status("180", "Ringing");
remove_body_part("application/sdp");
}
}

Fabian

- Ursprüngliche Mail -
Von: "Monideth Pen" 
An: "OpenSIPS users mailling list" 
Gesendet: Freitag, 18. Oktober 2019 10:06:42
Betreff: [OpenSIPS-Users] change_reply_status - dropping SDP from 183?

Hi, 
I am able to map 183 to 180 using the change_reply_status() function. 

However, I would also like to drop SDP if it is present in the 183. How could I 
achieve this? 

Thank you. 

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Re: [OpenSIPS-Users] Example of configuration "Full Sharing" Topology with NoSQL

[Social Boh]

Hello,

if i don't want use a SBC how can I known is a node of cluster is up or
down? which method do you advise?

Thank you

---
I'm SoCIaL, MayBe

El 28/10/2019 a las 03:21, Alexey Kazantsev via Users escribió:
> Hey Artiom,
>  
> the main difference in call-flow between federated and full-sharing
> architecture is as follows:
>  
>  
> Federated cluster:
>   
>  ++++ 
>  |||| 
>  | Osips1 |--> | Osips2 |--->UAC_2
>  ++++ 
>   ^   
>   |   
>   |   
>   |   
>UAC_1  
>   
>   
>   
>   
>   
> Full-sharing cluster:  
>   
> +++-+ 
> ||| | 
> |Osips1  ||Osips2   | 
> ++-\  +-+ 
>  ^  --\   
>  | ---\   
>  | --\
>   UAC_1   -> UAC_2
>  
>  
>  
>  
> So, as already written in the documention, the full-sharing
> architecture requires either an edge SBC or a lack of any
> network/IP routing problems between UACs and all OpenSIPS nodes.
>  
> ---
> BR, Alexey
> http://alexeyka.zantsev.com/
>  
>
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[OpenSIPS-Users] Fwd: [CFP] FOSDEM 2020, RTC devroom, speakers, volunteers neeeded

[fosdem-rtc-admin]

FOSDEM - Real Time Communications devroom CfP
=

Overview


[FOSDEM](https://fosdem.org) is one of the world's premier meetings of
free software developers, with over five thousand people attending each
year. FOSDEM 2020 takes place 1-2 February 2020 in Brussels, Belgium.

This document contains information about:

-   Real-Time Communications developer room (devroom) and lounge
-   speaking opportunities
-   volunteering in the devroom and lounge
-   social events (the legendary FOSDEM Beer Night and Saturday night
dinners provide endless networking opportunities)
-   the Planet aggregation sites for RTC blogs

**NEW:** Save yourself entering Free-RTC events and CFP deadlines into
your calendar and task list, follow our iCalendar feed:
https://freertc.org/events.ics

Call for participation - Real Time Communications (RTC)
---

The Real-Time devroom and Real-Time lounge are about all things
involving real-time communication, including: XMPP, SIP, WebRTC,
telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption.

**We are looking for speakers for the devroom and volunteers and
participants for the tables in the Real-Time lounge.**

The devroom is only on Sunday, 2nd of February 2020. The lounge will be
present for both days.

To discuss the devroom and lounge, please join the [Free-RTC mailing
list](http://lists.freertc.org/mailman/listinfo/discuss).

### Speaking opportunities

Note: if you used FOSDEM Pentabarf before, please use the same
account/username

Real-Time Communications devroom: deadline 23:59 UTC on 15th of
December. Please use the
[Pentabarf](https://penta.fosdem.org/submission/FOSDEM20/) system to
submit a talk proposal for the devroom. On the "General" tab, please
look for the "Track" option and choose "Real Time Communications
devroom".

Other devrooms and lightning talks: some speakers may find their topic
is in the scope of more than one devroom. It is encouraged to apply to
more than one devroom and also consider proposing a lightning talk, but
please be kind enough to tell us if you do this by filling out the notes
in the form. Here you can find the [full list of
devrooms](https://www.fosdem.org/2020/schedule/tracks/) and here you can
apply for a [lightning talk](https://fosdem.org/submit).

### First-time speaking?

FOSDEM devrooms are a welcoming environment for people who have never
given a talk before. Please feel free to contact the devroom
administrators personally if you would like to ask any questions about
it.

### Submission guidelines

The Pentabarf system will ask for many of the essential details. Please
remember to re-use your account from previous years if you have one.

In the "Submission notes", please tell us about:

-   The purpose of your talk
-   Any other talk applications (devrooms, lightning talks, main track)
-   Availability constraints and special needs

You can use HTML and links in your bio, abstract and description.

If you maintain a blog, please consider providing us with the URL of a
feed with posts tagged for your RTC-related work.

We will be looking for relevance to the conference and devroom themes,
presentations aimed at developers of free and open source software about
RTC-related topics.

Please feel free to suggest a duration between 20 minutes and 55 minutes
but note that the final decision on talk durations will be made by the
devroom administrators based on the number of received proposals. As the
two previous devrooms have been combined into one, we may decide to give
shorter slots than in previous years so that more speakers can
participate.

Please note FOSDEM aims to record and live-stream all talks. The CC-BY
license is used.

Volunteers needed
-

To make the devroom and lounge run successfully, we are looking for
volunteers:

-   FOSDEM provides video recording equipment and live streaming,
volunteers are needed to assist in this
-   Organizing one or more restaurant bookings (dependending upon number
of participants) for the evening of Saturday, 1 February
-   Participation in the Real-Time lounge
-   Circulating this Call for Participation to other mailing lists

Social events and dinners
-

The traditional FOSDEM beer night occurs on Friday, 31st of January.

On Saturday night, there are usually dinners associated with each of the
devrooms. Most restaurants in Brussels are not so large so these dinners
have space constraints and reservations are essential. Please subscribe
to the [Free-RTC mailing
list](http://lists.freertc.org/mailman/listinfo/discuss) for further
details about the Saturday night dinner options and how you can register
for a seat.

Related events around FOSDEM


As per usual, the [XMPP
Summit](https://wiki.xmpp.org/web/Conferences/Summit_24) is happening
ahead of FOSDEM. This time it will take place on the 30th and 31st of