Re: [OpenSIPS-Users] [OpenSIPS-Devel] [Blog] Traffic balancing – load, weights, round robin ??
Hi Robert, All the time there is space for more :). This blog post is the first from a set of docs trying to explain the routing with OpenSIPS. Including some feedback from you, I can draft as following chapters: 1) how to insert a balancer in your SIP traffic - like dialog statefull proxy, transaction statefull proxy, stateless proxy, etc 2) routing modules in opensips, like dispatcher versus drouting versus load-balancer.. In your classification, I noticed many missing features for Dynamic Routing: - in memory matching with prefix-tree - O(prefix_len) - has gui in Control Panel - destination pinging, failover, re-enable - clustering capabilities for the state of the GW/destinations - rule fallback (on matching) .. Again, any feedback is more the welcome as it will give us some ideas about the hot topics from the user perspective. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 06/15/2017 09:36 PM, Mundkowsky, Robert wrote: This is helpful. But would like a little more details there. Such as: 1) openSIPS when it actions as a Load Balancer is always a SIP proxy during the complete dialog. In other words, the caller RTP is direct connected backend gateway, but openSIPS is always between them for the SIP traffic. 2) Some information for the traffic distributions modules is stored in the database, but some info is only in memory (e.g. which gateway is enabled, count of active dialogs) This might be naïve details for most telecom folks, but helpful for naïve users like myself. I haven't played with "Carrier Route","Dispatcher", and "Dynamic Routing", but similar notes for those would be helpful too. The following video was very helpful in describing the different traffic distributions modules openSIPS supports: https://www.youtube.com/watch?v=yTLPs1-X0SM Not sure if my notes are helpful, but they are below. Routing modules: Carrier Route - old module and not maintained, but seems to work - features: routing, load balancing, blacklisting - not installed by default - no web GUI - performs longest prefix matching - you use longest preefix to pull "least cost" route out of database table - strips prefix, adds prefix/suffix - probabilities to load balancer Load Balancer - light weight - balancing based on load - can config via GUI - single database table - features: - no prefix/suffix changing - with version 1.8, you can keep counters in database - docs: Tutorial http://www.opensips.org/Documentation/Tutorials-LoadBalancing-1-9 DB schema: https://www.opensips.org/Documentation/Install-DBSchema-2-2#AEN6455 module http://www.opensips.org/html/docs/modules/2.2.x/load_balancer.html Dispatcher - computes hash over parts of request and selects an address from the destination list - can config via GUI - CallerID, From URI, To URI, Request URI - via config file or databaseas - does not strip or append prefix/suffix - out of service destinations detected by probing - can mark as out of service - can mark as in service when gateway comes back online Dynamic Routing - heavy weight - multiple criteria to determine route - prefix, caller, group, time, priority - databse only 4 tables (groups, rules, carriers, gateways) - functions - do_routing() uses all 4 tables - route_to_carrier uses carriers & gateways tables - route_to_gateway - uses gateways table Robert Mundkowsky -Original Message- From: Devel [mailto:devel-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Thursday, June 15, 2017 2:01 PM To: users@lists.opensips.org; developensips ; n...@lists.opensips.org; busin...@lists.opensips.org Subject: [OpenSIPS-Devel] [Blog] Traffic balancing – load, weights, round robin ?? Many times, during discussion about various OpenSIPS based solutions, I noticed that people make a confusion when comes to the meaning of “load balancing”. https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fblog.opensips.org%2F2017%2F06%2F15%2Ftraffic-balancing-load-weights-round-robin%2F&data=02%7C01%7Crmundkowsky%40ets.org%7Ce9218868262d4ce369ac08d4b4189373%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331465027654678&sdata=0acaS2xE6vYbvzOIyBIJC48oMhOLKF7j3YqKnDp98nI%3D&reserved=0 Enjoy the reading :) -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.opensips-solutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ce9218868262d4ce369ac08d4b4189373%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331465027664682&sdata=lUNkjOnT%2FEZ%2FR%2Fvr%2B9wyPoECMq3B%2FFwU4OjzvlwN7yY%3D&reserved=0 OpenSIPS Bootcamp 2017, Houston, US https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fopensips.org%2Ftraining%2FOpenSIPS_Bootcamp_2017.html&data=02%7C01%7Crmundkowsky%40ets.org%7Ce9218868262d4ce
Re: [OpenSIPS-Users] Can't change From header
Hi,
Changing the FROM hdr works only when proxying the request, not when it
is locally consumed (like in your case, where the SUBSCRIBE is consumed
by the presence module).
What's the purpose of changing the FROM for the watcher ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2017, Houston, US
http://opensips.org/training/OpenSIPS_Bootcamp_2017.html
On 06/13/2017 01:32 PM, maatohewetbi wrote:
I want to change from header by using *uac_replace_from*, but when I call it
from request route it's still the same.
route[handle_presence]
{
if (is_method("PUBLISH")) {
handle_publish();
} else
if (is_method("SUBSCRIBE")) {
xlog ("fu: $fu");
uac_replace_from("","sip:ro...@gotham.org");
xlog ("fu: $fu");
handle_subscribe();
}
exit;
}
--
View this message in context:
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Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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Re: [OpenSIPS-Users] B2B prepaid.xml with one MediaServer
Hi Andreas,
You don;t have to change anything in the opensips script (for adding the
pre-call media server), but only in the XML scenario.
Take a look at the marketing scenario (in the B2B tutorial), where after
a caller is connected to an announcement, upon announcement termination,
the call is sent to callee side.
Could you post your XML scenario ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2017, Houston, US
http://opensips.org/training/OpenSIPS_Bootcamp_2017.html
On 06/13/2017 01:22 PM, Andreas Bøckmann wrote:
Hello
I wanted to test the B2B script-scenario prepaid from the B2BUA tutorials.
The only change I wanted was to only invoke a MediaServer before
connecting the call to B-party.
Note my OpenSIPS is running regular proxy/registrar functions besides B2B.
The B2B is able to setup the call towards MediaServer and trigger on
receiving BYE.
The next leg towards the actual B-party is not setup as OpenSIPS
doesn't see this as B2B? but does authorization for that INVITE
request, which fails.
I believe it's due to the B2B doing somethingsomething to my URIs
and/or CSEQs.
My issue seems very much like something I found in a previous post
(https://opensips.org/pipermail/users/2010-February/010931.html) but
there was no final solution.
I trigger this for the initial INVITE as such:
if ($rU =~ "^\+1234567890.*") {
# t_newtran is necessary because b2b should not handle
retransmissions.
if (!t_newtran()) {
sl_reply_error();
exit;
}
xlog("L_WARN","B2BUA cid=$ci, Call from $fU to $rU initialized
as B2B\n");
b2b_init_request("freecall", "sip:+@1.2.3.4:5080
");
exit;
}
My B2B scenario file: https://pastebin.com/KwpgJziL
Any clues to what the solution from Anca was in the original post
referenced above or any other ideas as to what I am doing wrong here?
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Re: [OpenSIPS-Users] fetch_event_params for E_UL_CONTACT_INSERT
Hi Andreas,
Note that the number and order of parameters provided by the
E_UL_CONTACT_INSERT event changed starting with 2.3. See:
http://www.opensips.org/html/docs/modules/2.4.x/usrloc.html#idp5808768
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2017, Houston, US
http://opensips.org/training/OpenSIPS_Bootcamp_2017.html
On 06/12/2017 09:43 PM, Andreas Bøckmann wrote:
Hello!
The same snippet used below worked in older versions and I couldn't
see any change in docs.
How should I correctly fetch event parameters? Even $ci is gone inside
event_route.
_version: opensips 2.4.0-dev (x86_64/linux)
flags: STATS: On, SHM_EXTRA_STATS, DISABLE_NAGLE, USE_MCAST, SHM_MMAP,
PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: 99ebafd
main.c compiled on 12:20:24 Jun 7 2017 with gcc 4.9.2_
event_route[E_UL_CONTACT_INSERT] {
fetch_event_params("$avp(aor_param);$avp(addr_param);$avp(callid_param);$avp(rec_param);$avp(cseq_param)");
xlog("L_NOTICE","E_UL_INSERT cid=$avp(callid_param), $avp(aor_param),
$avp(rec_param), $avp(cseq_param) - callid=$ci\n");
}
DBG:core:evi_raise_event_msg: found subscriber E_UL_CONTACT_INSERT
DBG:core:parse_headers: flags=40
ERROR:core:pv_get_callid: cannot parse Call-Id header
DBG:event_route:scriptroute_fetch: Fetching parameters for event
E_UL_CONTACT_INSERT
DBG:event_route:scriptroute_fetch: Successfully fetched 5 parameters
DBG:core:parse_headers: flags=40
ERROR:core:pv_get_callid: cannot parse Call-Id header
DBG:core:subst_run: running. r=1
DBG:core:subst_str: no match
DBG:avpops:ops_subst: subst to 0 avps
DBG:core:parse_headers: flags=40
ERROR:core:pv_get_callid: cannot parse Call-Id header
E_UL_INSERT cid=sip:1.2.3.4:53875 ,
+12345...@cooldomain.aq , , -1 -
callid=
DBG:core:parse_headers: flags=40
ERROR:core:pv_get_callid: cannot parse Call-Id header
The CallID header it complains about is
Call-ID: 142559_rel51ZjRhYWM2ZGZjNjdkZTgxYWU4ZDcxZWU1NjdiNTZmN2Q
Thanks!
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Re: [OpenSIPS-Users] Building a redirect server
Hi Royee, If you want the straight answer to your question : YES, you can - take a look at the uac + uac_redirect module which allows you to interpret the 3xx reply (on the SBC side) and fork the call to the PBX layer. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 06/12/2017 12:45 PM, Royee Tichauer via Users wrote: Hi, We are thinking of going through a path of building a redirectregistrar server using opensips. In addition we are planning to add an additional SBC layer also using opensips. My question is can we use opensips in such a way that when an authenticated INVITE comes in to the SBC it is sent over to the redirect registrar server which if authorized will respond with a 3xx that will indicate the SBC that it should redirect this INVITE to the user's assigned PBX. I would of course expect the SBC to absorb the 3xx response and have this all seem less for the client. Our objective here to get the authentication server out of the call path so future SIP traffic for a dialog does not go through this server. Any thoughts/ suggestions/ past experience with this kind of setup would be helpful. Thanks a lot, Royee ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Phone Registration :(
Hi,
As Rik asked - are you sure your OpenSIPS is actually running ?
Run in a shell, on your server : "ps auxw | grep opensips"
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2017, Houston, US
http://opensips.org/training/OpenSIPS_Bootcamp_2017.html
On 06/10/2017 12:11 AM, Voice TAC wrote:
The output of opensipsctl ul show is as below:
root@OpenSIPS:~#
root@OpenSIPS:~# opensipsctl ul show
ERROR: /tmp/opensips_fifo does not exist
ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name",
"/tmp/opensips_fifo")' in your config
ERROR: and also have loaded the mi_fifo module.
root@OpenSIPS:~#
Any help please?
On Thu, Jun 8, 2017 at 3:30 PM, Alexander Jankowsky
mailto:e75a4...@exemail.com.au>> wrote:
There is also Sipgrep. I find Sipgrep is very simple to use.
Always check this when using opensipsctl ul show, to see if
anything changes,
do you always consistently get exactly the same errors each and
every time you boot.
Maybe you can get hold of some used Voip phones. That makes it a
lot more interesting.
You might say if you are still running a virtual machine or are
now on a dedicated install.
What version of Opensips. What version of Linux.
Sometimes that might be important to know.
Alex
*From:*Users [mailto:users-boun...@lists.opensips.org
] *On Behalf Of *Voice TAC
*Sent:* Thursday, 8 June 2017 7:21 AM
*To:* OpenSIPS users mailling list
*Subject:* [OpenSIPS-Users] Phone Registration :(
Hello,
I'm totally new with OpenSIPS and I manged to install it and
adjust the configuration script. However, I tried to register a
phone using X-Lite and ZOPIPER but I could not.
I tried to sniff using ngrep but I did not find any SIP messages.
I used the following command:
ngrep -pqtw byline port 5060 >sip_Messages_Debug.sip
===
In addition, I used the opensipsctl ul show command but I got:
ERROR: /tmp/opensips_fifo does not exist
ERROR: Make sure you have the line 'modparam("mi_fifo",
"fifo_name", "/tmp/opensips_fifo")' in your config
ERROR: and also have loaded the mi_fifo module.
Any help please?
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Re: [OpenSIPS-Users] regex check
Hi Lorenzo,
The error you get means that OpenSIPS does not have any IPv4 + UDP
listener defined. IF you do "netstat -ulnp| grep opensips" , what do you
get ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2017, Houston, US
http://opensips.org/training/OpenSIPS_Bootcamp_2017.html
On 06/09/2017 06:24 PM, lorenzo tombini wrote:
Hi all,
always on my centos7 64bt with opensips 2.3
i'm trying to use regex.
I've make this change to catch call from user 5001
#if ($rU=~"^\+[1-9][0-9]+$") {-> this is the default line that works.
if (pcre_match("$fU","5001")) {
$rd="10.10.10.10"; #CUSTOMIZE ME
$rp=5060;
route(relay);
exit;
}
i've loaded module regex.so
this is in the logs
Seems that after log branch, there is no way to relay the call, but I
cannot understand way
Jun 9 17:15:41 sip-proxy /usr/sbin/opensips[5836]: new branch
5001,sip:5001@10.10.10.10 ->
sip:5002@10.10.10.10:5060;user=phone,5002
Jun 9 17:15:41 sip-proxy /usr/sbin/opensips[5836]:
ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 1 (no
corresponding listening socket)
Jun 9 17:15:41 sip-proxy /usr/sbin/opensips[5836]:
ERROR:tm:t_forward_nonack: failure to add branches
Jun 9 17:15:41 sip-proxy /usr/sbin/opensips[5837]:
ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 1 (no
corresponding listening socket)
Jun 9 17:15:41 sip-proxy /usr/sbin/opensips[5837]:
ERROR:tm:t_forward_nonack: failure to add branches
could someone help me to understand this?
thanks alot in advanced
--
Lorenzo Tombini
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Re: [OpenSIPS-Users] Call Pickup feature in Opensips 1.11
Hi, yes, you are right - it is not the 100% call pickup. To do a 100% call pickup you need an B2B (and not a proxy) - as you need a B2BUA to bridge the incoming call form the original caller with the incoming call from the person doing the pickup. In the Callpick Up with Event-based Routing, to overcome the lack of a B2B capabilities, we made a small scenario change - when the person doing the call pickup calls, his call will be rejected and he will receive (as incoming call) the call he picked up. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 06/07/2017 05:06 PM, maatohewetbi wrote: Is it any way to upload whole script to test Call pickup with 2.3? I've installed this version and pasted pickup config from module event_routing documentation. Call Pickup doesn't work as it should. When I call one leg, and want to handle this call by third user I get only 480 message after dialing "33". When I delete this line it works ok but on third user phone (Grandstream) I get incomming call on second line whil first is still calling. When I answer it looks ok. But I feel it doesn't work as it should. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Call-Pickup-feature-in-Opensips-1-11-tp7607496p7607647.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to monitor WSS socket connection?
Hi Kirill, What you mean by corrupted ?? maybe terminated ? You can monitor the TCP-based connections from OpenSIPS via MI command "list_tcp_conns" http://www.opensips.org/Documentation/Interface-CoreMI-2-2#toc5 Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 06/09/2017 09:15 AM, Kirill Galinurov wrote: Hi all. Is any event raised when WSS socket corrupted ? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] acc extra_fields
Are you sure ? I have systems installed via the 2.3 and 2.2 nightly builds and the repo works great. Could you check again ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/training/OpenSIPS_Bootcamp_2017.html On 06/08/2017 01:42 PM, Edwin wrote: The 'nightly build' gives me http 500 errors, which I will look into later. $acc_extra(subscriber_id) = $avp(caller_uuid) + 0; does work, so I can continue testing. deb http://apt.opensips.org jessie 2.3-nightly apt-cache policy opensips opensips: Installed: 2.3.0-1 Candidate: 2.3.0-1 Version table: *** 2.3.0-1 0 100 /var/lib/dpkg/status 2.3.0~20170607~5ecf882-1 0 500 http://apt.opensips.org/ jessie/2.3-nightly amd64 Packages -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/acc-extra-fields-tp7607614p7607662.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS-Devel] [Blog] Traffic balancing – load, weights, round robin ??
Thanks.
FYI, the main reason we decided to use openSIPS is that is a lot more
documentation than other solutions.
Anyways, I have read lot of the books ("Building Telephony..2nd and 1.6"), some
of the website (modules, how to, ); and the major missing things are:
1 - Need a "openSIPS CookBook of recipes" that has more route examples (e.g.
how to setup user authentication, how to do HA, security (drop ghost calls,
...), setup to work with RTP proxies, ...)
2 - module documentation needs better details on how to pass in and pass out
values. For example, some modules do not expand variables, so it is really hard
to pass in values.
Robert Mundkowsky
-Original Message-
From: Devel [mailto:devel-boun...@lists.opensips.org] On Behalf Of
Bogdan-Andrei Iancu
Sent: Friday, June 16, 2017 10:47 AM
To: OpenSIPS users mailling list
Cc: busin...@lists.opensips.org; n...@lists.opensips.org; OpenSIPS devel
mailling list
Subject: Re: [OpenSIPS-Devel] [OpenSIPS-Users] [Blog] Traffic balancing – load,
weights, round robin ??
Hi Robert,
All the time there is space for more :). This blog post is the first from a set
of docs trying to explain the routing with OpenSIPS.
Including some feedback from you, I can draft as following chapters:
1) how to insert a balancer in your SIP traffic - like dialog statefull
proxy, transaction statefull proxy, stateless proxy, etc
2) routing modules in opensips, like dispatcher versus drouting versus
load-balancer..
In your classification, I noticed many missing features for Dynamic Routing:
- in memory matching with prefix-tree - O(prefix_len)
- has gui in Control Panel
- destination pinging, failover, re-enable
- clustering capabilities for the state of the GW/destinations
- rule fallback (on matching)
..
Again, any feedback is more the welcome as it will give us some ideas about the
hot topics from the user perspective.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.opensips-solutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ccb9609d3270b4091081108d4b4c6c079%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636332213102517343&sdata=ejvSuDmud%2FPcsYLuGhCjfEX3Zug1WSCwBmWblGUTcus%3D&reserved=0
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On 06/15/2017 09:36 PM, Mundkowsky, Robert wrote:
> This is helpful. But would like a little more details there. Such as:
>
> 1) openSIPS when it actions as a Load Balancer is always a SIP proxy during
> the complete dialog. In other words, the caller RTP is direct connected
> backend gateway, but openSIPS is always between them for the SIP traffic.
> 2) Some information for the traffic distributions modules is stored in
> the database, but some info is only in memory (e.g. which gateway is
> enabled, count of active dialogs)
>
> This might be naïve details for most telecom folks, but helpful for naïve
> users like myself.
>
> I haven't played with "Carrier Route","Dispatcher", and "Dynamic Routing",
> but similar notes for those would be helpful too.
>
> The following video was very helpful in describing the different traffic
> distributions modules openSIPS supports:
>
> https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.y
> outube.com%2Fwatch%3Fv%3DyTLPs1-X0SM&data=02%7C01%7Crmundkowsky%40ets.
> org%7Ccb9609d3270b4091081108d4b4c6c079%7C0ba6e9b760b34fae92f37e6ddd9e9
> b65%7C0%7C0%7C636332213102517343&sdata=mDlSlzbtLQ%2FL3uDq67VkOa%2B8FPG
> 20nNx4AZ9duHVi1s%3D&reserved=0
>
> Not sure if my notes are helpful, but they are below.
>
> Routing modules:
> Carrier Route
> - old module and not maintained, but seems to work
> - features: routing, load balancing, blacklisting
> - not installed by default
> - no web GUI
>
> - performs longest prefix matching
> - you use longest preefix to pull "least cost" route out of database
> table
> - strips prefix, adds prefix/suffix
> - probabilities to load balancer
>
>
> Load Balancer
> - light weight
> - balancing based on load
> - can config via GUI
> - single database table
>
> - features:
> - no prefix/suffix changing
> - with version 1.8, you can keep counters in database
>
> - docs:
> Tutorial
> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.op
> ensips.org%2FDocumentation%2FTutorials-LoadBalancing-1-9&data=02%7C01%
> 7Crmundkowsky%40ets.org%7Ccb9609d3270b4091081108d4b4c6c079%7C0ba6e9b76
> 0b34fae92f37e6ddd9e9b65%7C0%7C0%7C636332213102517343&sdata=TpUc3EXNX1N
> 0lGf4UCuQsHBceQQYu%2FkdtGxOpYWe8MM%3D&reserved=0
> DB schema:
> https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.o
>
Re: [OpenSIPS-Users] Call Pickup feature in Opensips 1.11
Hi, I know two ways to do Call Pickup using OpenSIPS, The first one is almost effortless, use a phone capable to do it. Some phones support the RFC5359 https://datatracker.ietf.org/doc/rfc5359/ (Page 147). As far as I know Yealink and Polycom support it. You have to configure the phone to Notify the in session dialogs (dialog-info-call-pickup). I use this with Yealinks in my office and it works just fine. I didn't tried the buddy list using RLS, I will make a try in the future. The second one is to save in cache the dialog information regarding in progress calls. When picking up a call, build the replaces header using this information and send to the device, it will replace the call. You have to build your own script. Good luck. Flavio E. Goncalves 2017-06-16 12:19 GMT-03:00 Bogdan-Andrei Iancu : > Hi, > > yes, you are right - it is not the 100% call pickup. To do a 100% call > pickup you need an B2B (and not a proxy) - as you need a B2BUA to bridge > the incoming call form the original caller with the incoming call from the > person doing the pickup. > > In the Callpick Up with Event-based Routing, to overcome the lack of a B2B > capabilities, we made a small scenario change - when the person doing the > call pickup calls, his call will be rejected and he will receive (as > incoming call) the call he picked up. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Bootcamp 2017, Houston, US > http://opensips.org/training/OpenSIPS_Bootcamp_2017.html > > > On 06/07/2017 05:06 PM, maatohewetbi wrote: > >> Is it any way to upload whole script to test Call pickup with 2.3? I've >> installed this version and pasted pickup config from module event_routing >> documentation. Call Pickup doesn't work as it should. When I call one leg, >> and want to handle this call by third user I get only 480 message after >> dialing "33". When I delete this line it works ok but on third user phone >> (Grandstream) I get incomming call on second line whil first is still >> calling. When I answer it looks ok. But I feel it doesn't work as it >> should. >> >> >> >> -- >> View this message in context: http://opensips-open-sip-serve >> r.1449251.n2.nabble.com/Call-Pickup-feature-in-Opensips-1- >> 11-tp7607496p7607647.html >> Sent from the OpenSIPS - Users mailing list archive at Nabble.com. >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS Online Training
Hello, I'm enrolled in OpenSIPS online training and I'm not able to access the site. OpenSIPS.org>>training>> openSIPS QuickStart official portal With whom I should communicate? Sent from my iPhone ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
