[SR-Users] WebRTC-UDP voicemail and call transfer

[Social Boh via sr-users]

Hello list,

I'm trying implementing a solution like:

https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg

The 2 problems I have a this moment are:

Call from UDP phone to WebRTC phone, the WebRTC phone reject the call; 
the call go to voicemail but in the SDP still have WebRTC SDP


During a SIP UDP - WebRTC Call tranfer the call using phone button... 
the transfer don't work.


Any hint

Regards

--
---
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[SR-Users] Re: Kamailio, Asterisk and Parked calls

[Social Boh via sr-users]

Thank's all for your replies.

The problem is I can't find a pattern to do this. The Asterisk 
configuration is multitenant using chan_sip Channel. The customers 
parking the calls using the transfer sipphone button and pick up using 
BLF or calling the parking slot.


Each tenant have first 5 digits identifies him and then 3 digits more 
identifies each endpoint.


I tried using hash from URI and other configurations without success.

Regards

---
I'm SoCIaL, MayBe

El 25/04/2024 a las 3:44 p. m., Barry Flanagan escribió:

On Thu, 25 Apr 2024, at 21:17, Social Boh via sr-users wrote:

Hello list,

I'm using Kamailio to balance SIP request between two Asterisk. I have a
problem with Parking calls using Asterisk Application. When a user
parking a call, some times can't pick up parked call because the call to
pick up the Call go to the other Asterisk.

I'm using the 1 algorithm "hash over from URI" without success.


This is an Asterisk limitation/frustration really. Queues, Parking and Conferences need 
to run on the same server instance. So for a particular "tenant", for it to 
work you need to ensure that all parking for this tenant is directed to the same server. 
This also applies to any conference or queue.

If your individual endpoints can be recognised as being the same "tenant", try 
hashing over that tenant string and they will all get directed  to the same server.

If the above is not possible, but maybe the  number being dialled is 1:1 with 
the server (parking lot in your example) then you could hash over the $tU and 
achieve the same result.

In a nutshell, you need to find some common way that you can identify calls 
that should be grouped together, and then hash over that.

Hope this helps.

-Barry


Any hint?

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[SR-Users] Kamailio, Asterisk and Parked calls

[Social Boh via sr-users]

Hello list,

I'm using Kamailio to balance SIP request between two Asterisk. I have a 
problem with Parking calls using Asterisk Application. When a user 
parking a call, some times can't pick up parked call because the call to 
pick up the Call go to the other Asterisk.


I'm using the 1 algorithm "hash over from URI" without success.

Any hint?

--
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I'm SoCIaL, MayBe

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[SR-Users] Re: TLS module doesn't support TLSV1

[Social Boh via sr-users]

SIPTRACE module and SNGREP for TLS Capture:

https://www.voztovoice.org/?q=node/3020

Spanish

---
I'm SoCIaL, MayBe

El 4/04/2024 a las 3:12 a. m., Omar Atef via sr-users escribió:

Well, I've tried all versions lower than "TLSv1.2" and it didn't work just 
Kamailio running fine without activating TLS.

Also I wonder if there's a way to bug TLS SIP messages throw Kamailio itself because as 
you know it doesn't appear in "sngrep".

Thanks,
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[SR-Users] Re: Can we discuss the internals of Kamailio's routing logic and how to customize it for specific needs?

[Social Boh via sr-users]

can you sto to spam on many mailing lists using excuse like this?

thank you

---
I'm SoCIaL, MayBe

El 28/03/2024 a las 1:24 a. m., Inextrix Technologies Pvt Ltd via 
sr-users escribió:

This question digs into the heart of Kamailio's functionality. It asks about:

Routing Logic: How Kamailio decides where to send SIP messages (calls, 
registrations, etc.) based on various factors like caller ID, destination 
number, or custom rules.
Customization: How you can modify this built-in logic to fit your specific 
needs. This could involve using scripting languages, manipulating SIP messages, 
or defining custom routing rules.
The discussion would be for developers who want to unlock Kamailio's full 
potential and tailor it for their unique VoIP deployments.
Visit us for More Info : https://inextrix.com/services/kamailio-development/
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[SR-Users] Re: RTP Engine with Kamailio

[Social Boh via sr-users]

Take a look at documentation module:

https://www.kamailio.org/docs/modules/5.8.x/modules/rtpengine.html#rtpengine.p.rtpengine_sock

you can define more RTPEngine Servers with or without weight for balancing

You can use rtpengine table too:

https://www.kamailio.org/docs/db-tables/kamailio-db-5.8.x.html#idm7211

I implemented Kamailio with more than 20 RTPENgine servers and works 
very well.


Regards

---
I'm SoCIaL, MayBe

El 22/03/2024 a las 1:16 p. m., Vishal Pai via sr-users escribió:

Hello Everyone.

I am looking for documentation to have multiple RTP Engine servers 
connected with kamailio using db url and balance the load accordingly.



Thanks

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[SR-Users] Re: best approach to setup SIP Session sharing between servers

[Social Boh via sr-users]

Better use corosync and pacemaker to keeep call audio flow during moving 
virtual IP using master-slave redis


---
I'm SoCIaL, MayBe

El 27/02/2024 a las 9:07 a. m., Sergio Charrua via sr-users escribió:

Thanks Alex & Olle!
I've done some tests with Keepalived for Virtual IP failover, and I 
found that it works more or less as you mentioned, Olle: if an Invite 
comes from Server #1, and that server gets offline, the BYE will 
indeed be correctly processed by Server #2.


Isn't DMQ supposed to handle this? as i understand, DMQ sends the SIP 
messages to other servers, so wouldn't it be enough? (I assume, I 
haven't tested DMQ yet )


*
*

*Sérgio Charrua*

*www.voip.pt *
Tel.: +351 91 631 11 44

Email : *sergio.char...@voip.pt*

This message and any files or documents attached are strictly 
confidential or otherwise legally protected.


It is intended only for the individual or entity named. If you are not 
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On Tue, Feb 27, 2024 at 2:20 PM Olle E. Johansson via sr-users 
 wrote:


In theory if you set the record-route to a DNS name and not an Ip
address, then endpoints should be able to failover within a dialog
- but as Alex says - transactions may fail. In many cases there
are retry timers to restart transactions that fail.

This means that if a call starts with an INVITE transaction on one
server, it should in theory be able to find another server for the
BYE. Whether this DNS srv based fail over is implemented in
end-points is up to testing to prove. If not, then virtual IP
failover is your best friend.

/O

> On 27 Feb 2024, at 13:33, Alex Balashov via sr-users
 wrote:
>
> That would require transaction replication, which Kamailio
doesn't have.
>
> Most messages can be processed statelessly, so this isn't a huge
obstacle. However, CANCELs won't work.
>
>> On 27 Feb 2024, at 06:49, Sergio Charrua via sr-users
 wrote:
>>
>> Hi all!
>>
>> I am having some difficulties/doubts on what would be the best
approach to let multiple Kamailio servers share their SIP sessions
between each other.
>> The goal is to have HA on Kamailio cluster such that if a call
is received and initiated on Kamailio #1, and during any moment of
the call (before or while established) if the server #1 goes down
or Kamailio stops working for any reason, the call can be
processed by Kamailio server #n available in the cluster.
>> I do not want to mess with Virtual IPs or DNS, that is not the
point, but instead need to get a proper way to share SIP sessions
between Kamailio servers such that any server could continue the
call without issue.
>>
>> I have checked the Dialog and DMQ module, but I am not sure if
that is the way to go
>>
>> Could any one share a solution for this?
>>
>> Greatly appreciated.
>>
>> Cheers,
>>
>>
>> Sérgio Charrua
>> www.voip.pt 
>>
>>
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>
> --
> Alex Balashov
> Principal Consultant
> Evariste Systems LLC
> Web: https://evaristesys.com
> Tel: +1-706-510-6800
>
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[SR-Users] Re: Freeswitch and Kamailio Integration.

[Social Boh via sr-users]

Hello,

what type of integration do you need?

Only media services like voicemail or?

Regards

---
I'm SoCIaL, MayBe

El 21/02/2024 a las 6:26 a. m., jayapal--- via sr-users escribió:

Greetings,

I am new to Kamailio and seeking assistance. I have installed FreeSWITCH 
version 1.10.11 and Kamailio version 5.6.5. I have two inquiries:

I aim to configure FreeSWITCH with Kamailio. Could someone kindly guide me on 
achieving this integration?

I am curious about the storage location of Kamailio's log files. Could someone 
please provide information regarding this?

Thank you in advance for your help and guidance.
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[SR-Users] Re: PRACK RFC3262

[Social Boh via sr-users]

So Kamailio can't generate PRACK packets, only relay

Thank you

Regards

---
I'm SoCIaL, MayBe

El 9/02/2024 a las 3:54 a. m., Ihor Olkhovskyi via sr-users escribió:

Kamailio is a SIP proxy, it's not generating packets.
So,correct answer would be yes, Kamailio supports relay of the 
PRACK-related packets fully


Le jeu. 8 févr. 2024 à 22:16, Social Boh via sr-users 
 a écrit :


Does Kamailio support the SIP PRACK method? If yes, How?

Regards

-- 
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--
Best regards,
Ihor (Igor)

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[SR-Users] PRACK RFC3262

[Social Boh via sr-users]

Does Kamailio support the SIP PRACK method? If yes, How?

Regards

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[SR-Users] Re: LCR logic

[Social Boh via sr-users]

Thank you for your reply.

Regards

---
I'm SoCIaL, MayBe

El 26/01/2024 a las 6:21 p. m., Juha Heinanen escribió:

Social Boh via sr-users writes:


so I have 3 lcr_rule_target use the same
lcr_rule entry but 3 different Gateways and priority. So the order of
rules would be longest Request-URI (same) and the priority but the
result is:

added matched_gws[0]=[3, 5, 200, 2193562]
added matched_gws[1]=[1, 5, 2, 6888519]
added matched_gws[2]=[2, 5, 50, 2159388]

same prefix but first rule target with priority 200 and then 2 and then
50. I think the result would be priority 2,50 and 100.

¿Or not?

matched_gws are sorted after those debug messages.

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[SR-Users] LCR logic

[Social Boh via sr-users]

Hello,

LCR module logic is :

When the function /load_gws()/ is called, matching gateways (that are 
not currently designated as defunct) are ordered for forwarding purposes 
as follows:


1.

   according to longest Request-URI user part match

2.

   according to tuple's priority

3.

   according to tuple's randomized weight

or, if priority_ordering parameter is set to value 1, as follows:

1.

   according to tuple's priority

2.

   according to tuple's randomized weight

and Smaller priority value means higher priority (highest priority value 
being 0 and lowest being 255). so I have 3 lcr_rule_target use the same 
lcr_rule entry but 3 different Gateways and priority. So the order of 
rules would be longest Request-URI (same) and the priority but the 
result is:


added matched_gws[0]=[3, 5, 200, 2193562]
added matched_gws[1]=[1, 5, 2, 6888519]
added matched_gws[2]=[2, 5, 50, 2159388]

same prefix but first rule target with priority 200 and then 2 and then 
50. I think the result would be priority 2,50 and 100.


¿Or not?

---
I'm SoCIaL, MayBe
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[SR-Users] Change in LCR Debug

[Social Boh via sr-users]

Hello,

using DEBUG with LCR module to see how the module apply the rules, I see 
he use Gateway Index in the list. The Gateway Index could be different 
to Gateway ID and maybe can confuse the correct interpretation about the 
use of the rules.


Is it possible change the code:

matched_gws[gw_index].gw_index = t->gw_index;
    matched_gws[gw_index].rule_id = 
rule->rule_id;
    matched_gws[gw_index].prefix_len = 
pl->prefix_len;
    matched_gws[gw_index].priority = 
t->priority;
    matched_gws[gw_index].weight = 
t->weight * (kam_rand() >> 8);

    matched_gws[gw_index].duplicate = 0;
    LM_DBG("added matched_gws[%d]=[%u %u, 
%u, %u, %u]\n", gw_index,
    t->gw_index, 
pl->prefix_len, t->priority,

matched_gws[gw_index].weight);

to use gateway ID and not gateway Index?

Thank you

Regards

--
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[SR-Users] Re: webRTC Test Client?

[Social Boh via sr-users]

Hello,

I'm using this one:

https://github.com/InnovateAsterisk/Browser-Phone

Regards

---
I'm SoCIaL, MayBe

El 2/01/2024 a las 10:46 a. m., Andy Newlands via sr-users escribió:

Hi,

I have a Freeswitch installation, fronted by a Kamailio proxy

I would like to configure Kamailio to bridge SIP between webRTC 
clients and Freeswitch.


The first issue I hit is how to set up a test webRTC client to make 
calls into Kamailio/FS.


Can anyone point me to a simple webRTC client I can use to make 
test webRTC/SIP calls into my setup?  Thanks


BTW: I'm not JS experienced (I work with Python, C#, C/C++).

Kind regards,

Andy

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[SR-Users] Re: refer-to to new INVITE

[Social Boh via sr-users]

Hello,

I don't think dlg_bridge apply in this case.

I receive the REFER from remote PBX/SWITCH and I don't need to create a 
NEW call between two endpoints.


dlg_bridge, bridge a INVITE between two endpoints using REFER, Kamailio 
call first endpoint and when the endpoint answer send him a REFER with 
the callee data.


Right?

---
I'm SoCIaL, MayBe

El 25/11/2023 a las 12:09 p. m., David Villasmil via sr-users escribió:

https://sr-users.sip-router.narkive.com/1lg6Ov5U/problem-initiating-a-call-with-dlg-bridge


On Sat, 25 Nov 2023 at 06:39, Alex Balashov via sr-users 
 wrote:


Hi,

You may wish to explore this function, and its RPC correlate:

dialog Module

<https://kamailio.org/docs/modules/5.7.x/modules/dialog.html#dialog.f.dlg_bridge>
kamailio.org

<https://kamailio.org/docs/modules/5.7.x/modules/dialog.html#dialog.f.dlg_bridge>
favicon.ico

<https://kamailio.org/docs/modules/5.7.x/modules/dialog.html#dialog.f.dlg_bridge>



<https://kamailio.org/docs/modules/5.7.x/modules/dialog.html#dialog.f.dlg_bridge>


-- Alex

—
Sent from mobile, apologies for brevity and errors.


    On Nov 22, 2023, at 7:17 PM, Social Boh via sr-users
 wrote:

Hello,


I'd like to know if there is a way, from Kamailio side, take the
refer-to header Kamailio receive in a REFER request and create a
new INVITE whit this data; then bridge the call between the
originating user and the new destination user.

Thank you

Regards

-- 
---

I'm SoCIaL, MayBe

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[SR-Users] Re: WEBRTC-to-SIP & Vice Versa

[Social Boh via sr-users]

Hello,

wich kind od issue do you have?

Regards

---
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El 25/11/2023 a las 12:17 a. m., Vishal Pai via sr-users escribió:

Thanks Yuriy.

Basically here we need to use the path, nathelper module and when we 
receive the invite for the call we need to route it accordingly to the 
destination address.


Note - Here I am facing an issue in call specially from Asterisk PBX 
(UDP) - Kamailio (WSS).


Any reference would be highly appreciated.

On Thu, Nov 2, 2023 at 3:39 AM Yuriy G  wrote:

Most common rules are: Use Path
https://datatracker.ietf.org/doc/html/rfc3327
As well as use add_contact_alias/handle_ruri_alias from the
nathelper module or same pattern ( like add_path_receoved ) to not
destroy original Contact information, and, at the same time to
provide destination address where to send INVITE to...

On Wed, 1 Nov 2023, 20:28 Vishal Pai via sr-users,
 wrote:

Hello All

Good Day

Here I am trying to achieve something like this

WSS to UDP Ext registartion which works well using
https://github.com/havfo/WEBRTC-to-SIP/tree/master/etc
Note here i am forwarding the registartion to asterisk pbx
server and define WSS of kamailio

Now for calls

- Webrtc - SIP calls works

- SIP - Webrtc calls not working
Here I am getting the request of the call on kamailio but
kamailio send 404 not found to Asterisk PBX

Webrtc - Webrtc calls not working

Can anyone guide me on what I am doing wrong?

Thanks

Vishal P
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[SR-Users] Re: Online devel meeting on matrix - 14:30 UTC, Dec 5, 2023

[Social Boh via sr-users]

Maybe a topic would be add a module to process REFER SIP method at 
Kamailio Level like B2B opensips modules.


I have more than one customer Ask me to implement this type of solution..

Regards

---
I'm SoCIaL, MayBe

El 24/11/2023 a las 5:27 a. m., Daniel-Constantin Mierla via sr-users 
escribió:

Hello,

we should consider an online devel meeting sometime soon to summarize
what was done at (and still needs to be done after) devel meeting in
Dusseldorf and plan a bit the targets for next major release (5.8 or 6.0)?

If considered useful, I propose Dec 5 at 14:30UTC (15:30
Berlin/Paris/Madrid/Rome), but we can also look for other dates as well.

Topics to be discussed can be added at:

   -
https://github.com/kamailio/kamailio-wiki/blob/main/docs/devel/irc-meetings/2023a.md

Pull requests can be made by users without git access.

Cheers,
Daniel


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[SR-Users] refer-to to new INVITE

[Social Boh via sr-users]

Hello,

I'd like to know if there is a way, from Kamailio side, take the 
refer-to header Kamailio receive in a REFER request and create a new 
INVITE whit this data; then bridge the call between the originating user 
and the new destination user.


Thank you

Regards

--
---
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[SR-Users] Re: sips to sip

[Social Boh via sr-users]

Hello DAvid,

can you attach a pcap file about one example call?

Regards

---
I'm SoCIaL, MayBe

El 20/11/2023 a las 1:44 p. m., David Villasmil via sr-users escribió:

Hello guys,

I have this setup where one side is TLS and the other UDP. Normally 
this works fine, but we have this provider sending sips as the schema 
everywhere (from, to, rr, contacts), kamailio sends the same sips to 
the upstream usp freeswitch.


My problem is when FS sends back a 200OK and kamailio forwards it back 
to the provider, the provider sends an ACK and kamailio can't match it 
with the dialog and doesn't know where to forward it.


i think this is happening because FS when is sees SIPS is setting the 
contact port as 5081 instead of the usual 5080...



Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


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[SR-Users] Re: Obsoleting unmaintained modules

[Social Boh via sr-users]

I use uri_db module and the does_uri_exist() function to know if a user 
exist or not.


I don't know if there is other function I can use with the same task

Regards

---
I'm SoCIaL, MayBe

El 14/11/2023 a las 7:30 a. m., Daniel-Constantin Mierla via sr-users 
escribió:

Hello,

during the Kamailio Developers Meeting 2023 in Dusseldorf that took
place last week, it was proposed to obsolete modules that seem to be
unmaintained and no activity about them was noticed during the past
years. It is quite some overhead in packaging them and trying to keep
them compiling when they have external dependencies, therefore such step
should spare some resources in the future.

The list (see below) was built based on the options of those present at
the meeting, now we want to discuss it on the larger communities of
developers and users. If you are using any of these modules or you think
any of them worth keeping, reply with the names of the modules that you
want to be kept.

The proposed action is to relocate the obsoleted modules to a new git
repository "kamailio-obsolete" to still keep some visibility to them and
in the eventually of future interest on any of them, it can be
reintroduced in the main repository.

  Next is the initial list of modules proposed to be considered obsolete:

- app_java
- app_lua_sr
- app_mono
- app_python
- app_sqlang
- auth_identity
- call_control
- db2_ldap
- db2_ops
- db_cassandra
- db_perldvdb
- dnssec
- domainpolicy
- h350
- mediaproxy
- osp
- peering
- print
- print_lib
- pua_xmpp
- ratelimit
- uid_auth_db
- uid_avp_db
- uid_domain
- uid_gflags
- uid_uri_db
- uri_db
- xmpp
- xprint

Cheers,
Daniel


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[SR-Users] Re: UDP Fragmentation webRTC/UDP Call

[Social Boh via sr-users]

hello,

i only a example... not for production.

Regards

---
I'm SoCIaL, MayBe

El 16/10/2023 a las 10:37 a. m., Mirko Brankovic via sr-users escribió:
I'm maybe out of context here but i don't think you can remove ssrc 
headers from sdp since they are crucial in identification of the rtp 
streams


On Mon, Oct 16, 2023, 14:41 Social Boh via sr-users 
 wrote:


Hello,

in the rtpengine LOG i can see:

Oct 16 06:32:53 sip1 rtpengine[146001]: DEBUG:
[28vs0pbtdtu7udefti8m]: ... "*: [ "sdp-attr-remove-audio-ssrc::"
], "replace":* [ "origin", "session-connection" ],
"transport-protocol": "RTP/AVP", "rtcp-mux": [ "demux" ], "SDES":
[ "off" ], "call-id": "28vs0pbtdtu7udefti8m", "via-branch":
"z9hG4bK4978560", "received-from": [ "IP4", "186.98.231.105" ],
"from-tag": "h8l2da68sp", "command": "offer" }

but the ssrc lines still presents in the INVITE sent to sip device.

Regards

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 12:39 p. m., Social Boh via sr-users escribió:

Hello,

I'm trying to using sdp-attr without luck:

sdp-attr-remove-audio-ssrc:

in this sentence:

$xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + "
rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP
sdp-attr-remove-audio-ssrc:";

this to remove two ssrc lines

Regards

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 11:12 a. m., Richard Fuchs via sr-users
escribió:

On 14/10/2023 02.46, [EXT] Karsten Horsmann via sr-users wrote:

Hi,

did you pass over the ice stuff from webrtc to the udp side?
You could strip that of with rtpengine options.


With recent versions we even have explicit SDP manipulations
options, so you can use rtpengine to strip attributes that it
itself doesn't understand or process.

Cheers

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[SR-Users] Re: UDP Fragmentation webRTC/UDP Call

[Social Boh via sr-users]

Hello,

in the rtpengine LOG i can see:

Oct 16 06:32:53 sip1 rtpengine[146001]: DEBUG: [28vs0pbtdtu7udefti8m]: 
... "*: [ "sdp-attr-remove-audio-ssrc::" ], "replace":* [ "origin", 
"session-connection" ], "transport-protocol": "RTP/AVP", "rtcp-mux": [ 
"demux" ], "SDES": [ "off" ], "call-id": "28vs0pbtdtu7udefti8m", 
"via-branch": "z9hG4bK4978560", "received-from": [ "IP4", 
"186.98.231.105" ], "from-tag": "h8l2da68sp", "command": "offer" }


but the ssrc lines still presents in the INVITE sent to sip device.

Regards

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 12:39 p. m., Social Boh via sr-users escribió:

Hello,

I'm trying to using sdp-attr without luck:

sdp-attr-remove-audio-ssrc:

in this sentence:

$xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux 
DTLS=off SDES-off ICE=remove RTP/AVP sdp-attr-remove-audio-ssrc:";


this to remove two ssrc lines

Regards

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 11:12 a. m., Richard Fuchs via sr-users escribió:

On 14/10/2023 02.46, [EXT] Karsten Horsmann via sr-users wrote:

Hi,

did you pass over the ice stuff from webrtc to the udp side? You 
could strip that of with rtpengine options.


With recent versions we even have explicit SDP manipulations options, 
so you can use rtpengine to strip attributes that it itself doesn't 
understand or process.


Cheers

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[SR-Users] Re: UDP Fragmentation webRTC/UDP Call

[Social Boh via sr-users]

Hello,

I'm trying to using sdp-attr without luck:

sdp-attr-remove-audio-ssrc:

in this sentence:

$xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux 
DTLS=off SDES-off ICE=remove RTP/AVP sdp-attr-remove-audio-ssrc:";


this to remove two ssrc lines

Regards

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 11:12 a. m., Richard Fuchs via sr-users escribió:

On 14/10/2023 02.46, [EXT] Karsten Horsmann via sr-users wrote:

Hi,

did you pass over the ice stuff from webrtc to the udp side? You 
could strip that of with rtpengine options.


With recent versions we even have explicit SDP manipulations options, 
so you can use rtpengine to strip attributes that it itself doesn't 
understand or process.


Cheers

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[SR-Users] Re: UDP Fragmentation webRTC/UDP Call

[Social Boh via sr-users]

Thank you to answer.

this is my SDP annex (no ice o related stuff):


Content-Length: 1031


/v=0/

/o=- 326299663336889577 2 IN IP4 XXX.XXX.XXX.XXX/

/s=-/

/t=0 0/

/a=extmap-allow-mixed/

/a=msid-semantic: WMS 078639f4-3b58-4a29-90ef-62d05ccf01a2/

/m=audio 21766 RTP/AVP 111 63 9 0 8 13 110 126/

/c=IN IP4 //XXX.XXX.XXX.XXX/

/a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level/

/a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time/

/a=extmap:3 
http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01/


/a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid/

/a=msid:078639f4-3b58-4a29-90ef-62d05ccf01a2 
4cacf939-a056-4a1a-a0c5-748aaf5e3d43/


/a=ssrc:805220451 cname:wrpkmkhX7zbjXfUu/

/a=ssrc:805220451 msid:078639f4-3b58-4a29-90ef-62d05ccf01a2 
4cacf939-a056-4a1a-a0c5-748aaf5e3d43/


/a=mid:0/

/a=rtpmap:111 opus/48000/2/

/a=fmtp:111 useinbandfec=1; minptime=10/

/a=rtcp-fb:111 transport-cc/

/a=rtpmap:63 red/48000/2/

/a=fmtp:63 111/111/

/a=rtpmap:9 G722/8000/

/a=rtpmap:0 PCMU/8000/

/a=rtpmap:8 PCMA/8000/

/a=rtpmap:13 CN/8000/

/a=rtpmap:110 telephone-event/48000/

/a=rtpmap:126 telephone-event/8000/

/a=sendrecv/

/a=rtcp:21767/

I think I can eliminate  a=extmap lines and maybe other lines but i 
don't really know where I have to put *sdp_remove_line_by_prefix* 
function//in the script/

/

Regards/
/

---
I'm SoCIaL, MayBe

El 14/10/2023 a las 1:46 a. m., Karsten Horsmann escribió:

Hi,

did you pass over the ice stuff from webrtc to the udp side? You could 
strip that of with rtpengine options.


Social Boh via sr-users  schrieb am Sa., 
14. Okt. 2023, 01:00:


Hello,

I'm trying to solve a problem with calls WebRTC/UDP. I think is a
fragmentation problem because udp/webRTC calls works without
problem. The INVITE from Kamailio to the UDP device not receive
any type of answer.

I'm trying to reduce the SDP annex size using:


  |sdp_remove_line_by_prefix|

for some lines unsuccessfully. Maybe I don't know where put this
function in the script.

Or, is there another way to solve this issue?

Thank you in advance

Regards


-- 
---

I'm SoCIaL, MayBe

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[SR-Users] UDP Fragmentation webRTC/UDP Call

[Social Boh via sr-users]

Hello,

I'm trying to solve a problem with calls WebRTC/UDP. I think is a 
fragmentation problem because udp/webRTC calls works without problem. 
The INVITE from Kamailio to the UDP device not receive any type of answer.


I'm trying to reduce the SDP annex size using:


 |sdp_remove_line_by_prefix|

for some lines unsuccessfully. Maybe I don't know where put this 
function in the script.


Or, is there another way to solve this issue?

Thank you in advance

Regards


--
---
I'm SoCIaL, MayBe
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[SR-Users] MS Teams and ca_list.pem

[Social Boh]

Hello,

I'd like to know where i can download/search ca_list.pem file to use in 
tls.cfg configuration file for MS TEAMS.


Without the file:

ERROR: tls [tls_util.h:50]: tls_err_ret(): TLS write:error:0A86:SSL 
routines::certificate verify failed (sni: unknown)


 ERROR: tls [tls_server.c:1336]: tls_h_read_f(): src addr: 52.114.76.76:506

ERROR: tls [tls_server.c:1339]: tls_h_read_f(): dst addr: KamailioIP:0

ERROR:  [core/tcp_read.c:1476]: tcp_read_req(): ERROR: 
tcp_read_req: error reading - c: 0x7fb6a23ba4f8 r: 0x7fb6a23ba620 (-1)


Thank you

Regards


--
---
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[SR-Users] Re: Kamailio v5.7.0 Released - new major version is out

[Social Boh]

Hi Kamailians,

maybe would be useful known how compile TLS_WOLFSSL module in Rocky 
Linux 9 from sources (Spanish Article):


https://www.voztovoice.org/?q=node/3025

Regards

---
I'm SoCIaL, MayBe

El 17/05/2023 a las 9:08 a. m., Daniel-Constantin Mierla escribió:

Kamailio v5.7.0 is out – it comes with 4 new modules and a large
set of improvements touching again more than 60 existing modules.

You can read a bit more detailed release notes at:

    * https://www.kamailio.org/w/kamailio-v5-7-0-release-notes/

Many thanks to all developers and community members that made possible
this release.

v5.7.0 brings more flexibility and optimizations across many components,
new variables and transformations, expression-based preprocessor values
and conditions,
alternative TLS module using WolfSSL project, improvements to
STIR/SHAKEN, IMS/VoLTE,
more functions exported to KEMI interface, including a new KEMI Python3
interpreter.
... just to list only a few here. Enjoy Kamailio v5.7.0!

Thank you for flying Kamailio!
Daniel


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[SR-Users] Re: WebRTC Error

[Social Boh]

SOLVED

---
I'm SoCIaL, MayBe

El 9/03/2023 a las 8:11 a. m., Social Boh escribió:

Hello,

I have this error when trying to register this softphone: 
https://github.com/InnovateAsterisk/Browser-Phone


61096#015#012Host: sip1.kamailio.xyz:8443#015#012Connection: 
Upgrade#015#012Pragma: no-cache#015#012Cache-Control: 
no-cache#015#012Upgrade: websocket#015#012Origin: 
https://campus.voztovoice.org#015#012Sec-WebSocket-Version: 
13#015#012User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64) 
AppleWebKit/537.36 (KHTML, like Gecko) Chrome/110.0.0.0 
Safari/537.36#015#012Accept-Encoding: gzip, deflate, 
br#015#012Accept-Language: es-419,es;q=0.6#015#012sec-gpc: 
1#015#012Sec-WebSocket-Key: 
uj5cXle8zzBFUzv4kWB2pg==#015#012Sec-WebSocket-Extensions: 
permessage-deflate; 
client_max_window_bits#015#012Sec-WebSocket-Protocol: 
sip#015#012#015#012>
Mar  9 08:10:55 sip1 /sbin/kamailio[194807]: WARNING: 

[SR-Users] WebRTC Error

[Social Boh]

Hello,

I have this error when trying to register this softphone: 
https://github.com/InnovateAsterisk/Browser-Phone


61096#015#012Host: sip1.kamailio.xyz:8443#015#012Connection: 
Upgrade#015#012Pragma: no-cache#015#012Cache-Control: 
no-cache#015#012Upgrade: websocket#015#012Origin: 
https://campus.voztovoice.org#015#012Sec-WebSocket-Version: 
13#015#012User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64) 
AppleWebKit/537.36 (KHTML, like Gecko) Chrome/110.0.0.0 
Safari/537.36#015#012Accept-Encoding: gzip, deflate, 
br#015#012Accept-Language: es-419,es;q=0.6#015#012sec-gpc: 
1#015#012Sec-WebSocket-Key: 
uj5cXle8zzBFUzv4kWB2pg==#015#012Sec-WebSocket-Extensions: 
permessage-deflate; 
client_max_window_bits#015#012Sec-WebSocket-Protocol: sip#015#012#015#012>
Mar  9 08:10:55 sip1 /sbin/kamailio[194807]: WARNING: 

[SR-Users] TLS -> UDP ReInvite

[Social Boh]

hello,

I'm configuring Kamailio to permite calls like:

UDP -> UDP

TLS -> TLS

UDP -> TLS

TLS -> UDP

Until now all is ok; the only problem is with Microsip SoftPhone. When 
make a call from Microsip TLS to another phone UDP, when the callee 
answer, Microsip send a new INVITE and Kamailio don't delete all 
references to SRTP so the call happens but without audio.


I know I have to process the reINVITE where I do has_totag stuff but I 
don't know how.


Regards

--
---
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Re: [SR-Users] Kamailio + RTPEngine SIP Proxy...

[Social Boh]

Maybe:

if (loose_route()) {
    route(DLGURI);
    if (is_method("BYE")) {
    setflag(FLT_ACC); # do accounting ...
    setflag(FLT_ACCFAILED); # ... even if the 
transaction fails

    } else if ( is_method("ACK") ) {
    # ACK is forwarded statelessly
    route(NATMANAGE);
*} else if ( is_method("NOTIFY") ) {**
**    # Add Record-Route for in-dialog NOTIFY as per 
RFC 6665.**

**    record_route();**
**    }*
    route(RELAY);
    exit;
    }

---
I'm SoCIaL, MayBe

El 7/09/2022 a las 12:01 a. m., Jerry Kendall escribió:


Hi There

Hope someone can point me in the right direction.


The config below allows the device to register and will send the 200 
OK without issue


It also allows the SUBSCRIBES to get in and a 202 Accepted is returned 
properly.


It also allows the device to send OPTION keepalives and they get 
in/out OK



The issue I am having at the moment is the outbount NOTIFY ofr the 
SUBS and outbound OPTIONS for heartbeat.



Can someone please point me at what I need to change so the OUTBOUND 
messages get delivered.



The issue I see is NAT related as the devices are behind firewalls at 
location A.


The Kamailio system and the inner PBX are all on Public IP.





debug=3
memdbg=5
memlog=5

log_facility=LOG_LOCAL0

fork=yes
children=4

listen=udp:192.168.1.3:8081
port=8081

#!define FLAG_FROM_SWITCH 1
#!define FLAG_FROM_DEVICE 2

mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/"
loadmodule "tm.so"
loadmodule "sl.so"
loadmodule "pv.so"
loadmodule "rr.so"
loadmodule "textops.so"
loadmodule "textopsx.so"
loadmodule "usrloc.so"
loadmodule "xlog.so"
loadmodule "path.so"
loadmodule "nathelper.so"
loadmodule "siputils.so"

# - nathelper params -
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")


### Routing Logic 

request_route {

    route(CHECK_SOURCE_IP);

    # CANCEL processing
    if (is_method("CANCEL")) {
        if (t_check_trans()) {
            route(RELAY);
        }
        exit;
    }

    route(WITHINDLG);

    t_check_trans();

    if( is_method("OPTIONS") ) {
        t_relay();
    }

    if( is_method("REGISTER|SUBSCRIBE") ) {
        add_path();
    }

    if( is_method("INVITE|REFER") ) {
        record_route();
    }

    if( isflagset(FLAG_FROM_SWITCH) ) {
        t_on_reply("OUTBOUND_REPLY");
    } else {
        $du = "sip:192.168.3.3:5060";
    }

    route(RELAY);
}

# SIP dialogs
route[WITHINDLG] {

    if (has_totag()) {
        # sequential request withing a dialog should
        # take the path determined by record-routing
        if (loose_route()) {
            route(RELAY);
        } else {
            if (is_method("NOTIFY")) {
                route(RELAY);
            }
            if (is_method("SUBSCRIBE") && uri == myself) {
                # in-dialog subscribe requests
                exit;
            }
            if (is_method("ACK")) {
                if (t_check_trans()) {
                    # no loose-route, but stateful ACK;
                    # must be an ACK after a 487
                    # or e.g. 404 from upstream server
                    t_relay();
                    exit;
                } else {
                    # ACK without matching transaction ... ignore and 
discard
                    #xlog("ACK without matching transaction ... ignore 
and discard");

                    exit;
                }
            }
            sl_send_reply("404","Not here");
        }
        exit;
    }
}

onreply_route[OUTBOUND_REPLY]
{
    route(NAT_TEST_AND_CORRECT);
}

route[NAT_TEST_AND_CORRECT]
{
    if (nat_uac_test("3")) {
        if (is_method("REGISTER|SUBSCRIBE")) {
            fix_nated_register();
        } else {
            fix_nated_contact();
        }
        force_rport();
        }

    if (has_body("application/sdp") && nat_uac_test("8")) {
        fix_nated_sdp("10");
    }
}

route[RELAY]
{
    if (!t_relay()) {
        sl_reply_error();
    }
    exit;
}

route[LOGIT] {

    xlog("L_INFO", "LOGIT: $var(spot) -  $rm");

if( $ft == $null )
    xlog("L_INFO", "LOGIT: $var(spot) - From:$fU @ $fd - $fu");
else
    xlog("L_INFO", "LOGIT: $var(spot) - From:$fU @ $fd - $fu:$ft");

if( $tt == $null )
    xlog("L_INFO", "LOGIT: $var(spot) -   To:$tU @ $td - $tu");
else
    xlog("L_INFO", "LOGIT: $var(spot) -   To:$tU @ $td - $tu:$tt");

    xlog("L_INFO", "LOGIT: $var(spot) -  REQ:$rm @ $ru");
    xlog("L_INFO", "LOGIT: $var(spot) - ACT:$(rd{s.select,0,.})");
    xlog("L_INFO", "LOGIT: $var(spot) - HST:$(rd{s.select,1,.})");
    xlog("L_INFO", "LOGIT: $var(spot) -  CNT:$ct");
    xlog("L_INFO", "LOGIT: $var(spot) -  UAC:$ua");

if( $du != $null )
    xlog("L_INFO", "LOGIT: $var(spot) - DURI:$du");
}

route[CHECK_SOURCE_IP]
{
    $var(spot) = 'CHECK_SOURCE_IP';
    route(LOGIT);

Re: [SR-Users] Kamailio v5.6.0 Released - new major version is out

[Social Boh]

Hello,

there are two errors in:

https://www.kamailio.org/w/kamailio-v5-6-0-release-notes/

first:

slack module link to 
https://www.kamailio.org/docs/modules/5.4.x/modules/systemdops.html


tlsa module 
https://www.kamailio.org/docs/modules/5.6.x/modules/tsla.html : The 
requested URL was not found on this server.


Regards

---
I'm SoCIaL, MayBe

El 23/05/2022 a las 8:00 a. m., Daniel-Constantin Mierla escribió:

Kamailio v5.6.0 is out – it comes with 7 new modules and a considerable
set of improvements touching again more than 90 existing modules.

You can read detailed release notes at:

    * https://www.kamailio.org/w/kamailio-v5-6-0-release-notes/

Many thanks to all developers and community members that made possible
this release.

v5.6.0 brings more flexibility and optimizations across many components,
plenty of new variables and transformations, load balancer enhancements,
generic iterators for SIP headers and body lines, alternative http
client, NATS PubSub connector, improvements to STIR/SHAKEN, IMS/VoLTE,
rtpengine and tls as well as more functions exported to KEMI interface
... just to list only a few here. Enjoy Kamailio v5.5.0!

Thank you for flying Kamailio!
Daniel



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Re: [SR-Users] Accept IP/Domain Register

[Social Boh]

My way:

from original kamailio.cfg in global parameters:

/*#!define WITH_MYSQL*//*
*//*#!define WITH_USRLOCDB*//*
*//*#!define WITH_AUTH*/

then:

/*auto_aliases=no*/

one entry for each domain/subdomain, example:

/*alias="sip1.kamailio.org"
alias="sip2.kamailio.org"*/

add the user using IP, example:

*kamctl add 100@1.2.3.4 password*

Regards

---
I'm SoCIaL, MayBe

El 10/05/2022 a las 1:02 p. m., Raúl Alexis Betancor Santana escribió:

As allways, there are 1 thousand ways of skinning a cat

You have 3 options:

1) Create the 100 user in all the domains with the same password.
2) Disable domain in auth_check
3) On the routing logic, when you detect your 100 user, do an auth without 
domain chekin (kind of option 2 but only for specific users)

- Original Message -
From: "Social Boh"
To: "Kamailio", "Alex 
Balashov"
Sent: Tuesday, May 10, 2022 6:45:20 PM
Subject: Re: [SR-Users] Accept IP/Domain Register

I think my question not is clear.

I'd like create/configure, for example, user 100 and this user can
register with kamailio using Kamailio IP or Kamailio associated Domains;
this using digest authentication.

Regards

---
I'm SoCIaL, MayBe

El 10/05/2022 a las 12:26 p. m., Alex Balashov escribió:

Yes. There is no rule that you have to issue a digest authentication challenge 
in response to registrations.

You can handle them like any other request, including validating the source IP 
against an IP ACL (e.g. allow_trusted(), or hard-coded like if(src_ip == 
1.2.3.4)).

— Alex

—
Sent from mobile, with due apologies for brevity and errors.


On May 10, 2022, at 12:49 PM, Social Boh  wrote:

Hello,

maybe is a very basic question; How Kamailio can accept any kind of register, 
over IP or domains.

Example, accept authenticate REGISTER from ip 1.2.3.4 or sip.domain.org or 
sip1.domain.org, domains associated with ip 1.2.3.4

Thank you in advance.

Regards

--
---
I'm SoCIaL, MayBe


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Re: [SR-Users] Accept IP/Domain Register

[Social Boh]

I think my question not is clear.

I'd like create/configure, for example, user 100 and this user can 
register with kamailio using Kamailio IP or Kamailio associated Domains; 
this using digest authentication.


Regards

---
I'm SoCIaL, MayBe

El 10/05/2022 a las 12:26 p. m., Alex Balashov escribió:

Yes. There is no rule that you have to issue a digest authentication challenge 
in response to registrations.

You can handle them like any other request, including validating the source IP 
against an IP ACL (e.g. allow_trusted(), or hard-coded like if(src_ip == 
1.2.3.4)).

— Alex

—
Sent from mobile, with due apologies for brevity and errors.


On May 10, 2022, at 12:49 PM, Social Boh  wrote:

Hello,

maybe is a very basic question; How Kamailio can accept any kind of register, 
over IP or domains.

Example, accept authenticate REGISTER from ip 1.2.3.4 or sip.domain.org or 
sip1.domain.org, domains associated with ip 1.2.3.4

Thank you in advance.

Regards

--
---
I'm SoCIaL, MayBe


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[SR-Users] Accept IP/Domain Register

[Social Boh]

Hello,

maybe is a very basic question; How Kamailio can accept any kind of 
register, over IP or domains.


Example, accept authenticate REGISTER from ip 1.2.3.4 or sip.domain.org 
or sip1.domain.org, domains associated with ip 1.2.3.4


Thank you in advance.

Regards

--
---
I'm SoCIaL, MayBe


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Re: [SR-Users] SIP server port service cannot be changed from default value (5060)

[Social Boh]

you appear are using some web interface to configure Kamailio so ask on 
this interface mailing list


---
I'm SoCIaL, MayBe

El 18/02/2022 a las 5:40 p. m., bayu p escribió:
I tried for couple time, it still not working/fly kamailio by changing 
kamailio.cfg, and port SIP client. As captured previously.


On Thu, Feb 17, 2022, 12:31 bayu p  wrote:

Sorry, I left attachment the client port configuration, here it is
the same as the SIP server port:
image.png

SIP server:
listen=udp:127.0.0.1:9060 
voicemail.srv_port = "9060" desc "VoiceMail Port"

Pada tanggal Rab, 16 Feb 2022 pukul 19.18 Patrick Karton
 menulis:

There is no need to change source code for that.


First you have to change kamailio default port liké said before.

Second you restart kamailio.

Third you make sure your client connects kamailio on the new
port you provided and not the default one.

Use sngprep to troubleshoot.

Le 16 févr. 2022 13:00, bayu p  a écrit :

When I changed the kamailio.cfg, then I performed tcpdump
to tracing, found that "Type: 3 (Destination
unreachable) Code: 3 (Port unreachable)".
That means, the defined port is cannot be reached by the
client sip server side.
Any other process to change the port ?


Thanks & Regards



Pada tanggal Rab, 16 Feb 2022 pukul 18.53 bayu p
 menulis:

It is my basic process, that changes default port in
kamailio.cfg, then I share this problem here.
Does need more procedure to change the port besides
assign other port in kamailio.cfg ?
How to compile the source, because there are many
parameter use 5060 port in the source, it may need
more changes further.

Pada tanggal Rab, 16 Feb 2022 pukul 15.59 Patrick
Karton  menulis:

you have to explicitly set the new port you want.

listen=udp:127.0.0.1:7070 


*De :* sr-users
 de la part
de bayu p 
*Envoyé :* mercredi 16 février 2022 09:16
*À :* sr-users@lists.kamailio.org

*Objet :* [SR-Users] SIP server port service
cannot be changed from default value (5060)

Hello Experts,

Please help me to change port service from default
5060 port into other port due to security reason.
Since I have change port in kamailio.cfg file, but
services cannot fly normally that shown client
unable login online (status not online).
image



Besides, I need to run the kamailio services after
install default kamailio, I have not found where
procedure found in this installation process on
github.com/kamailio .

Could you please share if there another steps of
compiling Kamailio source, install, deploy
process, and running command for this specific
installation process?

Thanks
Gb

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Re: [SR-Users] Comunication between devices RTP/SRTP and viceversa

[Social Boh]

Thank you to all for the tips...

Regards

---
I'm SoCIaL, MayBe

El 27/01/2022 a las 6:39 p. m., Sergiu Pojoga escribió:
You would achieve this in a failure route, detecting the 488|415 
negative replies then resuming the branch with a different SDP profile.


However, as others have said already, not all UAs will accept this 
"hack". Some will consider the new branch with a CSeq NOT incrementing 
as retransmission - and reject it with something like "482 Merged 
Request".


https://tools.ietf.org/html/rfc3261#section-8.2.2.2

8.2.2.2 Merged Requests

If the request has no tag in the To header field, the UAS core MUST
check the request against ongoing transactions. If the From tag,
Call-ID, and CSeq exactly match those associated with an ongoing
transaction, but the request does not match that transaction (based
on the matching rules in Section 17.2.3), the UAS core SHOULD
generate a 482 (Loop Detected) response and pass it to the server
transaction.

Regards,
--Sergiu

On Thu, Jan 27, 2022 at 6:26 PM Marrold  wrote:

Hi,

Unfortunately in my experience "Opportunistic SRTP" support is
broken in many UAs. There are (at least) two ways of doing it
covered indirectly in RFC3264
<https://tools.ietf.org/html/rfc3264> and the second explicitly in
RFC8643 <https://www.rfc-editor.org/rfc/rfc8643.html> and devices
seem to handle them differently. I've seen end devices that /can
/support SRTP choose to use standard RTP when offered the wrong
type of SDP body.

One thing I considered (but haven't tested so far) is using a late
offer to discover what format the end device expects before
sending an offer, but that might cause interop problems of its own.

    Cheers

On Thu, Jan 27, 2022 at 10:54 PM Social Boh 
wrote:

I can examine the request but how examine the answer if is a
Error?
Using a failure route?

Regards

---
I'm SoCIaL, MayBe

El 27/01/2022 a las 5:39 p. m., Alex Balashov escribió:
> Does an examination of the SDP offer and answer respectively
not divine the issue?
>
>> On Jan 27, 2022, at 5:35 PM, Social Boh
 wrote:
>>
>> I'm trying to configure Kamailio to permit communication
between devices:
>>
>> some use TLS and SRTP
>>
>> some UDP and RTP
>>
>> with RTPEngine in the middle.
>>
>> A priori I don't know if the device support SRTP or RTP so
in the routing I have to detect on the fly how to act. The
problem is this:
>>
>> Call from RTP to SRTP device: 488 Not Acceptable here
>>
>> Call from SRTP to RTP device: 415 Unsupported Media Type
>>
>> Any hint?
>>
>> Thank you
>>
>>   Regards
>>
>> --
>> ---
>> I'm SoCIaL, MayBe
>>
>>
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Re: [SR-Users] Comunication between devices RTP/SRTP and viceversa

[Social Boh]

I can examine the request but how examine the answer if is a Error? 
Using a failure route?


Regards

---
I'm SoCIaL, MayBe

El 27/01/2022 a las 5:39 p. m., Alex Balashov escribió:

Does an examination of the SDP offer and answer respectively not divine the 
issue?


On Jan 27, 2022, at 5:35 PM, Social Boh  wrote:

I'm trying to configure Kamailio to permit communication between devices:

some use TLS and SRTP

some UDP and RTP

with RTPEngine in the middle.

A priori I don't know if the device support SRTP or RTP so in the routing I 
have to detect on the fly how to act. The problem is this:

Call from RTP to SRTP device: 488 Not Acceptable here

Call from SRTP to RTP device: 415 Unsupported Media Type

Any hint?

Thank you

  Regards

--
---
I'm SoCIaL, MayBe


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[SR-Users] Comunication between devices RTP/SRTP and viceversa

[Social Boh]

I'm trying to configure Kamailio to permit communication between devices:

some use TLS and SRTP

some UDP and RTP

with RTPEngine in the middle.

A priori I don't know if the device support SRTP or RTP so in the 
routing I have to detect on the fly how to act. The problem is this:


Call from RTP to SRTP device: 488 Not Acceptable here

Call from SRTP to RTP device: 415 Unsupported Media Type

Any hint?

Thank you

 Regards

--
---
I'm SoCIaL, MayBe


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Re: [SR-Users] kamailio log dialog correlation

[Social Boh]

check the Kamailio LOG. is the module generates some error you can see 
the data on the LOG


---
I'm SoCIaL, MayBe

El 24/01/2022 a las 10:41 a. m., marek escribió:


[root@sbc~]# grep cfgutils /etc/kamailio/kamailio.cfg
loadmodule "cfgutils.so"

is there a way how can i check if module is loaded without errors? 
(something like Asterisk's "module show like")


kamailio 5.5.3

Marek


Dne 22/01/2022 v 17:27 Chad napsal(a):

loadmodule "cfgutils.so"


--
^C


On 1/22/22 2:36 AM, marek wrote:

thanks

now i understand difference between log_prefix and modparam("xlog", 
"prefix", "something")


so i'm trying this

log_prefix="{D$dlg(h_id) $cfg(route)}"

but i got this

ERROR:  [core/pvapi.c:924]: pv_parse_spec2(): error searching 
pvar "cfg"
ERROR:  [core/pvapi.c:1127]: pv_parse_spec2(): wrong char 
[r/114] in [$cfg(route)] at [5 (5)]
ERROR:  [core/dprint.c:463]: log_prefix_init(): wrong 
format[{D$dlg(h_id) $cfg(route)}]


it's the same if its used in

xlog("L_INFO", "D$dlg(h_id) $cfg(route) some log")

do you think there is a module missing?

Marek


Dne 19/01/2022 v 16:23 Henning Westerholt napsal(a):

Hello,

you could try to use log_prefix. Have not tried it yet with a 
dialog variable.


http://www.kamailio.org/wiki/cookbooks/5.5.x/core#log_prefix

Regarding output of a line number, you could try to use this one:

http://www.kamailio.org/wiki/cookbooks/5.4.x/pseudovariables#cfg_key_-_config_file_attributes 



Cheers,

Henning



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Re: [SR-Users] dispatcher lb

[Social Boh]

Maybe a SIP capture can help.

Regards

---
I'm SoCIaL, MayBe

El 21/01/2022 a las 3:01 p. m., Gokan Atmaca escribió:

Hello

I'm load balancing with Dispatcher. The call comes in but hangs up
without picking up the phone. What could be the problem? The settings
are as follows.


dispatcher.list
1 sip:18.X.X.X.X:5060 0 0 maxload=20
1 sip:67.X.X.X.X:5060 0 0 maxload=20


# Dispatch requests
route[DISPATCH] {
# round robin dispatching on gateways group '1'
if(!ds_select_dst("1", "4"))
 {
send_reply("404", "No destination");
exit;
 }
 xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");
 t_on_failure("RTF_DISPATCH");
 route(RELAY);
 exit;
}



{
 NRSETS: 1
 RECORDS: {
 SET: {
 ID: 1
 TARGETS: {
 DEST: {
 URI: sip:X.X.X.X:5060
 FLAGS: AP
 PRIORITY: 0
 ATTRS: {
 BODY: maxload=20
 DUID:
 MAXLOAD: 20
 WEIGHT: 0
 RWEIGHT: 0
 SOCKET:
 }
 LATENCY: {
 AVG: 4.13
 STD: 0.398000
 EST: 4.006000
 MAX: 6
 TIMEOUT: 0
 }
 }
 DEST: {
 URI: sip:X.X.X.X:5060
 FLAGS: AP
 PRIORITY: 0
 ATTRS: {
 BODY: maxload=20
 DUID:
 MAXLOAD: 20
 WEIGHT: 0
 RWEIGHT: 0
 SOCKET:
 }
 LATENCY: {
 AVG: 22.282000
 STD: 2.345000
 EST: 23.075000
 MAX: 42
 TIMEOUT: 0
 }
 }
 }
 }
 }
}



--
⢀⣴⠾⠻⢶⣦⠀
⣾⠁⢠⠒⠀⣿⡁ Debian - The universal operating system
⢿⡄⠘⠷⠚⠋⠀ https://www.debian.org
⠈⠳⣄

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Re: [SR-Users] Call limiting with Kamailio

[Social Boh]

Hi,

DMQ is a good option and you can test if restarting one Kamailio, this 
kamailio sync the data with the other/s.


DMQ_USRLOC using this logic with parameter sync.

Regards

---
I'm SoCIaL, MayBe

El 26/11/2021 a las 7:38 a. m., Karsten Horsmann escribió:

Hi,

using dialog or dlg makes your life easier.

-> Counting calls with htable and replicating with DMQ to other node
(can the data survive from a restart?)

htable can write back to the database. But that solution needs a bit 
more (getting the data - manually or via dialog/dlg and replicate that 
htable, how to work with htable)?


As an untested idea - you can define two htables. One htable_nodeA the 
other htable_nodeB.
Node A writes only into htable_nodeA and vice versa. So there should 
no race condition.
And then you may check htable_nodeA customerX calls and htable_nodeB 
customerX calls (sum that) on new calls.


Many ways.

Kind regards
Karsten


Am Fr., 26. Nov. 2021 um 13:26 Uhr schrieb Henning Westerholt 
:


Hello,

you could also use dialog profile functionality (single server or
also with DMQ sync), also probably also the new dlgs module
(single server)

Cheers,

Henning


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Re: [SR-Users] Access via script to location data in the memory

[Social Boh]

So, my final configuration is:

One domain with AWS Route53 same weight configuration: 50% server1 and 
50% server2


The user register over the domain.

Two Kamailio Servers with DMQ and DMQ_USRLOC modules configured; 1.2.3.4 
first server IP and 5.6.7.8 second server IP


Most important configurations *Server1*:

/*listen=udp:1.2.3.4:5060
listen=udp:1.2.3.4:5222*/

/*#  dmq params 
modparam("dmq", "server_address", "sip:1.2.3.4")
modparam("dmq", "notification_address", "sip:5.6.7.8")
modparam("dmq", "multi_notify", 0)
modparam("dmq", "num_workers", 4)
modparam("dmq", "ping_interval", 30)

#  dmq_usrloc params 
modparam("dmq_usrloc", "enable", 1)
modparam("dmq_usrloc", "sync", 1)
modparam("dmq_usrloc", "batch_size", 10)
modparam("dmq_usrloc", "batch_usleep", 1000)
modparam("dmq_usrloc", "batch_msg_size", 6)
modparam("dmq_usrloc", "batch_msg_contacts", 50)
modparam("dmq_usrloc", "usrloc_domain", "location")
modparam("dmq_usrloc", "replicate_socket_info", 0)
modparam("dmq_usrloc", "usrloc_delete", 1)*/

before the block begin with:

/*$avp(oexten) = $rU;
    if (!lookup("location")) {*/

I put

/*if (!dmq_is_from_node()) {*//*
*//*if(reg_fetch_contacts("location", "$ru", "contacto")) {*//*
*//*if($(ulc(contacto=>socket))) == 0 {*//*
*//*rewritehost("5.6.7.8");*//*
*//*    }*//*
*//*    route(RELAY);*//*
*//*    }*//*
*//*    }*/

If the socket value is null (0) means the user is registered on the 
second Kamailio so send INVITE to second Kamailio


/*route[AUTH] {*//*
*//*#!ifdef WITH_AUTH*//*
*//*
*//*if (dmq_is_from_node()) {*//*
*//*    $ru = $tu;*//*
*//*return;*//*
*//*}*/

If INVITE comes from other DMQ node don't need authentication but I have 
to change RURI with To URI because I work in a multidomain setting and 
I'm  using the does_uri_exist function to know if the request is local 
or not.


*Server2*:

/*listen=udp:5.6.7.8:5060
listen=udp:5.6.7.8:5222*/

/**/

/*#  dmq params 
modparam("dmq", "server_address", "sip:5.6.7.8")
modparam("dmq", "notification_address", "sip:1.2.3.4")
modparam("dmq", "multi_notify", 0)
modparam("dmq", "num_workers", 4)
modparam("dmq", "ping_interval", 30)

#  dmq_usrloc params 
modparam("dmq_usrloc", "enable", 1)
modparam("dmq_usrloc", "sync", 1)
modparam("dmq_usrloc", "batch_size", 10)
modparam("dmq_usrloc", "batch_usleep", 1000)
modparam("dmq_usrloc", "batch_msg_size", 6)
modparam("dmq_usrloc", "batch_msg_contacts", 50)
modparam("dmq_usrloc", "usrloc_domain", "location")
modparam("dmq_usrloc", "replicate_socket_info", 0)
modparam("dmq_usrloc", "usrloc_delete", 1)*/

before the block begin with:

/*$avp(oexten) = $rU;
    if (!lookup("location")) {*/

/**/

/*if (!dmq_is_from_node()) {*//*
*//*    if(reg_fetch_contacts("location", "$ru", 
"contacto")) {*//*

*//*    if($(ulc(contacto=>socket))) == 0 {*//*
*//*    rewritehost("1.2.3.4");*//*
*//*    }*//*
*//*    route(RELAY);*//*
*//*    }*//*
*//*    }*/

If the socket value is null (0) means the user is registered on the 
first Kamailio so send INVITE to first Kamailio


/*route[AUTH] {
#!ifdef WITH_AUTH

if (dmq_is_from_node()) {
    $ru = $tu;
return;
}*/

If INVITE comes from other DMQ node don't need authentication but I have 
to change RURI with To URI because I work in a multidomain setting and 
I'm  using the does_uri_exist function to know if the request is local 
or not.


LIMITATIONS:

 * works only with a device for user. If a user registered from two o
   more devices I don't know how achieve the same behavior
 * works with Two Kamailio due to rewritehost function...
 * maybe the routing can be better

Regards

---
I'm SoCIaL, MayBe

El 31/07/2021 a las 5:23 a. m., Henning Westerholt escribió:


Great, thanks for the confirmation.

Henning

*From:* sr-users  *On Behalf Of 
*Social Boh

*Sent:* Friday, July 30, 2021 8:36 PM
*To:* Kamailio (SER) - Users Mailing List 
*Subject:* Re: [SR-Users] Access via script to location data in the memory

Works!

I can access location data presents only in the memory. Now, next ste

Re: [SR-Users] Access via script to location data in the memory

[Social Boh]

Works!

I can access location data presents only in the memory. Now, next step 
is using $(ulc(contacto=>socket)) to decide which Kamailio have to 
process the call.


Thank you

---
I'm SoCIaL, MayBe

El 30/07/2021 a las 11:51 a. m., Henning Westerholt escribió:


Hi,

just quickly looked in the registrar code, it uses the standard usrloc 
functions to get data from memory or database.


Maybe just give it a try, I think it should work also in in-memory/dmq 
mode.


Cheers,

Henning

*From:* sr-users  *On Behalf Of 
*Social Boh

*Sent:* Friday, July 30, 2021 6:42 PM
*To:* Kamailio (SER) - Users Mailing List 
*Subject:* Re: [SR-Users] Access via script to location data in the memory

Hello,

reg_fetch_contacts use a table (I understood) to looking for data 
about a user.


I'm using USRLOC module without database.

Regards

---
I'm SoCIaL, MayBe

El 30/07/2021 a las 11:18 a. m., Henning Westerholt escribió:

Hello,

have you looked to the reg_fetch_contact as already pointed out?
Then you can access e.g. the socket with the $ulc PV, see the end
of the registrar docs for an example.

Cheers,

Henning

-- 


Henning Westerholt – https://skalatan.de/blog/
<https://skalatan.de/blog/>

Kamailio services – https://gilawa.com <https://gilawa.com/>

*From:* sr-users 
<mailto:sr-users-boun...@lists.kamailio.org> *On Behalf Of *Social Boh
*Sent:* Friday, July 30, 2021 6:11 PM
*To:* Kamailio (SER) - Users Mailing List
 <mailto:sr-users@lists.kamailio.org>
*Subject:* Re: [SR-Users] Access via script to location data in
the memory

Hello,

maybe is better I explain what I'd like to achieve:

I have two Kamailio using DMQ and DMQ_USRLOC modules so each
REGISTER is replicate on each Kamailio. I'm using DNS weight to
distribute the REGISTERs between the 2 Kamailios.

The only difference I have seen in the location data, with *kamctl
ul show* command, is the Kamailio receives REGISTER have socket
parameter with transport, ip, y port like udp:1.2.3.4:5060 where
1.2.3.4 is Kamailio public IP; the other Kamailio have this field
empty.

When a call arrive to one Kamailio, querying the Socket field I
can known if I have to send the INVITE locally or forward the
INVITE to second Kamailio.

I don't know if there is other way to achieve this goal.

I think PATH protocol not apply here because If I reply the
REGISTER to the second Kamailio adding path header, I don't need
DMQ_USRLOC.

The idea is to take advantage of using these DMQ related modules.

Regards

---

I'm SoCIaL, MayBe

El 29/07/2021 a las 5:16 p. m., Henning Westerholt escribió:

Hello,

  


There are different functions in the registrar module, check them 
out:https://kamailio.org/docs/modules/devel/modules/registrar.html#idm576  
<https://kamailio.org/docs/modules/devel/modules/registrar.html#idm576>

  


reg_fetch_contacts(..) might be something you could use.

  


Cheers,

  


Henning

  

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Re: [SR-Users] Access via script to location data in the memory

[Social Boh]

Works!

I can access location data presents only in the memory. Now, next step 
is using $(ulc(contacto=>socket)) to decide which kamailio have to 
process the call.


Thank you

---
I'm SoCIaL, MayBe

El 30/07/2021 a las 11:51 a. m., Henning Westerholt escribió:


Hi,

just quickly looked in the registrar code, it uses the standard usrloc 
functions to get data from memory or database.


Maybe just give it a try, I think it should work also in in-memory/dmq 
mode.


Cheers,

Henning

*From:* sr-users  *On Behalf Of 
*Social Boh

*Sent:* Friday, July 30, 2021 6:42 PM
*To:* Kamailio (SER) - Users Mailing List 
*Subject:* Re: [SR-Users] Access via script to location data in the memory

Hello,

reg_fetch_contacts use a table (I understood) to looking for data 
about a user.


I'm using USRLOC module without database.

Regards

---
I'm SoCIaL, MayBe

El 30/07/2021 a las 11:18 a. m., Henning Westerholt escribió:

Hello,

have you looked to the reg_fetch_contact as already pointed out?
Then you can access e.g. the socket with the $ulc PV, see the end
of the registrar docs for an example.

Cheers,

Henning

-- 


Henning Westerholt – https://skalatan.de/blog/
<https://skalatan.de/blog/>

Kamailio services – https://gilawa.com <https://gilawa.com/>

*From:* sr-users 
<mailto:sr-users-boun...@lists.kamailio.org> *On Behalf Of *Social Boh
*Sent:* Friday, July 30, 2021 6:11 PM
*To:* Kamailio (SER) - Users Mailing List
 <mailto:sr-users@lists.kamailio.org>
*Subject:* Re: [SR-Users] Access via script to location data in
the memory

Hello,

maybe is better I explain what I'd like to achieve:

I have two Kamailio using DMQ and DMQ_USRLOC modules so each
REGISTER is replicate on each Kamailio. I'm using DNS weight to
distribute the REGISTERs between the 2 Kamailios.

The only difference I have seen in the location data, with *kamctl
ul show* command, is the Kamailio receives REGISTER have socket
parameter with transport, ip, y port like udp:1.2.3.4:5060 where
1.2.3.4 is Kamailio public IP; the other Kamailio have this field
empty.

When a call arrive to one Kamailio, querying the Socket field I
can known if I have to send the INVITE locally or forward the
INVITE to second Kamailio.

I don't know if there is other way to achieve this goal.

I think PATH protocol not apply here because If I reply the
REGISTER to the second Kamailio adding path header, I don't need
DMQ_USRLOC.

The idea is to take advantage of using these DMQ related modules.

Regards

---

I'm SoCIaL, MayBe

El 29/07/2021 a las 5:16 p. m., Henning Westerholt escribió:

Hello,

  


There are different functions in the registrar module, check them 
out:https://kamailio.org/docs/modules/devel/modules/registrar.html#idm576  
<https://kamailio.org/docs/modules/devel/modules/registrar.html#idm576>

  


reg_fetch_contacts(..) might be something you could use.

  


Cheers,

  


Henning

  

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Re: [SR-Users] Access via script to location data in the memory

[Social Boh]

I'll try and report.

Thank you

---
I'm SoCIaL, MayBe

El 30/07/2021 a las 11:51 a. m., Henning Westerholt escribió:


Hi,

just quickly looked in the registrar code, it uses the standard usrloc 
functions to get data from memory or database.


Maybe just give it a try, I think it should work also in in-memory/dmq 
mode.


Cheers,

Henning

*From:* sr-users  *On Behalf Of 
*Social Boh

*Sent:* Friday, July 30, 2021 6:42 PM
*To:* Kamailio (SER) - Users Mailing List 
*Subject:* Re: [SR-Users] Access via script to location data in the memory

Hello,

reg_fetch_contacts use a table (I understood) to looking for data 
about a user.


I'm using USRLOC module without database.

Regards

---
I'm SoCIaL, MayBe

El 30/07/2021 a las 11:18 a. m., Henning Westerholt escribió:

Hello,

have you looked to the reg_fetch_contact as already pointed out?
Then you can access e.g. the socket with the $ulc PV, see the end
of the registrar docs for an example.

Cheers,

Henning

-- 


Henning Westerholt – https://skalatan.de/blog/
<https://skalatan.de/blog/>

Kamailio services – https://gilawa.com <https://gilawa.com/>

*From:* sr-users 
<mailto:sr-users-boun...@lists.kamailio.org> *On Behalf Of *Social Boh
*Sent:* Friday, July 30, 2021 6:11 PM
*To:* Kamailio (SER) - Users Mailing List
 <mailto:sr-users@lists.kamailio.org>
*Subject:* Re: [SR-Users] Access via script to location data in
the memory

Hello,

maybe is better I explain what I'd like to achieve:

I have two Kamailio using DMQ and DMQ_USRLOC modules so each
REGISTER is replicate on each Kamailio. I'm using DNS weight to
distribute the REGISTERs between the 2 Kamailios.

The only difference I have seen in the location data, with *kamctl
ul show* command, is the Kamailio receives REGISTER have socket
parameter with transport, ip, y port like udp:1.2.3.4:5060 where
1.2.3.4 is Kamailio public IP; the other Kamailio have this field
empty.

When a call arrive to one Kamailio, querying the Socket field I
can known if I have to send the INVITE locally or forward the
INVITE to second Kamailio.

I don't know if there is other way to achieve this goal.

I think PATH protocol not apply here because If I reply the
REGISTER to the second Kamailio adding path header, I don't need
DMQ_USRLOC.

The idea is to take advantage of using these DMQ related modules.

Regards

---

I'm SoCIaL, MayBe

El 29/07/2021 a las 5:16 p. m., Henning Westerholt escribió:

Hello,

  


There are different functions in the registrar module, check them 
out:https://kamailio.org/docs/modules/devel/modules/registrar.html#idm576  
<https://kamailio.org/docs/modules/devel/modules/registrar.html#idm576>

  


reg_fetch_contacts(..) might be something you could use.

  


Cheers,

  


Henning

  

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Re: [SR-Users] Access via script to location data in the memory

[Social Boh]

Hello,

reg_fetch_contacts use a table (I understood) to looking for data about 
a user.


I'm using USRLOC module without database.

Regards


---
I'm SoCIaL, MayBe

El 30/07/2021 a las 11:18 a. m., Henning Westerholt escribió:


Hello,

have you looked to the reg_fetch_contact as already pointed out? Then 
you can access e.g. the socket with the $ulc PV, see the end of the 
registrar docs for an example.


Cheers,

Henning

--

Henning Westerholt – https://skalatan.de/blog/ <https://skalatan.de/blog/>

Kamailio services – https://gilawa.com <https://gilawa.com/>

*From:* sr-users  *On Behalf Of 
*Social Boh

*Sent:* Friday, July 30, 2021 6:11 PM
*To:* Kamailio (SER) - Users Mailing List 
*Subject:* Re: [SR-Users] Access via script to location data in the memory

Hello,

maybe is better I explain what I'd like to achieve:

I have two Kamailio using DMQ and DMQ_USRLOC modules so each REGISTER 
is replicate on each Kamailio. I'm using DNS weight to distribute the 
REGISTERs between the 2 Kamailios.


The only difference I have seen in the location data, with *kamctl ul 
show* command, is the Kamailio receives REGISTER have socket parameter 
with transport, ip, y port like udp:1.2.3.4:5060 where 1.2.3.4 is 
Kamailio public IP; the other Kamailio have this field empty.


When a call arrive to one Kamailio, querying the Socket field I can 
known if I have to send the INVITE locally or forward the INVITE to 
second Kamailio.


I don't know if there is other way to achieve this goal.

I think PATH protocol not apply here because If I reply the REGISTER 
to the second Kamailio adding path header, I don't need DMQ_USRLOC.


The idea is to take advantage of using these DMQ related modules.

Regards

---
I'm SoCIaL, MayBe

El 29/07/2021 a las 5:16 p. m., Henning Westerholt escribió:

Hello,

There are different functions in the registrar module, check them 
out:https://kamailio.org/docs/modules/devel/modules/registrar.html#idm576  
<https://kamailio.org/docs/modules/devel/modules/registrar.html#idm576>

reg_fetch_contacts(..) might be something you could use.

Cheers,

Henning

__
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  * sr-users@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the 
sender!
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Re: [SR-Users] Access via script to location data in the memory

[Social Boh]

Hello,

maybe is better I explain what I'd like to achieve:

I have two Kamailio using DMQ and DMQ_USRLOC modules so each REGISTER is 
replicate on each Kamailio. I'm using DNS weight to distribute the 
REGISTERs between the 2 Kamailios.


The only difference I have seen in the location data, with *kamctl ul 
show* command, is the Kamailio receives REGISTER have socket parameter 
with transport, ip, y port like udp:1.2.3.4:5060 where 1.2.3.4 is 
Kamailio public IP; the other Kamailio have this field empty.


When a call arrive to one Kamailio, querying the Socket field I can 
known if I have to send the INVITE locally or forward the INVITE to 
second Kamailio.


I don't know if there is other way to achieve this goal.

I think PATH protocol not apply here because If I reply the REGISTER to 
the second Kamailio adding path header, I don't need DMQ_USRLOC.


The idea is to take advantage of using these DMQ related modules.

Regards

---
I'm SoCIaL, MayBe

El 29/07/2021 a las 5:16 p. m., Henning Westerholt escribió:

Hello,

There are different functions in the registrar module, check them out: 
https://kamailio.org/docs/modules/devel/modules/registrar.html#idm576

reg_fetch_contacts(..) might be something you could use.

Cheers,

Henning

__
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[SR-Users] Access via script to location data in the memory

[Social Boh]

Hello List,

is there a way to access location data from kamailio script?

I mean data I can see with the command:

kamctl ul show

Regards

--
---
I'm SoCIaL, MayBe


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 * sr-users@lists.kamailio.org
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Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record

[Social Boh]

hello,

sorry to send the new message directly to Henning



Hello,

with dialog db_mode 1 and Primary/Primary MariaDB Replication If I 
switch Off primary Kamailio during a call and move the call to Replica 
and then hangup, no entry on acc_cdrs table. acc table is ok.


Regards



---
I'm SoCIaL, MayBe

El 20/07/2021 a las 4:09 a. m., Henning Westerholt escribió:


Hello,

Usage of DB to store dialogs. See 
https://kamailio.org/docs/modules/devel/modules/dialog.html#dialog.p.db_mode 
<https://kamailio.org/docs/modules/devel/modules/dialog.html#dialog.p.db_mode>– 
in particular mode 1 and 3.


Cheers,

Henning

--

Henning Westerholt – https://skalatan.de/blog/ <https://skalatan.de/blog/>

Kamailio services – https://gilawa.com <https://gilawa.com/>

*From:* sr-users  *On Behalf Of 
*Social Boh

*Sent:* Tuesday, July 20, 2021 12:55 AM
*To:* Kamailio (SER) - Users Mailing List 
*Subject:* Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record

Hello

what means "DB loading" ?

Regards

---
I'm SoCIaL, MayBe

El 19/07/2021 a las 9:25 a. m., Henning Westerholt escribió:

Hello,

so, you are saying that in a DMQ replicated case the acc CDR
generation is not working in failover cases.

I just looked briefly to the code; others might have more details.

But I think this is because the acc module do not register to a
callback function for dialog loading on startup for the DMQ
replication. This is done for DB loading, but seems to be not
implemented yet for DMQ synchronisation.

You could open a feature request on our github tracker for this.

Cheers,

Henning

-- 


Henning Westerholt – https://skalatan.de/blog/
<https://skalatan.de/blog/>

Kamailio services – https://gilawa.com <https://gilawa.com/>

*From:* sr-users 
<mailto:sr-users-boun...@lists.kamailio.org> *On Behalf Of *Social Boh
*Sent:* Monday, July 19, 2021 1:58 PM
*To:* Kamailio (SER) - Users Mailing List
 <mailto:sr-users@lists.kamailio.org>
*Subject:* Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record

Hi,

I'm using 5.4.6

I switch off Kamailio on the first server and keepalived move the
virtual IP on the second server so First Kamailio is off. This
during the call.

Before switch off Kamailio and during the call I can see data
dialog on both server but in the LOG /var/log/messages in my case:

server 1 INVITE

server 2 BYE

acc_cdrs table empty on both servers.

acc table first server INVITE; acc table second server BYE

Regards

---

I'm SoCIaL, MayBe

El 18/07/2021 a las 2:33 p. m., M Arqum CH escribió:

Hi SoCIal,

which Kamailio version you are using.

did you able to get the dialog information using the dialog
command.

https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.dlg_list

<https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.dlg_list>

when you received bye on secondary server, is that server
    recognized the bye.

On Sun, Jul 18, 2021 at 3:38 AM Social Boh mailto:soc...@bohboh.info>> wrote:

Hello,

if during a call I stop Primary Kamailio the Secondary
Kamailio take de Virtual IP but don't save call data:

In the Primary database, acc table, I have INVITE data but
not BYE data; no data in acc_cdrs for the call. In the
Secondary Kamailio, acc table, I have BYE data but not
INVITE data.

in the file LOG no data about call cdr.

Maybe a solution is write call data whene the call end?

Regards

---

I'm SoCIaL, MayBe

El 17/07/2021 a las 11:34 a. m., M Arqum CH escribió:

Make sure that your table columns matched the field's
you are trying to insert, please check the syslog for
acc module log, acc module print the cdr in log
facility too, if acc module facing any issue while
writing cdr in db, it will throw error in logs.

Hope this will help you.

On Sat, Jul 17, 2021, 7:33 PM Social Boh
mailto:soc...@bohboh.info>> wrote:

Hello,

i tested with DMQ module and happens the same.

DMQ module loaded before dialog module and on
dialog module parameters:

modparam("dialog", "enable_dmq", 1)

During a call I can see on both Kamailio servers
the dialog data but when switch off the active
kamailio server and the second kamailio server
take the control and hangup the call, no data
   

Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record

[Social Boh]

Hello

what means "DB loading" ?

Regards

---
I'm SoCIaL, MayBe

El 19/07/2021 a las 9:25 a. m., Henning Westerholt escribió:


Hello,

so, you are saying that in a DMQ replicated case the acc CDR 
generation is not working in failover cases.


I just looked briefly to the code; others might have more details.

But I think this is because the acc module do not register to a 
callback function for dialog loading on startup for the DMQ 
replication. This is done for DB loading, but seems to be not 
implemented yet for DMQ synchronisation.


You could open a feature request on our github tracker for this.

Cheers,

Henning

--

Henning Westerholt – https://skalatan.de/blog/ <https://skalatan.de/blog/>

Kamailio services – https://gilawa.com <https://gilawa.com/>

*From:* sr-users  *On Behalf Of 
*Social Boh

*Sent:* Monday, July 19, 2021 1:58 PM
*To:* Kamailio (SER) - Users Mailing List 
*Subject:* Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record

Hi,

I'm using 5.4.6

I switch off Kamailio on the first server and keepalived move the 
virtual IP on the second server so First Kamailio is off. This during 
the call.


Before switch off Kamailio and during the call I can see data dialog 
on both server but in the LOG /var/log/messages in my case:


server 1 INVITE

server 2 BYE

acc_cdrs table empty on both servers.

acc table first server INVITE; acc table second server BYE

Regards

---
I'm SoCIaL, MayBe

El 18/07/2021 a las 2:33 p. m., M Arqum CH escribió:

Hi SoCIal,

which Kamailio version you are using.

did you able to get the dialog information using the dialog
command.

https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.dlg_list

<https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.dlg_list>

when you received bye on secondary server, is that server
recognized the bye.

    On Sun, Jul 18, 2021 at 3:38 AM Social Boh mailto:soc...@bohboh.info>> wrote:

Hello,

if during a call I stop Primary Kamailio the Secondary
Kamailio take de Virtual IP but don't save call data:

In the Primary database, acc table, I have INVITE data but not
BYE data; no data in acc_cdrs for the call. In the Secondary
Kamailio, acc table, I have BYE data but not INVITE data.

in the file LOG no data about call cdr.

Maybe a solution is write call data whene the call end?

Regards

---

I'm SoCIaL, MayBe

El 17/07/2021 a las 11:34 a. m., M Arqum CH escribió:

Make sure that your table columns matched the field's you
are trying to insert, please check the syslog for acc
module log, acc module print the cdr in log facility too,
if acc module facing any issue while writing cdr in db, it
will throw error in logs.

Hope this will help you.

On Sat, Jul 17, 2021, 7:33 PM Social Boh
mailto:soc...@bohboh.info>> wrote:

Hello,

i tested with DMQ module and happens the same.

DMQ module loaded before dialog module and on dialog
module parameters:

modparam("dialog", "enable_dmq", 1)

During a call I can see on both Kamailio servers the
dialog data but when switch off the active kamailio
server and the second kamailio server take the control
and hangup the call, no data acc_cdrs table.

Regards

---

I'm SoCIaL, MayBe

El 17/07/2021 a las 1:24 a. m., M Arqum CH escribió:

Hi Social,

Replication of dialog using dmq will solve
your problem, you need to load dmq module before 
loading dialog module and enable replication of
dialogs in dialog module, please check dialog
module conf option for more details.

Cheers

On Fri, Jul 16, 2021, 7:32 PM Social Boh
mailto:soc...@bohboh.info>>
wrote:

Im' testinf Kamailio HA with Keepalived.

I have a problem with acc_cdrs table in this
situation:

A call B and when the two users are in the
call I switch off the active
Kamailio. Keepalived pass the Virtual IP to
the secondary Kamailio and
audio still work because I'm using a remote
RTPENGINE server for both
Kamailio.

The problem is when the call end. On the acc
table i can see INVITE and
  

Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record

[Social Boh]

Hi,

I think could be very useful replicate acc and acc_cdrs data via DMQ.

Regards

---
I'm SoCIaL, MayBe

El 19/07/2021 a las 9:25 a. m., Henning Westerholt escribió:


Hello,

so, you are saying that in a DMQ replicated case the acc CDR 
generation is not working in failover cases.


I just looked briefly to the code; others might have more details.

But I think this is because the acc module do not register to a 
callback function for dialog loading on startup for the DMQ 
replication. This is done for DB loading, but seems to be not 
implemented yet for DMQ synchronisation.


You could open a feature request on our github tracker for this.

Cheers,

Henning

--

Henning Westerholt – https://skalatan.de/blog/ <https://skalatan.de/blog/>

Kamailio services – https://gilawa.com <https://gilawa.com/>

*From:* sr-users  *On Behalf Of 
*Social Boh

*Sent:* Monday, July 19, 2021 1:58 PM
*To:* Kamailio (SER) - Users Mailing List 
*Subject:* Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record

Hi,

I'm using 5.4.6

I switch off Kamailio on the first server and keepalived move the 
virtual IP on the second server so First Kamailio is off. This during 
the call.


Before switch off Kamailio and during the call I can see data dialog 
on both server but in the LOG /var/log/messages in my case:


server 1 INVITE

server 2 BYE

acc_cdrs table empty on both servers.

acc table first server INVITE; acc table second server BYE

Regards

---
I'm SoCIaL, MayBe

El 18/07/2021 a las 2:33 p. m., M Arqum CH escribió:

Hi SoCIal,

which Kamailio version you are using.

did you able to get the dialog information using the dialog
command.

https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.dlg_list

<https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.dlg_list>

when you received bye on secondary server, is that server
recognized the bye.

    On Sun, Jul 18, 2021 at 3:38 AM Social Boh mailto:soc...@bohboh.info>> wrote:

Hello,

if during a call I stop Primary Kamailio the Secondary
Kamailio take de Virtual IP but don't save call data:

In the Primary database, acc table, I have INVITE data but not
BYE data; no data in acc_cdrs for the call. In the Secondary
Kamailio, acc table, I have BYE data but not INVITE data.

in the file LOG no data about call cdr.

Maybe a solution is write call data whene the call end?

Regards

---

I'm SoCIaL, MayBe

El 17/07/2021 a las 11:34 a. m., M Arqum CH escribió:

Make sure that your table columns matched the field's you
are trying to insert, please check the syslog for acc
module log, acc module print the cdr in log facility too,
if acc module facing any issue while writing cdr in db, it
will throw error in logs.

Hope this will help you.

On Sat, Jul 17, 2021, 7:33 PM Social Boh
mailto:soc...@bohboh.info>> wrote:

Hello,

i tested with DMQ module and happens the same.

DMQ module loaded before dialog module and on dialog
module parameters:

modparam("dialog", "enable_dmq", 1)

During a call I can see on both Kamailio servers the
dialog data but when switch off the active kamailio
server and the second kamailio server take the control
and hangup the call, no data acc_cdrs table.

Regards

---

I'm SoCIaL, MayBe

El 17/07/2021 a las 1:24 a. m., M Arqum CH escribió:

Hi Social,

Replication of dialog using dmq will solve
your problem, you need to load dmq module before 
loading dialog module and enable replication of
dialogs in dialog module, please check dialog
module conf option for more details.

    Cheers

On Fri, Jul 16, 2021, 7:32 PM Social Boh
mailto:soc...@bohboh.info>>
wrote:

Im' testinf Kamailio HA with Keepalived.

I have a problem with acc_cdrs table in this
situation:

A call B and when the two users are in the
call I switch off the active
Kamailio. Keepalived pass the Virtual IP to
the secondary Kamailio and
audio still work because I'm using a remote
RTPENGINE server for both
Kamailio.

The problem is when the call end. On the acc
   

Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record

[Social Boh]

Hi,

I'm using 5.4.6

I switch off Kamailio on the first server and keepalived move the 
virtual IP on the second server so First Kamailio is off. This during 
the call.


Before switch off Kamailio and during the call I can see data dialog on 
both server but in the LOG /var/log/messages in my case:


server 1 INVITE

server 2 BYE

acc_cdrs table empty on both servers.

acc table first server INVITE; acc table second server BYE

Regards

---
I'm SoCIaL, MayBe

El 18/07/2021 a las 2:33 p. m., M Arqum CH escribió:

Hi SoCIal,

which Kamailio version you are using.

did you able to get the dialog information using the dialog command. 
https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.dlg_list 
<https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.dlg_list>


when you received bye on secondary server, is that server recognized 
the bye.





On Sun, Jul 18, 2021 at 3:38 AM Social Boh <mailto:soc...@bohboh.info>> wrote:


Hello,

if during a call I stop Primary Kamailio the Secondary Kamailio
take de Virtual IP but don't save call data:

In the Primary database, acc table, I have INVITE data but not BYE
data; no data in acc_cdrs for the call. In the Secondary Kamailio,
acc table, I have BYE data but not INVITE data.

in the file LOG no data about call cdr.

Maybe a solution is write call data whene the call end?

Regards

---
I'm SoCIaL, MayBe

El 17/07/2021 a las 11:34 a. m., M Arqum CH escribió:

Make sure that your table columns matched the field's you are
trying to insert, please check the syslog for acc module log, acc
module print the cdr in log facility too, if acc module facing
any issue while writing cdr in db, it will throw error in logs.

Hope this will help you.

    On Sat, Jul 17, 2021, 7:33 PM Social Boh mailto:soc...@bohboh.info>> wrote:

Hello,

i tested with DMQ module and happens the same.

DMQ module loaded before dialog module and on dialog module
parameters:

modparam("dialog", "enable_dmq", 1)

During a call I can see on both Kamailio servers the dialog
data but when switch off the active kamailio server and the
second kamailio server take the control and hangup the call,
no data acc_cdrs table.

Regards

---
I'm SoCIaL, MayBe

El 17/07/2021 a las 1:24 a. m., M Arqum CH escribió:

Hi Social,

Replication of dialog using dmq will solve your problem, you
need to load dmq module before  loading dialog module and
enable replication of dialogs in dialog module, please check
dialog module conf option for more details.

Cheers


On Fri, Jul 16, 2021, 7:32 PM Social Boh mailto:soc...@bohboh.info>> wrote:

Im' testinf Kamailio HA with Keepalived.

I have a problem with acc_cdrs table in this situation:

A call B and when the two users are in the call I switch
off the active
Kamailio. Keepalived pass the Virtual IP to the
secondary Kamailio and
audio still work because I'm using a remote RTPENGINE
server for both
Kamailio.

The problem is when the call end. On the acc table i can
see INVITE and
BYE entries but on the acc_cdrs table nothing. Second
problem the dialog
still present on the table dialog in a Primary/Primary
replication.

Is there a way to solve it?

Thank you

Regards

-- 
---

I'm SoCIaL, MayBe


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Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record

[Social Boh]

Hello,

if during a call I stop Primary Kamailio the Secondary Kamailio take de 
Virtual IP but don't save call data:


In the Primary database, acc table, I have INVITE data but not BYE data; 
no data in acc_cdrs for the call. In the Secondary Kamailio, acc table, 
I have BYE data but not INVITE data.


in the file LOG no data about call cdr.

Maybe a solution is write call data whene the call end?

Regards

---
I'm SoCIaL, MayBe

El 17/07/2021 a las 11:34 a. m., M Arqum CH escribió:
Make sure that your table columns matched the field's you are trying 
to insert, please check the syslog for acc module log, acc module 
print the cdr in log facility too, if acc module facing any issue 
while writing cdr in db, it will throw error in logs.


Hope this will help you.

On Sat, Jul 17, 2021, 7:33 PM Social Boh <mailto:soc...@bohboh.info>> wrote:


Hello,

i tested with DMQ module and happens the same.

DMQ module loaded before dialog module and on dialog module
parameters:

modparam("dialog", "enable_dmq", 1)

During a call I can see on both Kamailio servers the dialog data
but when switch off the active kamailio server and the second
kamailio server take the control and hangup the call, no data
acc_cdrs table.

Regards

---
I'm SoCIaL, MayBe

El 17/07/2021 a las 1:24 a. m., M Arqum CH escribió:

Hi Social,

Replication of dialog using dmq will solve your problem, you need
to load dmq module before  loading dialog module and enable
replication of dialogs in dialog module, please check dialog
module conf option for more details.

Cheers


On Fri, Jul 16, 2021, 7:32 PM Social Boh mailto:soc...@bohboh.info>> wrote:

Im' testinf Kamailio HA with Keepalived.

I have a problem with acc_cdrs table in this situation:

A call B and when the two users are in the call I switch off
the active
Kamailio. Keepalived pass the Virtual IP to the secondary
Kamailio and
audio still work because I'm using a remote RTPENGINE server
for both
Kamailio.

The problem is when the call end. On the acc table i can see
INVITE and
BYE entries but on the acc_cdrs table nothing. Second problem
the dialog
still present on the table dialog in a Primary/Primary
replication.

Is there a way to solve it?

Thank you

Regards

-- 
---

I'm SoCIaL, MayBe


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Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record

[Social Boh]

Hello,

i tested with DMQ module and happens the same.

DMQ module loaded before dialog module and on dialog module parameters:

modparam("dialog", "enable_dmq", 1)

During a call I can see on both Kamailio servers the dialog data but 
when switch off the active kamailio server and the second kamailio 
server take the control and hangup the call, no data acc_cdrs table.


Regards

---
I'm SoCIaL, MayBe

El 17/07/2021 a las 1:24 a. m., M Arqum CH escribió:

Hi Social,

Replication of dialog using dmq will solve your problem, you need to 
load dmq module before  loading dialog module and enable replication 
of dialogs in dialog module, please check dialog module conf option 
for more details.


Cheers


On Fri, Jul 16, 2021, 7:32 PM Social Boh <mailto:soc...@bohboh.info>> wrote:


Im' testinf Kamailio HA with Keepalived.

I have a problem with acc_cdrs table in this situation:

A call B and when the two users are in the call I switch off the
active
Kamailio. Keepalived pass the Virtual IP to the secondary Kamailio
and
audio still work because I'm using a remote RTPENGINE server for both
Kamailio.

The problem is when the call end. On the acc table i can see
INVITE and
BYE entries but on the acc_cdrs table nothing. Second problem the
dialog
still present on the table dialog in a Primary/Primary replication.

Is there a way to solve it?

Thank you

Regards

-- 
---

I'm SoCIaL, MayBe


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[SR-Users] Kamailio KeepAlived HA acc_cdrs table record

[Social Boh]

Im' testinf Kamailio HA with Keepalived.

I have a problem with acc_cdrs table in this situation:

A call B and when the two users are in the call I switch off the active 
Kamailio. Keepalived pass the Virtual IP to the secondary Kamailio and 
audio still work because I'm using a remote RTPENGINE server for both 
Kamailio.


The problem is when the call end. On the acc table i can see INVITE and 
BYE entries but on the acc_cdrs table nothing. Second problem the dialog 
still present on the table dialog in a Primary/Primary replication.


Is there a way to solve it?

Thank you

Regards

--
---
I'm SoCIaL, MayBe


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Re: [SR-Users] DMQ_USRLOC module and calls

[Social Boh]

Hello Alex,

thank you for your answer... now I understand a little bit more.

Thank you

Reagrds

---
I'm SoCIaL, MayBe

El 13/07/2021 a las 3:40 p. m., Alex Balashov escribió:
No, you don’t need to, but it is a vastly superior replication 
mechanism to database replication. It was created in order to get 
around some of the common technical and architectural limitations of 
the database replication pattern.


—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 4:37 PM, Social Boh  wrote:



Hello,

 in an active/stanby kamailio setup with keepalived I'm using 
Database replication primary/primary so I don't need DMQ_USRLOC


I'm testing just now and when active kamailio go down, I can make 
calls between extensions without connection problems with the stanby 
kamailio.


So my question is: where I "NEED" to use DMQ_USRLOC?

Regards

---
I'm SoCIaL, MayBe
El 13/07/2021 a las 9:48 a. m., Barry Flanagan escribió:

On 13/07/2021 15:41, Social Boh wrote:

Hello,

If I have to use path protocol or other routing logic to share 
REGISTER between the Two Kamailio so I can call From USERA on 
KamailioA to USERB on KamailioB, I think I don't need to still use 
DMQ_USRLOC module.


Can you offer a practical use of this module, please?



For example, if you had an active/standby kamailio setup with a 
floating IP managed by e.g. Keepalived then when a failover event 
occurs, the usrloc is up to date and the failover is completely 
transparent.


Hope this helps

-Barry Flanagan



Thank you

Regards

---
I'm SoCIaL, MayBe

El 13/07/2021 a las 9:01 a. m., Alex Balashov escribió:
To replicate knowledge of the registration contact bindings 
automatically.


It doesn’t provide magic routing to reach them. It just shares the 
knowledge. :-)


—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 9:46 AM, Social Boh  wrote:

If I have to use Path or other solution, which is the main idea 
behind the DMQ_USRLOC module?


Regards

---
I'm SoCIaL, MayBe


El 13/07/2021 a las 8:36 a. m., Alex Balashov escribió:
Yes, you need the Path module and special logic for lateral 
routing of requests received from A and transiting through a 
Path hop of B.


—
Sent from mobile, with due apologies for brevity and errors.

On Jul 13, 2021, at 8:47 AM, Social Boh  
wrote:

Hello,

I'm testing DMQ_USRLOC modulo between two Kamailio servers (A) 
and (B)


The two servers share the same domain and using Amazon Route53 
to distributes 50% requests each Server.


The problem is when I try to make a call from user A 
(registered on kamailio A) and the user B (registered on 
Kamailio B).


The call follow this flow:

User A --> KamailioA --> UserB

go directly to userB because KamailioA know, via DMQ_USRLOC IP 
and port of UserB. With some Softphone the call work with audio 
without problem; with other Softphone USERB never answer the 
INVITE sends from Kamailio. Is it possible use this flow?


USER A --> KamailioA --> KamailioB --> User B

to resolve this kind of problem or is there other available 
option?


Thank you

--
---
I'm SoCIaL, MayBe


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Re: [SR-Users] DMQ_USRLOC module and calls

[Social Boh]

Hello,

 in an active/stanby kamailio setup with keepalived I'm using Database 
replication primary/primary so I don't need DMQ_USRLOC


I'm testing just now and when active kamailio go down, I can make calls 
between extensions without connection problems with the stanby kamailio.


So my question is: where I "NEED" to use DMQ_USRLOC?

Regards

---
I'm SoCIaL, MayBe

El 13/07/2021 a las 9:48 a. m., Barry Flanagan escribió:

On 13/07/2021 15:41, Social Boh wrote:

Hello,

If I have to use path protocol or other routing logic to share 
REGISTER between the Two Kamailio so I can call From USERA on 
KamailioA to USERB on KamailioB, I think I don't need to still use 
DMQ_USRLOC module.


Can you offer a practical use of this module, please?



For example, if you had an active/standby kamailio setup with a 
floating IP managed by e.g. Keepalived then when a failover event 
occurs, the usrloc is up to date and the failover is completely 
transparent.


Hope this helps

-Barry Flanagan



Thank you

Regards

---
I'm SoCIaL, MayBe

El 13/07/2021 a las 9:01 a. m., Alex Balashov escribió:
To replicate knowledge of the registration contact bindings 
automatically.


It doesn’t provide magic routing to reach them. It just shares the 
knowledge. :-)


—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 9:46 AM, Social Boh  wrote:

If I have to use Path or other solution, which is the main idea 
behind the DMQ_USRLOC module?


Regards

---
I'm SoCIaL, MayBe


El 13/07/2021 a las 8:36 a. m., Alex Balashov escribió:
Yes, you need the Path module and special logic for lateral 
routing of requests received from A and transiting through a Path 
hop of B.


—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 8:47 AM, Social Boh  wrote:

Hello,

I'm testing DMQ_USRLOC modulo between two Kamailio servers (A) 
and (B)


The two servers share the same domain and using Amazon Route53 to 
distributes 50% requests each Server.


The problem is when I try to make a call from user A (registered 
on kamailio A) and the user B (registered on Kamailio B).


The call follow this flow:

User A --> KamailioA --> UserB

go directly to userB because KamailioA know, via DMQ_USRLOC IP 
and port of UserB. With some Softphone the call work with audio 
without problem; with other Softphone USERB never answer the 
INVITE sends from Kamailio. Is it possible use this flow?


USER A --> KamailioA --> KamailioB --> User B

to resolve this kind of problem or is there other available option?

Thank you

--
---
I'm SoCIaL, MayBe


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Re: [SR-Users] DMQ_USRLOC module and calls

[Social Boh]

Sorry,

I explained myself wrong. I know how using PATH and what PATH do.

Thank you

Regards

---
I'm SoCIaL, MayBe

El 13/07/2021 a las 10:02 a. m., Alex Balashov escribió:

Path doesn’t share registrations. It just allows routing from B back through A 
in order to reach the endpoint, rather than trying to reach the endpoint 
directly. It’s just a header that says “when contacting this registrant, 
hop/detour through the original registrant”.

But, Path doesn’t do anything else. So, if you want registrar B to know that 
the registrant is registered anywhere at all — A or B — and the Contact/URI at 
which to reach it (whether directly or through a Path hop), you need some means 
of replicating the registration database. dmq_usrloc provides this aspect.

Path and dmq_usrloc do entirely different things, of different degrees of 
significance to the overall process. The “heavy” part is done by 
usrloc/dmq_usrloc.

—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 10:43 AM, Social Boh  wrote:

Hello,

If I have to use path protocol or other routing logic to share REGISTER between 
the Two Kamailio so I can call From USERA on KamailioA to USERB on KamailioB, I 
think I don't need to still use DMQ_USRLOC module.

Can you offer a practical use of this module, please?

Thank you

Regards

---
I'm SoCIaL, MayBe


El 13/07/2021 a las 9:01 a. m., Alex Balashov escribió:
To replicate knowledge of the registration contact bindings automatically.

It doesn’t provide magic routing to reach them. It just shares the knowledge. 
:-)

—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 9:46 AM, Social Boh  wrote:

If I have to use Path or other solution, which is the main idea behind the 
DMQ_USRLOC module?

Regards

---
I'm SoCIaL, MayBe


El 13/07/2021 a las 8:36 a. m., Alex Balashov escribió:
Yes, you need the Path module and special logic for lateral routing of requests 
received from A and transiting through a Path hop of B.

—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 8:47 AM, Social Boh  wrote:

Hello,

I'm testing DMQ_USRLOC modulo between two Kamailio servers (A) and (B)

The two servers share the same domain and using Amazon Route53 to distributes 
50% requests each Server.

The problem is when I try to make a call from user A (registered on kamailio A) 
and the user B (registered on Kamailio B).

The call follow this flow:

User A --> KamailioA --> UserB

go directly to userB because KamailioA know, via DMQ_USRLOC IP and port of 
UserB. With some Softphone the call work with audio without problem; with other 
Softphone USERB never answer the INVITE sends from Kamailio. Is it possible use 
this flow?

USER A --> KamailioA --> KamailioB --> User B

to resolve this kind of problem or is there other available option?

Thank you

--
---
I'm SoCIaL, MayBe


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Re: [SR-Users] DMQ_USRLOC module and calls

[Social Boh]

Thank you Barry,

I'll try.

Regards

---
I'm SoCIaL, MayBe

El 13/07/2021 a las 9:48 a. m., Barry Flanagan escribió:

On 13/07/2021 15:41, Social Boh wrote:

Hello,

If I have to use path protocol or other routing logic to share 
REGISTER between the Two Kamailio so I can call From USERA on 
KamailioA to USERB on KamailioB, I think I don't need to still use 
DMQ_USRLOC module.


Can you offer a practical use of this module, please?



For example, if you had an active/standby kamailio setup with a 
floating IP managed by e.g. Keepalived then when a failover event 
occurs, the usrloc is up to date and the failover is completely 
transparent.


Hope this helps

-Barry Flanagan



Thank you

Regards

---
I'm SoCIaL, MayBe

El 13/07/2021 a las 9:01 a. m., Alex Balashov escribió:
To replicate knowledge of the registration contact bindings 
automatically.


It doesn’t provide magic routing to reach them. It just shares the 
knowledge. :-)


—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 9:46 AM, Social Boh  wrote:

If I have to use Path or other solution, which is the main idea 
behind the DMQ_USRLOC module?


Regards

---
I'm SoCIaL, MayBe


El 13/07/2021 a las 8:36 a. m., Alex Balashov escribió:
Yes, you need the Path module and special logic for lateral 
routing of requests received from A and transiting through a Path 
hop of B.


—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 8:47 AM, Social Boh  wrote:

Hello,

I'm testing DMQ_USRLOC modulo between two Kamailio servers (A) 
and (B)


The two servers share the same domain and using Amazon Route53 to 
distributes 50% requests each Server.


The problem is when I try to make a call from user A (registered 
on kamailio A) and the user B (registered on Kamailio B).


The call follow this flow:

User A --> KamailioA --> UserB

go directly to userB because KamailioA know, via DMQ_USRLOC IP 
and port of UserB. With some Softphone the call work with audio 
without problem; with other Softphone USERB never answer the 
INVITE sends from Kamailio. Is it possible use this flow?


USER A --> KamailioA --> KamailioB --> User B

to resolve this kind of problem or is there other available option?

Thank you

--
---
I'm SoCIaL, MayBe


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Re: [SR-Users] DMQ_USRLOC module and calls

[Social Boh]

Hello,

If I have to use path protocol or other routing logic to share REGISTER 
between the Two Kamailio so I can call From USERA on KamailioA to USERB 
on KamailioB, I think I don't need to still use DMQ_USRLOC module.


Can you offer a practical use of this module, please?

Thank you

Regards

---
I'm SoCIaL, MayBe

El 13/07/2021 a las 9:01 a. m., Alex Balashov escribió:

To replicate knowledge of the registration contact bindings automatically.

It doesn’t provide magic routing to reach them. It just shares the knowledge. 
:-)

—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 9:46 AM, Social Boh  wrote:

If I have to use Path or other solution, which is the main idea behind the 
DMQ_USRLOC module?

Regards

---
I'm SoCIaL, MayBe


El 13/07/2021 a las 8:36 a. m., Alex Balashov escribió:
Yes, you need the Path module and special logic for lateral routing of requests 
received from A and transiting through a Path hop of B.

—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 8:47 AM, Social Boh  wrote:

Hello,

I'm testing DMQ_USRLOC modulo between two Kamailio servers (A) and (B)

The two servers share the same domain and using Amazon Route53 to distributes 
50% requests each Server.

The problem is when I try to make a call from user A (registered on kamailio A) 
and the user B (registered on Kamailio B).

The call follow this flow:

User A --> KamailioA --> UserB

go directly to userB because KamailioA know, via DMQ_USRLOC IP and port of 
UserB. With some Softphone the call work with audio without problem; with other 
Softphone USERB never answer the INVITE sends from Kamailio. Is it possible use 
this flow?

USER A --> KamailioA --> KamailioB --> User B

to resolve this kind of problem or is there other available option?

Thank you

--
---
I'm SoCIaL, MayBe


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Re: [SR-Users] DMQ_USRLOC module and calls

[Social Boh]

If I have to use Path or other solution, which is the main idea behind 
the DMQ_USRLOC module?


Regards

---
I'm SoCIaL, MayBe

El 13/07/2021 a las 8:36 a. m., Alex Balashov escribió:

Yes, you need the Path module and special logic for lateral routing of requests 
received from A and transiting through a Path hop of B.

—
Sent from mobile, with due apologies for brevity and errors.


On Jul 13, 2021, at 8:47 AM, Social Boh  wrote:

Hello,

I'm testing DMQ_USRLOC modulo between two Kamailio servers (A) and (B)

The two servers share the same domain and using Amazon Route53 to distributes 
50% requests each Server.

The problem is when I try to make a call from user A (registered on kamailio A) 
and the user B (registered on Kamailio B).

The call follow this flow:

User A --> KamailioA --> UserB

go directly to userB because KamailioA know, via DMQ_USRLOC IP and port of 
UserB. With some Softphone the call work with audio without problem; with other 
Softphone USERB never answer the INVITE sends from Kamailio. Is it possible use 
this flow?

USER A --> KamailioA --> KamailioB --> User B

to resolve this kind of problem or is there other available option?

Thank you

--
---
I'm SoCIaL, MayBe


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Re: [SR-Users] DMQ_USRLOC module and calls

[Social Boh]

Thank you Olle to take time to answer my question.

So the solution is send the INVITE from the same Kamailio where the 
device registered.


The question now is: How Can I do that taking advantage of DMQ_USRLOC 
module?


Thank you

Regards

---
I'm SoCIaL, MayBe

El 13/07/2021 a las 8:04 a. m., Olle E. Johansson escribió:



On 13 Jul 2021, at 14:46, Social Boh  wrote:

Hello,

I'm testing DMQ_USRLOC modulo between two Kamailio servers (A) and (B)

The two servers share the same domain and using Amazon Route53 to distributes 
50% requests each Server.

The problem is when I try to make a call from user A (registered on kamailio A) 
and the user B (registered on Kamailio B).

The call follow this flow:

User A --> KamailioA --> UserB

go directly to userB because KamailioA know, via DMQ_USRLOC IP and port of 
UserB. With some Softphone the call work with audio without problem; with other 
Softphone USERB never answer the INVITE sends from Kamailio. Is it possible use 
this flow?

USER A --> KamailioA --> KamailioB --> User B

to resolve this kind of problem or is there other available option?

If the softphone is behind NAT it has a relation to one IP address in the NAT. 
You need to send the INVITE from that IP and no other IP in order to reach the 
device. Even without NAT, many devices just don’t respond to messages from 
unknown IP addresses.

/O
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[SR-Users] DMQ_USRLOC module and calls

[Social Boh]

Hello,

I'm testing DMQ_USRLOC modulo between two Kamailio servers (A) and (B)

The two servers share the same domain and using Amazon Route53 to 
distributes 50% requests each Server.


The problem is when I try to make a call from user A (registered on 
kamailio A) and the user B (registered on Kamailio B).


The call follow this flow:

User A --> KamailioA --> UserB

go directly to userB because KamailioA know, via DMQ_USRLOC IP and port 
of UserB. With some Softphone the call work with audio without problem; 
with other Softphone USERB never answer the INVITE sends from Kamailio. 
Is it possible use this flow?


USER A --> KamailioA --> KamailioB --> User B

to resolve this kind of problem or is there other available option?

Thank you

--
---
I'm SoCIaL, MayBe


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Re: [SR-Users] Kamailio SBC+ Freeswitch as media server

[Social Boh]

*2021-06-08 10:35:19.015670 [WARNING] sofia.c:10569 IP 192.168.1.15 
Rejected by acl "domains"*


*You have to authorize this IP 192.168.1.15 on FreeSWITCH ACL, normally:*

*acl.conf.xml*

*Regards
*

---
I'm SoCIaL, MayBe

El 8/06/2021 a las 2:38 a. m., John Tuxies escribió:
*2021-06-08 10:35:19.015670 [WARNING] sofia.c:10569 IP 192.168.1.15 
Rejected by acl "domains"*
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Re: [SR-Users] DLGLOAD stuck in Dispatcher Module(HA Mode)

[Social Boh]

Maybe a option is leave the secondary like primary when primary go up newly.

Example: Primary go down, Secondary is the new Primary. Primary came 
back up but Secondary still is the VIP and still process the calls.


Only if Secondary go down, primary is the new VIP.

The keepalived parameter for this behaviour is:

*nopreempt*

---
I'm SoCIaL, MayBe

El 16/12/2020 a las 11:57 a. m., harneet singh escribió:

Hi All,

I am using Kamailio in HA mode with Keepalived providing the 
VIP(Virtual IP) functionality, and have run into a rather peculiar issue.


_Setup:_

Caller  KamailioVIP(*Primary *and *Secondary* 
Kamailio)--- Callee


    HA provided by Keepalived
    DMQ is used between the Primary and Secondary Kamailio for dialog sync

*/_Issue Seen:_/*
*/_
_/*
*/_Suppose the Primary Kamailio has been brought down_/* and the 
Secondary one is actually active and tied to the VIP.


When a call is fired from the Caller, it traverses through the 
Secondary Kamailio and lands onto the callee. The dialogs are updated 
properly. At this point, the Primary Kamailio is brought up, and 
dialog state is synced due to the DMQ module.
The Keepalived will now attach the VIP to the Primary Instance. If the 
caller hangs up the phone at this point, the BYE sip message traverses 
through the Primary Kamailio instance to the callee and the call is 
cleared, but there are two issues here.


  1.  The Primary Kamailio throws an error in *ds_load_remove() 
*saying that it cannot find the load for the specific call id. This is 
apparently due to the fact that the dispatcher data is not synced 
between the two modules but dialog data is. So dialog wise, things are 
fine. Can this be fixed somehow?


  2. The above is still as grave a problem as the dialogs are cleared. 
BUT, if we check the '*kamcmd dispatcher.list*' on the *SECONDARY* 
kamailio even well after the call is cleared, the *DLGLOAD* shows 1. 
Since we are using the *Call Load based distribution* for the 
dispatching, this is effectively one call stuck on the dispatcher, 
which leads to resource leak.


Is this a known issue, and if so, do we have a way to circumvent this?

Regards,
Harneet Singh
--
"Once you eliminate the impossible, whatever remains, no matter how 
improbable, must be the truth" - Sir Arthur Conan Doyle


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Re: [SR-Users] Advertise concept

[Social Boh]

Sorry for private message, Fred

Thank you for your answer.


You could use it for all communication, regardless of method... so if
behind NAT, it would certainly make sense to use advertise for
registers.


you means outbound register because behind NAT kamailio can't accept 
inbound register over public IP, right?


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[SR-Users] Advertise concept

[Social Boh]

Hello,

maybe i don't understood completely the use of Advertise parameter.

We use it only if Kamailio is behind a NAT and Kamailio have to 
comunicate over is PublicIP with provider, other SIP proxy and PBX but 
never to accept Users REGISTER.


Is it right?

Regards

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Re: [SR-Users] SECFILTER reload from Siremis

[Social Boh]

thank you. I'll check soon.

Regards

---
I'm SoCIaL, MayBe

On 24/06/20 3:57, Daniel-Constantin Mierla wrote:

Hello,

for the records, jsonrpc command panel has to be installed/configured:

   - http://kb.asipto.com/siremis:install53x:jsonrpc-commands

Cheers,
Daniel

On 12.06.20 21:32, Henning Westerholt wrote:

Hello,

look for the option in the GUI to execute arbitrary RPC commands. You might 
need to configure it before, if it gives only an error.

Cheers,

Henning

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Kamailio services – https://gilawa.com

-Original Message-
From: sr-users  On Behalf Of Social Boh
Sent: Friday, June 12, 2020 9:14 PM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] SECFILTER reload from Siremis

Hello list,

is posible to reload SECFILTER data directly from Siremis?

or the only way is:

kamcmd secfilter.reload

Regards

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[SR-Users] SECFILTER reload from Siremis

[Social Boh]

Hello list,

is posible to reload SECFILTER data directly from Siremis?

or the only way is:

kamcmd secfilter.reload

Regards

--
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Re: [SR-Users] Homer 7 and Kamailio

[Social Boh]

sorry,

with:

modparam("siptrace", "trace_mode", 1)

I have duplicated messages on Homer.

Regards

---
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On 8/06/20 23:26, Social Boh wrote:


Hello,

this configuration works for me:

# - siptrace params
loadmodule "siptrace.so"
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "trace_mode", 0)
modparam("siptrace", "duplicate_uri", "sip:1.2.3.4:9060")
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "hep_mode_on", 1)
modparam("siptrace", "hep_version", 3)
modparam("siptrace", "hep_capture_id", 234)

and:

request_route {

       sip_trace_mode("t");

with:

modparam("siptrace", "trace_mode", 0)

I have duplicated messages on Homer. My homer configuration is:

HEPAddr   = "0.0.0.0:9060"
HEPTCPAddr    = "0.0.0.0:9060 "
HEPTLSAddr    = ""

Thank you

 Regards

---
I'm SoCIaL, MayBe
On 8/06/20 14:49, Karsten Horsmann wrote:

Hi,

The Kamailio version should be interesting. And in 5.3.4 things 
changed. See for example my config snippet that works for me and what 
Henning told then


https://lists.kamailio.org/pipermail/sr-users/2020-June/109447.html



Social Boh mailto:soc...@bohboh.info>> schrieb 
am Mo., 8. Juni 2020, 21:20:


Hello,

I'm trying to using homer 7 and Kamaikio Siptrace module.

My kamailio configuration:

# - siptrace params
loadmodule "siptrace.so"
modparam("siptrace", "db_url", DBURL)
modparam("siptrace", "trace_flag",22)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "duplicate_uri", "sip:1.2.3.4:9060")
modparam("siptrace", "trace_to_database", 1)
modparam("siptrace", "hep_mode_on", 1)
modparam("siptrace", "hep_version", 2)
modparam("siptrace", "hep_capture_id", 234)

request_route {

       sip_trace();
       setflag(22);

But on Homer 7 LOG:

...

2020-06-08T14:12:33-05:00 WARN malformed packet with length 754
which is neither hep nor protobuf encapsulated
2020-06-08T14:12:37-05:00 WARN malformed packet with length 525
which is neither hep nor protobuf encapsulated
2020-06-08T14:12:48-05:00 WARN malformed packet with length 599
which is neither hep nor protobuf encapsulated

    ...

If I using:

modparam("siptrace", "hep_version", 3)

No captured data on Homer SIP Capture Server.

Regards

---
I'm SoCIaL, MayBe

On 8/06/20 13:06, Karsten Horsmann wrote:

Hi,

short answer - yes it does.

You can capture the traffic within Kamailio (siptrace module) or
did that (if the traffic is plain sip) with 3rd party stuff like
heplify.

Cheers
Karsten

Social Boh mailto:soc...@bohboh.info>>
schrieb am Mo., 8. Juni 2020, 19:25:

Hello,

Does Kamailio works with homer sip capture server 7 and HEPv3?

I'm trying to test without success.

Regards

-- 
---

I'm SoCIaL, MayBe

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Re: [SR-Users] Homer 7 and Kamailio

[Social Boh]

Hello,

this configuration works for me:

# - siptrace params
loadmodule "siptrace.so"
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "trace_mode", 0)
modparam("siptrace", "duplicate_uri", "sip:1.2.3.4:9060")
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "hep_mode_on", 1)
modparam("siptrace", "hep_version", 3)
modparam("siptrace", "hep_capture_id", 234)

and:

request_route {

       sip_trace_mode("t");

with:

modparam("siptrace", "trace_mode", 0)

I have duplicated messages on Homer. My homer configuration is:

HEPAddr   = "0.0.0.0:9060"
HEPTCPAddr    = "0.0.0.0:9060 "
HEPTLSAddr    = ""

Thank you

 Regards

---
I'm SoCIaL, MayBe

On 8/06/20 14:49, Karsten Horsmann wrote:

Hi,

The Kamailio version should be interesting. And in 5.3.4 things 
changed. See for example my config snippet that works for me and what 
Henning told then


https://lists.kamailio.org/pipermail/sr-users/2020-June/109447.html



Social Boh mailto:soc...@bohboh.info>> schrieb am 
Mo., 8. Juni 2020, 21:20:


Hello,

I'm trying to using homer 7 and Kamaikio Siptrace module.

My kamailio configuration:

# - siptrace params
loadmodule "siptrace.so"
modparam("siptrace", "db_url", DBURL)
modparam("siptrace", "trace_flag",22)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "duplicate_uri", "sip:1.2.3.4:9060")
modparam("siptrace", "trace_to_database", 1)
modparam("siptrace", "hep_mode_on", 1)
modparam("siptrace", "hep_version", 2)
modparam("siptrace", "hep_capture_id", 234)

request_route {

       sip_trace();
       setflag(22);

But on Homer 7 LOG:

...

2020-06-08T14:12:33-05:00 WARN malformed packet with length 754
which is neither hep nor protobuf encapsulated
2020-06-08T14:12:37-05:00 WARN malformed packet with length 525
which is neither hep nor protobuf encapsulated
2020-06-08T14:12:48-05:00 WARN malformed packet with length 599
which is neither hep nor protobuf encapsulated

...

If I using:

    modparam("siptrace", "hep_version", 3)

No captured data on Homer SIP Capture Server.

Regards

---
I'm SoCIaL, MayBe

On 8/06/20 13:06, Karsten Horsmann wrote:

Hi,

short answer - yes it does.

You can capture the traffic within Kamailio (siptrace module) or
did that (if the traffic is plain sip) with 3rd party stuff like
heplify.

Cheers
Karsten

Social Boh mailto:soc...@bohboh.info>>
schrieb am Mo., 8. Juni 2020, 19:25:

Hello,

Does Kamailio works with homer sip capture server 7 and HEPv3?

I'm trying to test without success.

Regards

-- 
---

I'm SoCIaL, MayBe

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Re: [SR-Users] Homer 7 and Kamailio

[Social Boh]

Hello,

My Kamailio version is 5.3.4. Hepv3 is only TCP? The traffic is sended 
over TCP?


Regards

---
I'm SoCIaL, MayBe

On 8/06/20 14:49, Karsten Horsmann wrote:

Hi,

The Kamailio version should be interesting. And in 5.3.4 things 
changed. See for example my config snippet that works for me and what 
Henning told then


https://lists.kamailio.org/pipermail/sr-users/2020-June/109447.html



Social Boh mailto:soc...@bohboh.info>> schrieb am 
Mo., 8. Juni 2020, 21:20:


Hello,

I'm trying to using homer 7 and Kamaikio Siptrace module.

My kamailio configuration:

# - siptrace params
loadmodule "siptrace.so"
modparam("siptrace", "db_url", DBURL)
modparam("siptrace", "trace_flag",22)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "duplicate_uri", "sip:1.2.3.4:9060")
modparam("siptrace", "trace_to_database", 1)
modparam("siptrace", "hep_mode_on", 1)
modparam("siptrace", "hep_version", 2)
modparam("siptrace", "hep_capture_id", 234)

request_route {

       sip_trace();
       setflag(22);

But on Homer 7 LOG:

...

2020-06-08T14:12:33-05:00 WARN malformed packet with length 754
which is neither hep nor protobuf encapsulated
2020-06-08T14:12:37-05:00 WARN malformed packet with length 525
which is neither hep nor protobuf encapsulated
2020-06-08T14:12:48-05:00 WARN malformed packet with length 599
which is neither hep nor protobuf encapsulated

...

If I using:

modparam("siptrace", "hep_version", 3)

No captured data on Homer SIP Capture Server.

Regards

---
I'm SoCIaL, MayBe

On 8/06/20 13:06, Karsten Horsmann wrote:

    Hi,

    short answer - yes it does.

You can capture the traffic within Kamailio (siptrace module) or
did that (if the traffic is plain sip) with 3rd party stuff like
heplify.

Cheers
Karsten

Social Boh mailto:soc...@bohboh.info>>
schrieb am Mo., 8. Juni 2020, 19:25:

Hello,

Does Kamailio works with homer sip capture server 7 and HEPv3?

I'm trying to test without success.

Regards

-- 
---

I'm SoCIaL, MayBe

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Re: [SR-Users] Homer 7 and Kamailio

[Social Boh]

Hello,

I'm trying to using homer 7 and Kamaikio Siptrace module.

My kamailio configuration:

# - siptrace params
loadmodule "siptrace.so"
modparam("siptrace", "db_url", DBURL)
modparam("siptrace", "trace_flag",22)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "duplicate_uri", "sip:1.2.3.4:9060")
modparam("siptrace", "trace_to_database", 1)
modparam("siptrace", "hep_mode_on", 1)
modparam("siptrace", "hep_version", 2)
modparam("siptrace", "hep_capture_id", 234)

request_route {

       sip_trace();
       setflag(22);

But on Homer 7 LOG:

...

2020-06-08T14:12:33-05:00 WARN malformed packet with length 754 which is 
neither hep nor protobuf encapsulated
2020-06-08T14:12:37-05:00 WARN malformed packet with length 525 which is 
neither hep nor protobuf encapsulated
2020-06-08T14:12:48-05:00 WARN malformed packet with length 599 which is 
neither hep nor protobuf encapsulated


...

If I using:

modparam("siptrace", "hep_version", 3)

No captured data on Homer SIP Capture Server.

Regards

---
I'm SoCIaL, MayBe

On 8/06/20 13:06, Karsten Horsmann wrote:

Hi,

short answer - yes it does.

You can capture the traffic within Kamailio (siptrace module) or did 
that (if the traffic is plain sip) with 3rd party stuff like heplify.


Cheers
Karsten

Social Boh mailto:soc...@bohboh.info>> schrieb am 
Mo., 8. Juni 2020, 19:25:


Hello,

Does Kamailio works with homer sip capture server 7 and HEPv3?

I'm trying to test without success.

Regards

-- 
---

I'm SoCIaL, MayBe

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[SR-Users] Homer 7 and Kamailio

[Social Boh]

Hello,

Does Kamailio works with homer sip capture server 7 and HEPv3?

I'm trying to test without success.

Regards

--
---
I'm SoCIaL, MayBe

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Re: [SR-Users] mid-registrar example

[Social Boh]

wrong list

---
I'm SoCIaL, MayBe

On 5/15/20 16:50, Easy Rider wrote:

Hi everyone

I'm trying to implement OpenSIPS-like mid registrar functionality 
using UAC module


Up to now, I was simply performing authentication on my kamailio 
instance, saving to local domain and forwarding the request to another 
proxy
I'd like to lower the number of requests sent to this other proxy, by 
forwarding only a few of the incoming requests. This is not a problem 
to do, but then comes the handling of expiration. If the SIP UAC is 
disconnected, the registration expires on my kamailio instance, but I 
don't know how to correctly notify the other proxy on the 
unregistration (I was trying to build and send a request using 
uac_send_req(), but this is useless as I can't set the CSeq value)


While looking for a solution to my problem, I discovered that OpenSIPS 
had a mid registrar module doing exactly what I'd like to do 
(according to the documentation at least). I also saw old messages on 
this list that it was possible to achieve the same result with UAC 
module on kamailio


In particular, I don't see why (or if) I need to populate the uacreg 
table. I don't see why it should be necessary, and it would have a 
great impact on performance to use a db for this matter. I also don't 
need (I think) to have different local and remote from and to URIs


I think I need to build the a new REGISTER request using $uac_req, but 
I don't see any contact header in the outgoing request, despite having 
set reg_contact_addr value


Thanks for any clue you could give me

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Re: [SR-Users] secfilter table error

[Social Boh]

Hello,

with latest 5.3 version, the problem disappear.

My Database server is: 5.5.65-MariaDB on CentOS 7.8

Thank you

Regards

---
I'm SoCIaL, MayBe

On 5/7/20 01:24, Daniel-Constantin Mierla wrote:

Hello,

can you fetch the latest sources from branch 5.3 and try again?

Also, what type and version of database server do you have?

Cheers,
Daniel

On 06.05.20 17:43, Social Boh wrote:

Yes:

INFO: test server charset
INFO: creating database kamailio ...
INFO: granting privileges to database kamailio ...
INFO: creating standard tables into kamailio ...
INFO: Core Kamailio tables succesfully created.
Install presence related tables? (y/n): y
INFO: creating presence tables into kamailio ...
INFO: Presence tables succesfully created.
Install tables for imc cpl siptrace domainpolicy carrierroute
         drouting userblacklist htable purple uac pipelimit mtree sca
mohqueue
         rtpproxy rtpengine secfilter? (y/n): y
INFO: creating extra tables into kamailio ...
ERROR 1067 (42000) at line 1: Invalid default value for 'action'
ERROR: Creating extra tables failed at secfilter!

Thank you

Regards

---
I'm SoCIaL, MayBe

On 5/6/20 03:25, Daniel-Constantin Mierla wrote:

Hello,

is this when you run kamdbctl create?

Cheers,
Daniel

On 05.05.20 20:30, Social Boh wrote:

Hello,

any help with this error?

ERROR 1067 (42000) at line 1: Invalid default value for 'action'
ERROR: Creating extra tables failed at secfilter!

Kamilio version 5.3.3 compiled from source.

Thank you

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Re: [SR-Users] secfilter table error

[Social Boh]

Yes:

INFO: test server charset
INFO: creating database kamailio ...
INFO: granting privileges to database kamailio ...
INFO: creating standard tables into kamailio ...
INFO: Core Kamailio tables succesfully created.
Install presence related tables? (y/n): y
INFO: creating presence tables into kamailio ...
INFO: Presence tables succesfully created.
Install tables for imc cpl siptrace domainpolicy carrierroute
        drouting userblacklist htable purple uac pipelimit mtree sca 
mohqueue

        rtpproxy rtpengine secfilter? (y/n): y
INFO: creating extra tables into kamailio ...
ERROR 1067 (42000) at line 1: Invalid default value for 'action'
ERROR: Creating extra tables failed at secfilter!

Thank you

Regards

---
I'm SoCIaL, MayBe

On 5/6/20 03:25, Daniel-Constantin Mierla wrote:

Hello,

is this when you run kamdbctl create?

Cheers,
Daniel

On 05.05.20 20:30, Social Boh wrote:

Hello,

any help with this error?

ERROR 1067 (42000) at line 1: Invalid default value for 'action'
ERROR: Creating extra tables failed at secfilter!

Kamilio version 5.3.3 compiled from source.

Thank you

--
---
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[SR-Users] secfilter table error

[Social Boh]

Hello,

any help with this error?

ERROR 1067 (42000) at line 1: Invalid default value for 'action'
ERROR: Creating extra tables failed at secfilter!

Kamilio version 5.3.3 compiled from source.

Thank you

--
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Re: [SR-Users] TLS verification error LetsEncrypt

[Social Boh]

I'm not sure but with let's encrypt you can create only server 
certificate, not client certificate so you can't require and verify 
client certificate.


Regards

---
I'm SoCIaL, MayBe

El 24/01/2020 a las 09:01, Bugaian A. Vitalie escribió:

Ok, thanks.

But my question is still about why verification fails/or what should 
be chked to make it work. Not how to disable it.


Thanks.

Vitalie.

On Fri, Jan 24, 2020 at 2:54 PM Social Boh <mailto:soc...@bohboh.info>> wrote:


Hello,

changing:

[client:default]
#method = TLSv1.2+
verify_certificate = yes

require_certificate = yes

with

[client:default]
#method = TLSv1.2+
verify_certificate = no
require_certificate = no

---
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El 24/01/2020 a las 08:46, Bugaian A. Vitalie escribió:

Hello list,

I have tried to setup my tls config tish LetsEncrypt following
this post:

https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/

My tls config looks like this:


[server:default]
method = TLSv1.2+
verify_certificate = no
require_certificate = no
private_key = /etc/letsencrypt/live/sbc.example.net-0001/privkey.pem
certificate =
/etc/letsencrypt/live/sbc.example.net-0001/fullchain.pem
ca_list = /etc/letsencrypt/live/sbc.example.net-0001/ca_list.pem
#ca_list = /usr/local/etc/kamailio/tls/cacert.pem
#crl = /usr/local/etc/kamailio/tls/crl.pem
server_name = sbc.example.net <http://sbc.example.net>
server_id = sbc.example.net <http://sbc.example.net>

#ca_list = /usr/local/etc/fullchain.pem
#ca_list = /usr/local/etc/kamailio/tls/cacert.pem
#crl = /usr/local/etc/kamailio/tls/crl.pem


# ---
# This is the default client domain profile.
# Settings in this domain will be used for all outgoing
# TLS connections that do not match any other
# client domain in this configuration file.
# We require that servers present valid certificate.
#
[client:default]
#method = TLSv1.2+
verify_certificate = yes
require_certificate = yes

===
My ca_list has all certificates from
cat /etc/ssl/certs/ca-certificates.crt >>
/etc/letsencrypt/live/sbcc.example.net/ca_list.pem
<http://sbcc.example.net/ca_list.pem>

I keep getting certificate validation failed see bellow:

an 24 08:39:56 sbc.example.net <http://sbc.example.net>
/usr/local/sbin/kamailio[6371]: ERROR: tls [tls_util.h:42]:
tls_err_ret(): TLS write:error:1416F086:SSL
routines:tls_process_server_certificate:certificate verify failed
Jan 24 08:39:56 sbc.example.net <http://sbc.example.net>
/usr/local/sbin/kamailio[6371]: ERROR: 
[core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req:
error reading - c: 0x7f0474421f68 r: 0x7f0474422028 (-1)
Jan 24 08:39:56 sbc.example.net <http://sbc.example.net>
/usr/local/sbin/kamailio[6370]: ERROR: tls [tls_util.h:42]:
tls_err_ret(): TLS write:error:1416F086:SSL
routines:tls_process_server_certificate:certificate verify failed
Jan 24 08:39:56 sbc.example.net <http://sbc.example.net>
/usr/local/sbin/kamailio[6370]: ERROR: 
[core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req:
error reading - c: 0x7f0474401cb8 r: 0x7f0474401d78 (-1)

=

What params should I change or where to look for a solution on
this one?

Thanks.

Vitalie A. Bugaian.

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Re: [SR-Users] PUBLISH ERROR

[Social Boh]

The very strange think is I have disabled all VoIP settings on the Router

---
I'm SoCIaL, MayBe

El 10/10/2019 a las 10:53, Alex Balashov escribió:
> Pretty typical of a SIP ALG in a consumer router, yes.
>
> —
> Sent from mobile, with due apologies for brevity and errors.
>
>> On Oct 10, 2019, at 11:50 AM, Social Boh  wrote:
>>
>> I think is a Internet Router Issue. If I use 5060 port on Kamailio, the
>> Publish have Content-Lenght Header = 0
>>
>> If I change the Kamailio listen port to 5080, for example, the PUBLISH
>> have Content-Lenght Header > 0
>>
>> Strange?
>>
>> ---
>> I'm SoCIaL, MayBe
>>
>>> El 10/10/2019 a las 10:27, Social Boh escribió:
>>> Hello,
>>>
>>> Kamailio 5.2.5.
>>>
>>> I'm testing presence on my Kamailio using different Clients (Jitsi,
>>> Bria5, Linphone).
>>>
>>> when they send the PUBLISH to Kamailio, all receive the same answer: 400
>>> Invalid Request
>>>
>>> On the LOG file: ERROR: {1 21 PUBLISH O8-BtqLnzw} presence
>>> [publish.c:421]: ki_handle_publish_uri(): No E-Tag and no body found
>>>
>>> I'm pretty sure same configuration work with 5.2.3 version.
>>>
>>> Any hint?
>>>
>>> Regards
>>>
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Re: [SR-Users] PUBLISH ERROR

[Social Boh]

I think is a Internet Router Issue. If I use 5060 port on Kamailio, the
Publish have Content-Lenght Header = 0

If I change the Kamailio listen port to 5080, for example, the PUBLISH
have Content-Lenght Header > 0

Strange?

---
I'm SoCIaL, MayBe

El 10/10/2019 a las 10:27, Social Boh escribió:
> Hello,
>
> Kamailio 5.2.5.
>
> I'm testing presence on my Kamailio using different Clients (Jitsi,
> Bria5, Linphone).
>
> when they send the PUBLISH to Kamailio, all receive the same answer: 400
> Invalid Request
>
> On the LOG file: ERROR: {1 21 PUBLISH O8-BtqLnzw} presence
> [publish.c:421]: ki_handle_publish_uri(): No E-Tag and no body found
>
> I'm pretty sure same configuration work with 5.2.3 version.
>
> Any hint?
>
> Regards
>

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[SR-Users] PUBLISH ERROR

[Social Boh]

Hello,

Kamailio 5.2.5.

I'm testing presence on my Kamailio using different Clients (Jitsi,
Bria5, Linphone).

when they send the PUBLISH to Kamailio, all receive the same answer: 400
Invalid Request

On the LOG file: ERROR: {1 21 PUBLISH O8-BtqLnzw} presence
[publish.c:421]: ki_handle_publish_uri(): No E-Tag and no body found

I'm pretty sure same configuration work with 5.2.3 version.

Any hint?

Regards

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Re: [SR-Users] Plans for next major release - v5.3.0

[Social Boh]

Me too. Thank you

---
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On 5/20/19 02:49, Juha Heinanen wrote:

Daniel-Constantin Mierla writes:


Any preference for one of the two options?

I would prefer release after holidays.  Freeze and testing OK during
that period.

-- Juha

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Re: [SR-Users] How to configure Kamailio 5.1 with Asterisk pjsip

[Social Boh]

Try and failure. The best for you.

---
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On 5/8/19 07:31, PICCORO McKAY Lenz wrote:
El dom., 5 de may. de 2019 a la(s) 15:25, David Villasmil 
(david.villasmil.w...@gmail.com 
) escribió:


http://www.kamailio.org/wiki/tutorials/getting-started/main

Pretty vage, VAGE!

http://kb.asipto.com/asterisk:index which is about integrating
Kamailio and asterisk. It’s pretty old (4.0) but most still applies.

not "pretty"
VERY VERY OLD!

why there's no a good documentation and users must liate with english 
here in maillist?


by example i must "waith" to something if "wants" to answer "my" 
question! in that time i waste the time! du i cannot do anything due 
lack of good documentation!



After installing kamailio, check out the configuration file, it
should pretty much work for you unchanged by just enabling or
disabling “define”s

Good luck

David

On Sun, 5 May 2019 at 18:12, przeqpiciel mailto:przeqpic...@gmail.com>> wrote:

I would like to have an infrastructure where there are two
machines with Kamailio installed (K1, K2) and another two with
Asterisk installed (A1, A2).
Ultimately, I want to be able to make calls through any
Asterisk dialplan regardless of which Kamailio the
registration took place.
When this is achieved, I would like to register an account
provided by the VoIP provider and be able to accept and make
calls. Of course, the logic of these connections, IVR, Queue
is to be based on Asterisk.
So the question. Where can I find descriptions of how to do this.
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-- 
Regards,


David Villasmil
email: david.villasmil.w...@gmail.com

phone: +34669448337
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Re: [SR-Users] Kamailio TLS + letsencrypt

[Social Boh]

Hello,

I think you need a real domain or sub domain point to the Public IP of 
your server.


regards

---
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On 8/12/18 15:37, Amar Tinawi wrote:

Hello

i'm wondering if i can generate tls certificate for kamailio 
using letsencrypt with test domain name, in testing enviroment


following this post : 
https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/


got an error by letsecrypt because of checking of the domain name 
authority

and no certificate was generated.

Regards


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Re: [SR-Users] FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service, ClueCon Presentation Slides

[Social Boh]

Very interessing...

Thank you to share.

Regards

---
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On 7/25/18 11:29, Giovanni Maruzzelli wrote:

Hello fellow RTCers,

just finished presenting about scaling videoconferencing, chat and 
moderation/direction controls in SIP, with web clients, deskphones, 
smartphone apps, server push (google fcm) via cordova and SIP.js at 
www.cluecon.com .


Here's the slides of my presentation:

http://178.79.181.125/Maruzzelli_FreeSWITCH_OpenSIPS_WebRTC_Pure_SIP_Video_Call_Conferencing_and_Chat_ClueCon_2018_Chicago.pdf

or, shorter,

http://178.79.181.125/Maruzzelli_ClueCon_2018.pdf


From ClueCon blurb:


FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and
Conference Service

On top of traditional SIP deskphones and softphones, WebRTC let us 
extend the reach of SIP to all browsers, and to smartphone apps that 
wake up with a server push. OpenSIPS has superb SIP and WebRTC support 
and can protect, balance and scale FreeSWITCH insuperable 
conferencing, video MCU, media switching/mixing, SIP SIMPLE message 
processing capabilities. We will see how to build and implement such a 
complete platform, touching both on servers and on clients. Hint: 
SIMPLE, the SIP Instant Messaging Protocol, can be leveraged to 
transport statuses, commands and chats between participants and 
administrators of SIP audio/video calls/conferences.



  Giovanni Maruzzelli


Principal at OpenTelecom.IT



Giovanni is a consultant for the Telco sector, developing software and 
training courses for FreeSWITCH, SIP, WebRTC and Kamailio. He is 
heavily engaged with FreeSWITCH, of which he wrote the interface with 
Skype and with cellular phones. An Internet tech pioneer, in 1996 
Giovanni was cofounder of Italia Online, the most popular Italian 
portal and consumer ISP, and architect of its Internet technologies – 
www.italiaonline.it  Then supervisor of 
Internet operations and architect of the first engine for paid access 
to www.ilsole24ore.com  , the most read 
financial newspaper in Italy and to its databases (migrated from 
mainframe). After that, he was CEO of venture capital funded Matrice, 
developing Telemail unified messaging and multi language phone access 
to email (Text To Speech), and CTO of incubator funded Open4, an open 
source managed applications provider. Then he was for two years in 
Serbia as Internet and Telecommunication Investment Expert for World 
Bank – IFC. Since 2005 he’s based in Italy, and serves ICT and Telco 
companies worldwide.



-giovanni

--

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18




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[SR-Users] Understanding port changing

[Social Boh]

Hello list,

I'm looking for to understand a problem have with Kamailio.

The configuration is:

user (behind NAT) -> Kamilio (Public IP) -> user (behind NAT)

When I make a call from user A to user B (both registered) , User B 
receive the INVITE correctly but in the ringing SIP message change the 
port present on the Contact Header and same think on 200OK so when 
Kamailio try to send it ACK SIP message to this port fail because the 
Softphone listen on the port where has received the INVITE from Kamailio.


This happens with 3 different Softphone: BRIA, Xlite, Linphone on the 
same computer.


Thank you

Regards

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Re: [SR-Users] Resend Authenticate REGISTER to two or more Asterisk PBX

[Social Boh]

I'll try.

Thank you.

---
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El 08/03/2018 a las 02:52, Daniel-Constantin Mierla escribió:
> Hello,
>
> try with t_replicate() from tm module.
>
> Cheers,
> Daniel
>
> On 21.02.18 14:45, Social Boh wrote:
>> Hello,
>>
>> I'd like know which is the best way to resend a Authenticate REGISTER to
>> two or more Asterisk.
>>
>> Repeating rewritehostport and forward commands for the number of
>> Asterisk Servers, there is a problem with the Via Header; Kamailio add
>> one with his IP for each forward. I try to remove the first Via before
>> send the REGISTER to the second Asterisk without success.
>>
>> Any hint or tip?
>>
>> Thank you
>>
>> Regards
>>


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[SR-Users] KeepAlive Module

[Social Boh]

Hello,

I'm using keepalive module with drouting module... the event route works
fine:

NOTICE: 

[SR-Users] Resend Authenticate REGISTER to two or more Asterisk PBX

[Social Boh]

Hello,

I'd like know which is the best way to resend a Authenticate REGISTER to
two or more Asterisk.

Repeating rewritehostport and forward commands for the number of
Asterisk Servers, there is a problem with the Via Header; Kamailio add
one with his IP for each forward. I try to remove the first Via before
send the REGISTER to the second Asterisk without success.

Any hint or tip?

Thank you

Regards

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Re: [SR-Users] RTPengine - Transcoding example available?

[Social Boh]

Hello,

with this command:

*rtpengine --codecs*

            PCMA: fully supported
    PCMU: fully supported
    G723: fully supported
    G722: fully supported
   QCELP: supported for decoding only
    G729: supported for decoding only
   speex: fully supported
 GSM: fully supported
    iLBC: not supported
    opus: fully supported
  vorbis: fully supported
 ac3: fully supported
    eac3: fully supported
  ATRAC3: supported for decoding only
 ATRAC-X: supported for decoding only
 AMR: fully supported
  AMR-WB: fully supported
   PCM-S16LE: fully supported
 MP3: fully supported

you can see the list of codecs are supported or not and how use them

Regards

---
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El 19/02/2018 a las 05:48, Denys Pozniak escribió:
> Hello!
>
> Small note.
> In case of opus use: codec-transcode-opus (opus is lowercase).
>
> On 17 February 2018 at 02:53, Richard Fuchs  > wrote:
>
> On 2018-02-16 07:55 PM, Carsten Bock wrote:
>
> Hi,
>
> Quick (and simple) question:
>
> Can someone provide a working RTPengine request, to enable
> Transcoding?
>
> I know, this is possible, but I thought before looking into
> deep, how this should be configured for the rtpengine_offer()
> command, could someone provide an example?
>
>
> It's pretty straight forward.
>
> Example: A offers some SDP to B.
>
> Let's say you think B probably speaks PCMU, so you want to make
> sure to offer it, even if A didn't offer it: forward transcoding.
> Add `transcode-PCMU`.
>
> Let's say you also know B is allergic to PCMA, so you want to
> remove it from the offer to B, but still be able to speak it to A:
> backward transcoding. Add `codec-mask-PCMA`.
>
> These two cases are really independent of each other, but can be
> combined.
>
> Full example with some other options:
>
> rtpengine_offer('ICE=remove rtcp-mux-demux codec-mask-PCMA
> transcode-PCMU')
>
> Simple!
>
> Cheers
>
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>
>
>
> -- 
>
> BR
> Denys Pozniak
>
>
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Re: [SR-Users] RTPEngine Transcoding [SOLVED]

[Social Boh]

Hello,

before I'm using 2.6.8 FFMPEG version

Installing 2.8.13 from rpmfusion repositories (CentOS 7) the problem was
solved.

Thank you.

Regards

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El 15/02/2018 a las 07:50, Richard Fuchs escribió:
> On 2018-02-15 07:35 AM, Social Boh wrote:
>> Hello,
>>
>> I'm trying to use this configuration:
>>
>> Softphone A audio codec g722 and Opus
>>
>> Softphone B audio codec PCMA and PCMU
>>
>> Kamailio with rtpengine_manage with codec-transcode-PCMU
>> codec-transcode-PCMA in the request
>>
>> Softphone A call Softphone B
>>
>> The RTPEngine DEBUG show many Decoder Error:
>>
>> https://pastebin.com/k1Zhw8HS
> Probably an issue with your version of the ffmpeg libs. Try a newer
> version if you can. Or try removing the '#if' and '#endif' lines in
> resample.c and recompile. (Check the values of the LIBAVUTIL_VERSION_*
> macros in your libavutil/version.h - there doesn't seem to be a
> consistent version test to check for the existence of
> av_frame_get_channels())
>
> Cheers
>
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Re: [SR-Users] RTPEngine Transcoding

[Social Boh]

Hello,

I'm trying to use this configuration:

Softphone A audio codec g722 and Opus

Softphone B audio codec PCMA and PCMU

Kamailio with rtpengine_manage with codec-transcode-PCMU
codec-transcode-PCMA in the request

Softphone A call Softphone B

The RTPEngine DEBUG show many Decoder Error:

https://pastebin.com/k1Zhw8HS

Thank you

Regards

---
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El 13/02/2018 a las 20:05, Richard Fuchs escribió:
> On 2018-02-13 07:22 PM, Social Boh wrote:
>> Thank you.
>>
>> which is the best way to implement ir
>>
>> codec-strip-all and the add codecs?
>>
>> Is there a document about how work the transcoding?
> It's really up to you and depends on what you want to achieve. Doing a
> strip-all and then adding codecs you want to offer is definitely one
> way. Another is to just add codecs that you know or think the peer may
> want to see.
>
> Cheers
>


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Re: [SR-Users] RTPEngine Transcoding

[Social Boh]

Thank you for your answer but I don't see any option to activate
transcoding on Kamailio RTPEngine module

Regards

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El 13/02/2018 a las 17:49, Richard Fuchs escribió:
> On 2018-02-13 03:34 PM, Social Boh wrote:
>> Hi List,
>>
>> I just read on the RTPEngine page the Proxy RTP now support transcoding:
>>
>> https://github.com/sipwise/rtpengine#transcoding
>>
>> is there a plan to support it?
>
> Kamailio's rtpengine module can already make full use of rtpengine's
> transcoding capabilities right now. (There are some slight
> improvements to be made to the control protocol, which I'll submit
> soon, but those are entirely optional.)
>
> Cheers
>
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[SR-Users] RTPEngine Transcoding

[Social Boh]

Hi List,

I just read on the RTPEngine page the Proxy RTP now support transcoding:

https://github.com/sipwise/rtpengine#transcoding

is there a plan to support it?

Regards

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Re: [SR-Users] kamailio 5.1 from git compile error

[Social Boh]

Hello Abdul,

check src/modules.lst

file

exclude_modules=

line must contain:

phonenum

phonenum is the new module present from 5.1 Kamailio version and depend
on libphonenumber library

https://kamailio.org/docs/modules/5.1.x/modules/phonenum.html

Regards

---
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El 17/01/2018 a las 15:40, Abdul Basit escribió:
> Hi Daniel,
>
> That's what i was thinking about phonenum module.
>
> I simply git clone form kamailio repo
>
> make cfg include_modules="utils app_lua db_postgres"
> make all
> make install
>
> Nothing else. I wonder why kamailio 5.1 want to use libphonenumber 
> where there was no such dependency for version 5.0.x
>
>
> --
> regards,
>
> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445
>
> On 17 January 2018 at 19:22, Daniel-Constantin Mierla
> mailto:mico...@gmail.com>> wrote:
>
> Hello,
>
> how did you compile kamailio? The phonenum module should not be
> compiled by default, only on demand...
>
> Cheers,
> Daniel
>
>
> On 16.01.18 17:27, Abdul Basit wrote:
>> Hi,
>>
>> I am able to compile kamailio 5.1 now.
>>
>> Added libphonenumber repo for latest debian packages
>>
>> echo “deb http://ftp.de.debian.org/debian
>>  sid main” >>
>> /etc/apt/sources.list.d/libphonenumber7.list
>>
>> apt-get update && apt-get upgrade
>>
>> Additional updated packages and dependencies to install are
>>
>> apt install libphonenumber7 libphonenumber-dev libicu57
>> libprotobuf10 libstdc++6 libgeocoding7 libboost-date-time1.62.0
>> libboost-filesystem1.62.0 libboost-system1.62.0
>> libboost-thread1.62.0 libboost-thread-dev
>>
>> apt install libprotobuf-dev
>>
>> Once done, kamailio 5.1 was compiled successfully.
>> I hope this will make someone's life easier.
>>
>> --
>> regards,
>>
>> abdul basit
>>
>>
>> On 16 January 2018 at 18:48, Abdul Basit > > wrote:
>>
>> Hi,
>>
>> I am trying to compile kamailio 5.1 from git following
>> https://kamailio.org/docs/tutorials/5.1.x/kamailio-install-guide-git/
>> 
>> 
>>  
>> on Debian GNU/Linux 8.10 (jessie)
>>
>> I also installed packages after getting dependency errors
>> apt-get install libhiredis-dev libphonenumber6-dev libicu-dev
>>
>> apt-get install unicode unicode-data
>>
>> Kamailio compile stopped at error
>> .
>> LD (gcc) [M uid_avp_db.so]  uid_avp_db.so
>> CC (gcc) [M pdb.so] pdb.o
>> LD (gcc) [M pdb.so] pdb.so
>> CC (gcc) [M phonenum.so]    phonenum_mod.o
>> CC (gcc) [M phonenum.so]    phonenum_pv.o
>> Compiling cphonenumber.cpp
>> g++ -fPIC -DPIC -g -funroll-loops -Wcast-align -m64
>> -minline-all-stringops -falign-loops -ftree-vectorize
>> -fno-strict-overflow -Wall -Wno-write-strings -Wno-deprecated
>> -Wno-unused-f
>> unction -Wno-sign-compare -Wno-strict-aliasing -fPIC -DPIC -g
>> -funroll-loops -Wcast-align -m64 -minline-all-stringops
>> -falign-loops -ftree-vectorize -fno-strict-overflow -Wall  -DNA
>> ME='"kamailio"' -DVERSION='"5.1.0"' -DARCH='"x86_64"'
>> -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.9.2"'
>> -D__CPU_x86_64 -D__OS_linux -DSER_VER=5001000
>> -DCFG_DIR='"/usr/loc
>> al/etc/kamailio/"' -DRUN_DIR='"/var/run/kamailio/"'
>> -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST
>> -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE
>> -DUSE_DNS
>> _FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP
>> -DMEM_JOIN_FREE -DF_MALLOC -DQ_MALLOC -DTLSF_MALLOC
>> -DDBG_SR_MEMORY -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS
>> -DSTATISTICS -DMALLO
>> C_STATS -DWITH_AS_SUPPORT -DUSE_SCTP -DFAST_LOCK
>> -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM
>> -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD
>> -DHAVE_MSG_
>> NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H
>> -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_IP_MREQN
>> -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND
>> -DUSE_FUTEX -DHAVE_SE
>> LECT  -I/opt/local/include -DKAMAILIO_MOD_INTERFACE
>> -DMOD_NAME='"phonenum"' -c cphonenumber.cpp -o cphonenumber.o
>> In file included from
>> 
>> /usr/include/phonenumbers/geocoding/phonenumber_offline_geocoder.h:26:0,
>>  from cphonenumber.cpp:24:
>> /usr/include/phonenumbers/base/memory/scoped_ptr.h:25:45:
>> fatal error: phonenumbers/base/template_util.h: No such file
>> or directory
>>  #include "phonenumbers/base/template_util.h"
>>