Thank you, I just was not sure what else would cause the relayed packets to not
be sent out to my fios router. As mentioned, I can pick any other server in my
network and I can see, in the pcap file, that the relay is attempted to the
selected server. I verified our ACL and it is it open for
Hello All,
Can I point Kamilio to use a STUN server located on my network? Kamilio will
not relay to my fios router but will relay to other Servers on my network.
Thank you,
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
Hello,
I am attempting to change the Message Body of an SIP message using
replace_body_all; I want to replace all attributes that start with rtcp.
Here is my code =>
replace_body_all("rtcp:{1}[0-9]{5,}", newstring), and I call
msg_apply_changes() after this call. Note that I verified that
.
>
> On Wed, Oct 24, 2018 at 12:49:55PM +, Wilkins, Steve wrote:
>
> > I was looking at the sdpops_mod.c code hoping that there was an easy way to
> > remove a sess_version (Session Description Protocol Version (v): 0)
> > structure, but there does not appear to b
stanza - in the logic of your route script. It's not a bug in Kamailio.
I can't tell you exactly what the cause is, but I believe this avenue of
exploration will prove fruitful. It's a fairly common problem.
On Wed, Oct 24, 2018 at 12:49:55PM +, Wilkins, Steve wrote:
> I was look
I was looking at the sdpops_mod.c code hoping that there was an easy way to
remove a sess_version (Session Description Protocol Version (v): 0) structure,
but there does not appear to be that functionality in that module. It might be
a bit of work to remove a duplicate sess_version line.
I am
Alex,
If this problem is a Kamailio "bug", is there a proper site to report it to?
Thank you
-Original Message-
From: sr-users On Behalf Of Wilkins, Steve
Sent: Wednesday, October 24, 2018 7:44 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Doub
R) - Users Mailing List
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data
when using rtpengine
I would repeat my invitation to analyse RTPEngine logs for signs of a double
offer/answer.
On Wed, Oct 24, 2018 at 11:03:11AM +0000, Wilkins, Steve wrote:
> I sho
I should also note that when the 200 OK is received from Asterisk, it does not
have the double SDP, only after Kamailio forwards the 200 OK does the double
appear.
-Steve
-Original Message-
From: sr-users On Behalf Of Wilkins, Steve
Sent: Tuesday, October 23, 2018 10:00 PM
No calls to fix_nated_sdp().
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Tuesday, October 23, 2018 9:58 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data
when using rtpengine
Also, is there
Let me check...Thank you.
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Tuesday, October 23, 2018 9:58 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data
when using rtpengine
Also, is there any
for brevity and errors.
-Original Message-
From: "Wilkins, Steve"
To: "Kamailio (SER) - Users Mailing List"
Sent: Tue, 23 Oct 2018 9:55 PM
Subject: Re: [SR-Users] Double Session Description Protocol Version (v) 0 data
when using rtpengine
Entire section.
-
doubled, or just the v=0 line?
--
Sent from mobile. Apologies for brevity and errors.
-Original Message-
From: "Wilkins, Steve"
To: "Kamailio (SER) - Users Mailing List"
Sent: Tue, 23 Oct 2018 9:07 PM
Subject: [SR-Users] Double Session Description Protocol Version (v
Please ignore this question...error on my part. Sorry!
From: sr-users On Behalf Of Wilkins, Steve
Sent: Sunday, October 21, 2018 9:30 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio forwarding call via Public IP Address vs
Private IP Address
Sorry I meant
Good morning All,
I have two almost Identical (Kamailio=>Asterisk) systems set up, but one of
them is receiving calls and forwarding them to the Asterisk Public IP, and the
other is Forwarding to the Asterisk Private IP. The call is coming in to the
number and domain in both scenarios. Any
Hello All,
When db_mysql is selected in the make, the make looks for -lmariadb. However,
I want MySQL, not mariadb. Is there a way to let the make know that MySQL is
preferred ov mariadb?
Thank you
___
Kamailio (SER) - Users Mailing List
Hello All,
I have an issue where Kamailio-RTPEngine-Asterisk calls work good when a
softphone UAC on an IOS phone makes an Inbound call to a WebRTC client.
However, if the softphone UAC is on Windows, it does not work (No Audio/Video).
I noticed in the Wireshark trace that when using IOS I
only in dialog-forming - that is to say, initial -
INVITEs and their replies. A BYE is an in-dialog request and should have a
Route set constructed from those RRs, but should not contain any RRs.
On Wed, Sep 19, 2018 at 02:26:27PM +, Wilkins, Steve wrote:
> Hi Mojtaba,
>
> I
is needed to paste the log for better understanding.
With Regards.Mojtaba
On Mon, Sep 17, 2018 at 2:17 AM Wilkins, Steve wrote:
>
> Hi Mojtaba,
>
> But when I send the 'BYE' doesn't the double Record-Route from the 'INVITE'
> (from IOS) need to be there, so that IOS can find
Hi Mojtaba,
But when I send the 'BYE' doesn't the double Record-Route from the 'INVITE'
(from IOS) need to be there, so that IOS can find it's Proxies and complete the
transaction and send back a '200 OK'?
Thank you,
___
Kamailio (SER) - Users
is received by default. In other words, Kamailio remove all
record routes form downstream or upstream in INVITE request by default.
You should paste a simple wireshark to solve it as soon.
With Regards.Mojtaba
On Sun, Sep 16, 2018 at 6:10 PM Wilkins, Steve wrote:
>
> Hi Henning,
>
>
>
Mailing List
Subject: Re: [SR-Users] re- double record route
Route sets need to be fastidiously and scrupulously conserved by both UAs on
both sides in all in-dialog requests.
On Sun, Sep 16, 2018 at 01:40:24PM +, Wilkins, Steve wrote:
> Hi Henning,
>
>
>
> Yes I do have that
Hi Henning,
Yes I do have that enabled. What is happening is that one of the providers on
IOS is sending a double record route on the INVITE, but it is getting lost
somewhere so when I send a 'BYE', I get a "404 not here". When I look at the
SIP message I see only one of the record routes
Good Morning All,
Is there any way to add a double record route? I tried adding a second record
route and I always only get the first one added.
I have tried record_route(), and record_route_advertised_address(...), but
I still only get the first record route added.
Thank you,
-Steve
:56PM +, Wilkins, Steve wrote:
> Thank you,
>
> I am doing this just as a test, because I cannot get a soft-phone to hang up
> when the 'BYE' is initiated by a WebRTC client (although, some provider
> soft-phones do work ,Hang up that is).
>
> I have pcap logs of the on
, 2018 at 02:55:04PM +, Wilkins, Steve wrote:
> Did I miss something, what was your way
Yes, you clearly did not read my responses. :-)
> $var(x) = $ct;
> $(var(x){s.substr,1,0});
$var(x) = $ct;
$dlg_var(callercontact) = $(var(x){s.substr,1,0});
Or in the interest of s
:22PM +, Wilkins, Steve wrote:
> Here is what I actually do =>
>
> $var(x) = $ct;
> $(var(x){s.substr,1,0});
> $dlg_var(callercontact) = $var(x);
Well, you can't do that. You have to do it my way. ;-)
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800
mar doesn't allow for it.
-- Alex
On Fri, Sep 07, 2018 at 02:36:50PM +0000, Wilkins, Steve wrote:
> Yes, I will assign it to
> $dlg_var(X) = $var(x);
>
> -Original Message-
> From: sr-users On Behalf Of Alex
> Balashov
> Sent: Friday, September 7, 2018 10:31 AM
>
07, 2018 at 02:20:11PM +, Wilkins, Steve wrote:
> 0(1) CRITICAL: [core/cfg.y:3489]: yyerror_at(): parse error
> in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column
> 1-23: pvar with transformations in assignment left side
This is the real issue. Are you doing
$var(x) = "abcd";
$(var(x){s.substr,1,0});
Hello all,
I took the example from the documentation, but Kamailio.cfg has errors with
this example.
Errors=>
0(1) CRITICAL: [core/cfg.y:3489]: yyerror_at(): parse error in
config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 1-23:
?
Can you print $ru and $du of that BYE in the logs and send them here?
On Sat, Sep 1, 2018 at 14:41 Wilkins, Steve
mailto:swwilk...@mitre.org>> wrote:
Right before t_relay, $mb =>
[BYE sip:3128145656@10.10.10.10:5060;alias=125.10.1.15~32940~2
SIP/2.0#015#012Via: SIP/2.0/TCP
172.21
On Sat, Sep 1, 2018 at 14:41 Wilkins, Steve
mailto:swwilk...@mitre.org>> wrote:
Right before t_relay, $mb =>
[BYE sip:3128145656@10.10.10.10:5060;alias=125.10.1.15~32940~2
SIP/2.0#015#012Via: SIP/2.0/TCP
172.21.1.124:5060;rport;branch=z9hG4bKPj88f9c57d-5db6-4731-83c9-df478782fa39;alias#015
Re: [SR-Users] handle_ruri_alias() question or issue?
What is in $du if you log it right after handle_ruri_alias()?
Maybe you overwrite it somewhere later?
Regards, Igor
On Sep 1, 2018, 7:34 PM +0200, Wilkins, Steve
mailto:swwilk...@mitre.org>>, wrote:
Thank you Joel,
My issue is that, given t
ously called set_contact_alias)... Or
you can call set_contact_alias on not call handle_ruri_alias. Two ways of doing
the same.
On Sat, Sep 1, 2018 at 04:47 Wilkins, Steve
mailto:swwilk...@mitre.org>> wrote:
Good Morning All,
The following Incoming request came in =>
SIP Incomin
Good Morning All,
The following Incoming request came in =>
SIP Incoming Request: [[BYE
sip:2406506175@20.20.20.20:5060;alias=10.10.10.5~55157~2 SIP/2.0
I called handle_ruri_alias(), and expected the destination and port to be set
to 10.10.10.5:55157, but it was not.
It was left
Hello All,
If the below Requests from 40.40.40.40(Asterisk) is received by
10.10.10.10(Kamailio)
Request Line: INFO sip:3145553313@10.10.10.10:5060;alias=30.30.30.30~43508~2
SIP/2.0
Shouldn't Kamailio forward this request to 30.30.30.30? so that the '200 OK'
can sent back to Asterisk for the
Hello All,
I have had an issue for quite some time where I needed to open two UDP ports
(via LISTEN) for some softphone UAC's to stay connected. Funny enough there is
one Provider that works with the Single UDP. I think I finally see why I was
needing two listening UDP ports. When the ACK is
Mailing List
Subject: Re: [SR-Users] regular expression in kamailio
On Sat, Aug 25, 2018 at 11:32:21PM +, Wilkins, Steve wrote:
> Sorry, bad habit of adding // to everything after :, and it was just example
> of wanting to get the 10d number out of a string. I will be more c
hanks Alex,
-Steve
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Saturday, August 25, 2018 7:22 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] regular expression in kamailio
On Sat, Aug 25, 2018 at 11:16:29PM +0000, Wilkins, Steve wrote:
> O
expression in kamailio
On Sat, Aug 25, 2018 at 11:07:26PM +, Wilkins, Steve wrote:
> I was thinking that would also give me the IP or FQDN. I should not
> have assumed that. I will go try it now.
Ah, no, that's $fu. :-)
Another option: $(fu{nameaddr.uri}{uri.user})
Now, of course,
Subject: Re: [SR-Users] regular expression in kamailio
Why not just use $fU?
On Sat, Aug 25, 2018 at 10:46:36PM +, Wilkins, Steve wrote:
> Hi All,
>
> I am trying get the 10digit number called in on using the following
> $var(caller) =
> $(fu{re.subst,[0-9][0-9][0-9][0-9][0-9][0-9
Hi All,
I am trying get the 10digit number called in on using the following
$var(caller) =
$(fu{re.subst,[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]});
I thought I could use a regular expression for the expression in
re.subst,expression
Thank you,
-Steve
Hi Joel, and thank you for your response.
Actually, I use one IP but two different ports. I have not yet figured out why
some Providers need two listening ports but other need one. As I mentioned,
the ones that need one port will only have one way Media if two ports are used.
This is
Hi Joel,
Actually, I have a strange issue. If I have a single UDP IP:Port listening,
most Providers/Phones have two-way Audio/Video, however, with some
Providers/Phones, I need to have two UDP IP:Port listners in order to get
two-way Audio/Video. Here is the strange thing. If I enable two,
options? Not sure if they would work without more
details of what you are trying to achieve.
On Fri, Aug 24, 2018 at 10:35 Wilkins, Steve
mailto:swwilk...@mitre.org>> wrote:
Hi All,
Is it possible to add or remove “listen” dynamically?
Thank you,
Hi All,
Is it possible to add or remove "listen" dynamically?
Thank you,
-Steve
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi Florian,
Thank you for your response. I checked and direct-media is off. Just as a
recap, here is where I am
I just can't get rtpengine to work. I have tried multiple configurations, but
to no avail. Note that calls work good if rtpengine is disabled.
Here is my setup =>
Public IP:
: Do not open unexpected password-protected attachments.
>>>Email originates from a non-MITRE system. Use caution.<<<
On Wed, Aug 22, 2018 at 05:05:02PM +0000, Wilkins, Steve wrote:
> The SIP traffic is working this way for me but I still see RTP traffic going
> directly
Hello all,
I am still trying to get RTPEngine to work. At this point, when I make a call,
I do see the offer in the rtpengine log, I also see the update of the SDP (c)
changing it to 20.20.20.20. However, after that, there are no logs other than a
bunch of "timer run time = 0.nnn sec",
between UACs and Asterisk.
On Wed, Aug 22, 2018 at 9:57 AM Daniel Tryba
mailto:d.tr...@pocos.nl>> wrote:
On Wed, Aug 22, 2018 at 12:49:54PM +, Wilkins, Steve wrote:
> My offer and answer =>
> rtpengine_offer("trust-address replace-session-connection replace-origin")
with RTPProxy and RTPEngine
It may be more helpful to post some logs from rtpengine. You should never see
"Call-ID not found" from an offer.
Cheers
On 2018-08-22 08:49, Wilkins, Steve wrote:
Here is my start up =>
rtpengine --interface 111.121.22.11\!27.22.132.10 --listen-ng 127.0.0.1:1
De la part de Wilkins, Steve
Envoyé : mercredi 22 août 2018 14:50
À : Kamailio (SER) - Users Mailing List
mailto:sr-users@lists.kamailio.org>>
Objet : Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Here is my start up =>
rtpengine --interface 111.121.22.11\!27.22.132.10 --listen-ng
users On Behalf Of Wilkins, Steve
Sent: Wednesday, August 22, 2018 8:43 AM
To: Kamailio (SER) - Users Mailing List
Subject: [SR-Users] Struggling with RTPProxy and RTPEngine
Hello all,
I can't seem to get either RTPProxy or RTPEngine to work correctly. I have
decided to concentrate on RTPEngin
Hello all,
I can't seem to get either RTPProxy or RTPEngine to work correctly. I have
decided to concentrate on RTPEngine because I have read that it works the best.
I am using Asterisk with Kamailio in front. When I make calls, I see RTPEngine
being hit but I continually get the error
Hi Pravin,
I would start off by doing an clean of your repositories, and then possibly
re-installing MariaDB. I also use MariaDB for my installation with Kamailio and
the only issue I remember having was some missing (.h) files, which I then had
to go the develop libraries for MariaDB.
Hi All,
I would like Kamailio to use MySQL 5.7, however when Kamailio installs and I
say I want to use MySQL, it installs 8.0. Can I someway direct Kamailio to
install version 5.7?
Thank you,
-Steve
___
Kamailio (SER) - Users Mailing List
Are your ports open?
-Original Message-
From: sr-users On Behalf Of Henning
Westerholt
Sent: Tuesday, August 14, 2018 1:08 PM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] kamailio rtp proxy set not working
Am Montag, 13. August 2018, 13:32:29 CEST schrieb ANOOP V M:
> I have
Good Morning everyone,
Has anyone ever gotten rtpproxy/rtpengine to work with Kamailio and Asterisk?
I have tried both, and even though I see no errors with either, and when I make
calls I see rtpproxy/rtpengine traffic, I am still seeing media traffic go
directly to and from Asterisk. I just
e WITH_NAT parameter.
14.08.2018 1:51, Wilkins, Steve пишет:
HI All,
I am not sure if I understand it correctly but I thought that I could use
rtpengine to redirect media packets. My current SIP flow is =>
Softphone=>Kamailio=>Asterisk=>Kamailio=>Softphone, and Media flows from is
As
.08.2018 1:51, Wilkins, Steve пишет:
HI All,
I am not sure if I understand it correctly but I thought that I could use
rtpengine to redirect media packets. My current SIP flow is =>
Softphone=>Kamailio=>Asterisk=>Kamailio=>Softphone, and Media flows from is
Asterisk<=>softpho
HI All,
I am not sure if I understand it correctly but I thought that I could use
rtpengine to redirect media packets. My current SIP flow is =>
Softphone=>Kamailio=>Asterisk=>Kamailio=>Softphone, and Media flows from is
Asterisk<=>softphone. But I don't want Media to flow this way.
That is, I
Are you going through a PBX like Asterisk? I am using rtpengine but I cannot
get media packets to go from Asterisk->Kamailio(rtpengine)->softphone. I get
no errors and I see rtpengine traffic, but the calls go Asterisk->softphone.
Thanks ALL
From: sr-users On Behalf Of Nicolas Breuer
Sent:
Hi All,
Is it possible for Kamailio to interface with a particular Asterisk Server
based on the FQDN of a caller?
I would like to pass calls received by Kamailio through to different Asterisk
Servers based on the FQDN of the caller. I have used Load Balancing before,
but I want to select
Hello All,
There appears to be issues with running "kamdbctl create" when using MySQL 8.
When this is ran there are syntax SQL errors. One such error is when doing
grants using "IDENTIFIED BY 'password'"; This throws a sequel error for
version 8 of MySQL.
Thank you,
This does not happen if I am using MariaDB. It appears to be either a MySQL 8
issue or MySQL issue in general.
From: sr-users On Behalf Of Wilkins, Steve
Sent: Friday, August 3, 2018 6:53 AM
To: Kamailio (SER) - Users Mailing List
Subject: [SR-Users] Other issues encountered with MySQL 8
Good Morning all,
When I do get Kamailio to compile with MySQL 8, I encounter the following issue
when running "kamdbctl create"
ERROR 1449 (HY000) at line 1: The user specified as a definer
('mysql.infoschema'@'localhost') does not exist
WARNING: Your current default mysql characters set
Hello all,
I started getting the following error while trying to compile Kamailio 5.1+ on
Centos 7
CC (gcc) [M db_mysql.so]my_fld.o
In file included from my_fld.c:22:0:
my_fld.h:37:2: error: unknown type name 'my_bool'
my_bool is_null;
Has anyone seen this before.
Thank you
Hi All,
Can someone explain the difference between allow trusted and allow address.
Thank you!
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
ehalf Of Daniel
Tryba
Sent: Thursday, June 14, 2018 11:40 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio as outbound proxy for PBX
On Thu, Jun 14, 2018 at 10:56:51AM +, Wilkins, Steve wrote:
> If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Ka
Good Morning All!
If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio
actually make the SIP call?
At some point I would like outbound calls to be controlled by Kamailio so that
the outside endpoints never communicate with the PBX.
Currently a call goes through Kamailio
Hello All,
I have noticed that sometimes when a call is made from one endpoint to another
through Asterisk via Kamailio, Asterisk sends an INVITE to Kamailio even after
the call has been established. Sometimes this does not happen. When it does
happen, calls drop.
Why would an INVITE be
Kamailio
Dear Steve,
Would you mind sharing your findings and solution with the list?
With kind regards
Pan B. Christensen
Developer
Phonect AS
> -Original Message-
> From: sr-users On Behalf Of
> Wilkins, Steve
> Sent: mandag 11. juni 2018 12:30
> To: Kamailio (SER) - Use
Got it working. Thank you everyone.
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
Wilkins, Steve
Sent: Sunday, June 10, 2018 3:06 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] No Video between WebRTC Client
Alex, Pan, Daniel,...
Could this group => group:BUNDLE audio video in Message Body have anything to
do with my Kamailio Video issue.
Thank you!!
-Steve
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
Wilkins, Steve
Sent: Sunday, June 10, 2
10, 2018 11:44:47 AM EDT, "Wilkins, Steve" wrote:
>I just did a test where I disabled VP8 in Kamailio using SDPOPS and I
>now get Video on the softphone however, I lost two-way Audio. Kamailio
>seems to be doing something with the codecs but still can't put my
>fin
, and apparently Audio.
I'm not sure if I getting closer or just moving the problem around.
Thanks All!
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
Wilkins, Steve
Sent: Saturday, June 9, 2018 12:47 PM
To: Kamailio (SER) - Users Mailing List
Subject: [SR-Users] No Video between
mailto:sr-users-boun...@lists.kamailio.org>>
On Behalf Of Wilkins, Steve
Sent: lørdag 9. juni 2018 03:59
To: Kamailio (SER) - Users Mailing List
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls
Hello again Pan a
leg.
With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users
mailto:sr-users-boun...@lists.kamailio.org>>
On Behalf Of Wilkins, Steve
Sent: lørdag 9. juni 2018 03:59
To: Kamailio (SER) - Users Mailing List
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Pe
leg.
Asterisk answers WebRTC with:
Media Description, name and address (m): video 10328 UDP/TLS/RTP/SAVPF 96 100
96 is chosen for this call leg.
With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users
mailto:sr-users-boun...@lists.kamailio.org>>
On Behalf Of Wilkins,
to transcode.
With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users
mailto:sr-users-boun...@lists.kamailio.org>>
On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 15:09
To: Kamailio (SER) - Users Mailing List
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Use
, which probably negotiates two different
codecs without the ability to transcode.
With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users
mailto:sr-users-boun...@lists.kamailio.org>>
On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 15:09
To: Kamailio (SER) - Users M
regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users
mailto:sr-users-boun...@lists.kamailio.org>>
On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 15:09
To: Kamailio (SER) - Users Mailing List
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamai
does
Asterisk send back to device A?
With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users
mailto:sr-users-boun...@lists.kamailio.org>>
On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 14:03
To: Kamailio (SER) - Users Mailing List
mailto:sr-users@lists.kamail
Hi All,
Issue: when a Call is made through Kamailio and Asterisk. Asterisk uses
incorrect Video RTP Payload Type when sending Video packets.
I have a situation where I make a call from Device A to Device B and Device A
is Registered in Kamailio. When Device A Calls Device B, Kamailio sends
Hi All,
I normally get on this board to ask questions but today I just wanted to take a
minute and say -
Thank you to all of you who have taken the time to help me. Some of you have
spent a lot of time schooling me
on Kamailio, its Interactions with a PBX and different clients. I have
HI All,
I am curious, for those of you who use Kamailio with Asterisk or another PBX.
Who forwards Registrations to Asterisk or PBX, and who lets Kamailio maintain
Registrations?
Thanks All!
___
Kamailio (SER) - Users Mailing List
further.
Thanks again!
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Daniel
Tryba
Sent: Friday, May 25, 2018 9:47 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Switching form UDP to TCP causes authentication errors
On
Hello All,
When I switched from UDP to TCP I started getting Authentication Errors,
Asterisk responds to an INVITE via Kamailio with a '401 Unauthorized', but
Kamailio does nothing with it. Processing just stops near WITH_BLOCK401407.
Shouldn't the 401 be relayed so a new INVITE can be sent?
mailio not forwarding ACK
sent to it by Asterisk
You have to listen on two different ports and advertise them according to you
routing if you have only one interface (IP) where your Kamailio is listening to.
Cheers,
Federico
On Tue, May 22, 2018 at 2:02 PM, Wilkins, Steve
<swwilk...@mitre.org<m
he listen
directive along with its advertise parameter.
Regards,
Federico
On Tue, May 22, 2018 at 12:12 AM, Wilkins, Steve
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
What is actually happening is –
I M USING Kamailio 5.2 with Asterisk 14.6 and 15.3. Asterisk cannot re
i/cookbooks/5.1.x/core#listen
Best regards,
Federico
On Mon, May 21, 2018 at 7:32 PM, Wilkins, Steve
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
Hello,
I set that double route to ‘2’ and Kamailio still not hearing traffic coming in
from Public. I think I need to figu
sk
Hi,
if you have Kamailio listening on private IP and public IP you need two
record-routes.
Have a look at this param of the rr module:
http://www.kamailio.org/docs/modules/devel/modules/rr.html#rr.p.enable_double_rr
Best regards,
Federico
On Mon, May 21, 2018 at 7:01 PM, Wilkins, Steve
&
Hello All,
When Kamailio sends a '200 OK' to Asterisk, it is putting its Public IP into
the Record-Routes. In Asterisk 14.6, it would send the 'ACK' back to the
Private IP Address of Kamailio, but Asterisk 15.x is using the Public IP
Address that Kamailio placed in the Record-Routes so...
ERROR: [core/pvapi.c:1452]: pv_printf(): no more space for spec
value
ERROR: [core/pvapi.c:1461]: pv_printf(): buffer overflow -- increase the
buffer size...
I have
pv_buffer_size=16384;
tcp_rd_buf_size=16384;
Any ideas?
Thank you,
___
Hi All,
I am using Kamailio 5.2 and Asterisk (14.6 & 15.x) and I am having a very
strange issue. If I use Asterisk 14.6, the call (WebRTC<=>WebRTC) works
perfectly. However, if I use Asterisk 15.x, the call drops in 30 seconds. In
comparing tcpdump files, the first place I see a difference
Hello,
I am using Kamailio 5.2 and Asterisk 15.3 and while doing a WebRTC (client1)
to WebRTC (client2) call, Asterisks sends a re-invite and Kamailio responds
with a '200 OK', but when Asterisk sends back the "ACK" to Kamailio, it cannot
be relayed.
Errors =>
WARNING:
on’t know if this is a Kamailio or Asterisk issue. I have worked on this
one a long time and I sure hope I’m close
Thank you everyone!
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
Wilkins, Steve
Sent: Friday, May 18, 2018 3:58 PM
To: Kamailio (SER) - Users Mailing
-Address.
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
Wilkins, Steve
Sent: Friday, May 18, 2018 12:15 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] Transport issue thought
I sent the log but I just wanted to cl
ers@lists.kamailio.org>
Subject: Re: [SR-Users] Transport issue thought
I don't understand, are you able to send an INVITE to Asterisk via TCP?
Could you share a trace of the calls, masking the sensible information?
Cheers,
Federico
On Fri, May 18, 2018 at 4:56 PM, Wilkins, Steve
<swwilk...@
d.
Best regards,
Federico
On Fri, May 18, 2018 at 2:31 PM, Wilkins, Steve
<swwilk...@mitre.org<mailto:swwilk...@mitre.org>> wrote:
Thank you Alex and Federico,
I verified, and SO_REUSEPORT is defined on my OS. I am using Kamailio 5.2 and
I set ‘tcp_reuse_port=yes;’ and $fs; this ha
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