Hello,
I moved the Kamailio database to RDS from a local instance and noticed that
there were a large number of connections from the Kamailio user.
What is the normal number of connection that can be expected?
Thank you,
__
Kamailio - Users
If I don't use 'listen', then packets do arrive on 5060.
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* sr-users@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the
sender!
Edit m
Hello,
I am attempting to get Kamailio to run as a docker container using a docker
network of type network=overlay (everything works great if network=host). The
issue is that incoming calls are rejected because of 'connection refused'. I
noticed that when running on an overlay network, netsta
I'm trying to use the kamcmd tml.tc_uac_start command to send raw SIP
messages in Kamailio 5.5, but I always get a error: 400 - Invalid
headers. I've also tried sending OPTIONS messages with the same
result. I've looked for examples of sending SIP messages via kamcmd,
but can't find any.
ka
Hello,
Is there a way to suppress the printing "date & time, host, and Kamailio
execution path" at the begging of each xlog line?
Example current xlog line:
Nov 4 16:54:52 myserver /usr/local/sbin/kamailio[7021]: INFO:
On Tue, 17 Nov 2020 at 08:37, 陈理军 wrote:
> Hi
> I want to configure Kamailio SIP server to act as a SBC.
> I had read the article of Kamailio working as SBC to connect MS Team
> project:
> https://skalatan.de/en/blog/kamailio-sbc-teams
> But I can not find the kamailio.cfg file for this scenario.
D
From: "+44128084" ;tag=81dd
Call-ID: 9bbd818c-d1c8-4c62-9fed-7878ed05cf721
CSeq: 1 ACK
User-Agent: PartitionwareTM SIP Toolkit v4.0.30319
Content-Length: 0
Max-Forwards: 69
Allow: INVITE, ACK, CANCEL, BYE, INFO, OPTIONS, PRACK
Supported: timer,
Hello,
Does search_append_body require an "\r\n" in the attribute that is being added?
If I don't use a "\r\n", it appears to look correct in the trace, however one
provider says the attribute that was added was just a continuation of the prior
attribute, which Wireshark does not show to be th
s the INVITE.
As long as the caller waits (ie until we get a CANCEL) we'll deal with
incoming registers and release invites. And if they wait long enough there
will hopefully be a 200 from a client and a call is set up.
Steve
On Wed, 22 Apr 2020 at 10:53, German Cancio
wrote:
> Mich
r there if there's a
200 OK right behind it.
Perhaps a proxy like Drachtio would work better for you?
Steve
On Wed, 8 Apr 2020 at 17:44, Luis Rojas G. wrote:
> Hello, Henning,
>
> I am worried about this scenario, because it's a symptom of what may
> happen in other cas
eader!
Feb 7 13:07:58 ip-172-31-45-179 /sbin/kamailio[18048]: CRITICAL: cdp
[diameter_avp.c:702]: AAAUngroupAVPS(): AVP:<>
I suspect this is due to vendor, but not experienced here. The docs seem
to suggest the 3gpp is supported, and the diameter xml is set for 0 &
annot read the whole AVP header!
I assume that the cdp function AAAUngroupAVPS() cannot sort the data
into JSON due to a buffer issue. Any pointers?
Steve
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Hi,
I'm normally a bystander. But on this occasion I've got to comment - there
are broken SIP implementations, and there are BROKEN ones. Surely there is
no hope with this one? If they can't get this right just imagine how many
more problems it will have.
Steve
On Fri, 25 Oc
after another 30 seconds.
I tried to 'set_max_time' on cnxcc, but this did not seem to trip the
event route upon timeout?
Does anyone have a pointer to any examples? I also am unsure of how to
update the timer and keep a call in play!
Steve
__
" that now
forces the INT values where needed and the object now is passed as an
INT, and then by the jansson_get() function to the avp.
Should have thought about that one. Thanks again :-)
Steve
On 21/05/2019 09:19, Daniel-Constantin Mierla wrote:
I haven't tried and I am not the
Never quite got this. UDP packets can be up to 64k, right? And
fragmentation is a standard IP feature if a packet is bigger than MTU size.
Steve
On Mon, 1 Apr 2019 at 16:43, Alex Balashov
wrote:
> The customary inter-Kamailio solution is jumbo frames (if you control
> network end-to-e
edia}) == "yes") {
route(MEDIA);
} else {
route(RELAY);
}
exit;
This is just a small part, but maybe that principle may be of some help?
Steve
On 12/03/2019 07:48, YASIN CANER wrote:
Hello,
It is
Yasin,
Thanks for that... I was thinking the DNS possibly. I can nslookup the
FQDN fine at the cli of the server and that returns the correct IP. I
will concentrate on the DNS and have a play.
Thanks,
Steve
On 28/01/2019 14:46, YASIN CANER wrote:
Hello
Mra2 domain couldnt be resolved by
an 28 14:19:36 telco-lon1-02 journal: [Diameter Routing Agent][22801]:
WARNING: cdp [receiver.c:866]: peer_connect(): peer_connect(): Error
opening connection to :3868 >Name or service not known
Thanks,
Steve
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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 T
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,
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doing this so that I can add the IP address to the rtcp attribute
because this is required by a Provider.
Any ideas?
Thank you,
-Steve
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Session Description Protocol Version (v) 0 data
when using rtpengine
More generally, Steve, responding to some themes I see in your posts over the
past few months:
You have a habit of posting overblown speculations about problems and painting
yourself into grossly overcomplicated interpretations
icated SDP
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
using Kamailio 5.1, does anyone know if this is an issue in later versions?
Thank you,
-Steve
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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
Hi Alex,
I just ran a test, and yes, there is an error in the rtpengine log "Error
Parsing RTP Header" and there is a double SDP present.
Thank you,
-Steve
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Wednesday, October 24, 2018 7:28 AM
To: Kamailio (SE
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
To
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 a
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
Hello all,
I noticed double Session Description Protocol Version (v) 0 data in the SDP
section when using rtpengine with Kamailio. Has any else noticed this? Is
there a way for Kamailio to remove one of them?
Thank you,
-Steve
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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
Sorry I meant forwarded to Kamailio Public IP Address, not Asterisk Public IP
From: sr-users On Behalf Of Wilkins, Steve
Sent: Sunday, October 21, 2018 7:16 AM
To: Kamailio (SER) - Users Mailing List
Subject: [SR-Users] Kamailio forwarding call via Public IP Address vs Private
IP Address
Good
Any ideas on what would cause this?
Thank you,
-Steve
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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
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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 se
re present 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,
>
;INVITE' are missing and the Provider (
softphone) returns "404 Not Here".
Thank you,
-Steve
-Original Message-
From: sr-users On Behalf Of Mojtaba
Sent: Monday, September 17, 2018 2:51 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] re- double record
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,
___
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record route when
the INVITE request 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 wrot
AM
To: Kamailio (SER) - Users 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,
>
>
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 f
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
12:37:56PM +, 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
4 Not
Here'. My confusion is that since the Request is making it to the provider,
why does the Request Route matter at this point.
-Steve
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Tuesday, September 11, 2018 7:25 AM
To: Kamailio (SER) - Users Mailing List
Su
Hello All,
Can a Route: be removed?, if so, how?
Thank you?
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, 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
Kamailio's grammar doesn't allow for it.
-- Alex
On Fri, Sep 07, 2018 at 02:36:50PM +, 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, Septem
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: p
Hi Joel,
Can I contact you at your email address?
Thanks,
-Steve
From: sr-users On Behalf Of Joel Serrano
Sent: Sunday, September 2, 2018 12:26 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] handle_ruri_alias() question or issue?
Hi Steve,
Can you send a pcap of the call
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
want the BYE to be sent to the alias 125.10.1.15:32940
Nothing I do seems to let me get that alias and send the BYE to that
address:port
Thank you,
-Steve
From: sr-users On Behalf Of Igor Olhovskiy
Sent: Saturday, September 1, 2018 2:05 PM
To: Kamailio (SER) - Users Mailing List
Subject:
still get routed to 20.20.20.20.5060.
I have even tried storing the original Via from the INVITE and using that in
the BYE, however, I get a variable to maintain it’s state.
Thanks again,
-Steve
From: sr-users On Behalf Of Joel Serrano
Sent: Saturday, September 1, 2018 10:43 AM
To: Kamaili
.20:5060.
Is my thinking correct or is there another way to set the destination and port
to the alias?
Thanks All!,
-Steve
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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
k you,
-Steve
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Okay, I will work with it some more. been working since 6AM, maybe I just need
to pick it up tomorrow.
Thanks again for your patience and help!
-Steve
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Saturday, August 25, 2018 7:34 PM
To: Kamailio (SER) - Users
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 +, Wilkins, Steve wrote:
> Oh, b
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, if
Hi Alex,
I was thinking that would also give me the IP or FQDN. I should not have
assumed that. I will go try it now.
Thank you!
-Steve
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Saturday, August 25, 2018 6:51 PM
To: Kamailio (SER) - Users Mailing List
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
extremely Strang!
Yes, I am trying to use RTPEngine but have not been successful. I have earlier
posts (this week) which explains the problems I am having with RTPEngine.
Thanks again!,
-Steve
-Original Message-
From: sr-users On Behalf Of Joel Serrano
Sent: Friday, August 24, 2018 7:04 PM
, the
Providers/Phones that worked when I had a single listener only have one-way
Audi/Video.
I use Kamailio 5.1 and Asterisk 15.3 (pjsip). This behavior is so strange,
Thank you,
-Steve
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sr-users
, the
Providers/Phones that worked when I had a single listener only have one-way
Audi/Video.
I use Kamailio 5.1 and Asterisk 15.3 (pjsip). This behavior is so strange,
Thank you,
-Steve
From: sr-users On Behalf Of Joel Serrano
Sent: Friday, August 24, 2018 2:17 PM
To: Kamailio (SER) - Users
Hi All,
Is it possible to add or remove "listen" dynamically?
Thank you,
-Steve
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;trust-address replace-origin
replace-session-connection ICE=force RTP/SAVPF");
I have tried direction ext ext; and many other combinations, each producing its
own incorrect behavior.
Thanks again,
Steve
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: 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
engine log.
Thank you to all who have looked at this,
-Steve
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irect 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&q
slog.conf and
restarting rsyslog, I get no logs. I am on Cento7. I do this for Kamailio and
logging works.
Thank you for your response,
-Steve
From: sr-users On Behalf Of Richard Fuchs
Sent: Wednesday, August 22, 2018 8:57 AM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] Struggling with
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
rs 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 RTP
-configuration error that can cause this?
I know I can send all my logs and configurations but I really want to try and
resolve this as a learning experience.
Thanks all,
-Steve
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.
-Steve
From: sr-users On Behalf Of Pravin .
Sent: Monday, August 20, 2018 9:52 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] help
Herewith attaching the screenshot for the error we are getting after issuing
apt-get install mariadb-server command...
Pls have a look on attached
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
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sr-users
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
can't get the media packets to not go
directly from Asterisk to the Callee and visa-versa.
Any ideas on where I am messing up in my kamalio.cfg file?
Thank you!,
-Steve
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).
Do I need to add rtpproxy_manage() at every SIP message?
Thank you,
-Steve
From: sr-users On Behalf Of Yu Boot
Sent: Tuesday, August 14, 2018 2:12 AM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] Rtpengine?
Hello.
It's not enough to install rtpproxy/rtpengine and start it. You
.
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
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: M
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 whic
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 and
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 canno
Hello All,
When Kamailio installs, it installs mysql 8.0. I would like it to install
mysql 5.7. Is there a way to force this?
Thank you
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/li
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 to
sent back to Kamailio?
Also, Pan, I am working on the response you requested yesterday.
Thanks you All,
-Steve
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi Pan,
I will type something up. It may take me a bit to explain it correctly, but I
will definitely send you something. Although I now have a problem with
outbound calls hanging up, I at least get two-way Audio and Video. I think the
problem is just moving around.
Thank you,
-Steve
t=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob
SIP/2.0
Shouldn't Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio
send Incorrect Contact?
Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR.
Thank y
Hello,
Do you have a PBX between Kamailio and sipml5?
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of vinod
mn
Sent: Monday, June 11, 2018 6:19 AM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] SIP/2.0 603 Failed to get local SDP for sip to webrtc
(SIPML5)call
Hi
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 and
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
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