On Thu, Aug 23, 2018 at 09:42:09AM +0200, sewl...@gmail.com wrote:
> https://freeswitch.org/confluence/display/FREESWITCH/Kamailio+basic+setup+as+proxy+for+FreeSWITCH
>
> I would like to to ensure RTP proxy is always used(possible re-framing
> ptime) and we have multiple soft switches
Don't
On Thu, Aug 23, 2018 at 04:43:40AM -0400, Alex Balashov wrote:
> The expectation that the RTP will come from the same place as the
> signalling does exist in some sclerotic telco interconnects, since big
> brand SBCs would ordinarily meet this need and big brand SBCs are the
> only thing sclerotic
On Thu, Aug 23, 2018 at 11:39:32AM +, Wilkins, Steve wrote:
> If RTPEngine is on the same server as Kamailio (Asterisk being on
> another server), and RTP traffic is sent to and from RTPEngine, then
> the provider only needs to whitelist one IP-Address. I thought with
> RTPEngine that all RTP
On Sun, Jul 22, 2018 at 11:11:12AM +0200, Pali Roh??r wrote:
> Kamailio should try to send SIP messages directly to foreign SIP domains
> and when it fails (e.g. user not available or foreign SIP is not
> running), then forward message to that my SIP <--> XMPP gateway which
> will try to deliver
On Wed, Sep 05, 2018 at 06:36:20PM -0400, Alex Balashov wrote:
> Just grab it right before the consume_credentials() block, after all the
> challenge stuff.
consume_credentials() doesn't reset/clear $au, so anywhere after the
first if or after the route(that authenticates) will do.
On Fri, Aug 31, 2018 at 12:11:53PM +0200, Igor Olhovskiy wrote:
> Can I somehow dump xavp_dst var to check if ds_select_dst is correct?
You could use pv_xavp_print():
https://www.kamailio.org/docs/modules/5.1.x/modules/pv.html#pv.f.pv_xavp_print
If there is a better way I'd like to know myself.
On Tue, Jul 03, 2018 at 11:21:50PM +0100, Marrold wrote:
> For interoperability reasons I need to re-write the From URI Domain ($fd)
> in requests proxied by Kamailio to the source IP (socket) that Kamailio
> will send them via.
>
> I have experimented with onsend_route and can successfully
On Tue, Jul 03, 2018 at 09:11:55AM -0400, Mike Little wrote:
> Sorry if this has already been answered but can Kamailio be configured to
> support clustering
Yes.
> and use Anycast?
Anycast is handled at a different layer, don't see why it couldn't be
used (not that I know anything about
On Tue, Mar 06, 2018 at 04:05:43PM +0100, Stefan R??etschli wrote:
> If I execute the command "serctl acl grant 1234 mygrant" I get a warning
> which I have to accept with "Y":
> Non-existent user '1234'. Still proceeding? [Y|N]
>
> Is it possible to deactivate this warning? I know why this
On Wed, Apr 04, 2018 at 11:57:55AM +0200, Loic Chabert wrote:
> Question is simple (but answer will be probably more complex): how can i
> sent in dialog BYE request to ASTERISK 1, and not asterisk 2.
I'm confused, what does your config look like? The normal way to do this
just to t_relay()
On Tue, Mar 27, 2018 at 02:31:10PM +0300, Tomi Hakkarainen wrote:
> I would like to get a view of calls currently running through Kamailio?
> I should be able to see some details about the call like the calling called
> numbers and the time call has been active.
>
> I???m not sure if Siremis
On Tue, Mar 27, 2018 at 01:23:40PM +0200, Daniel Tryba wrote:
> Mar 27 12:17:10 kam kamailio[11647]: DEBUG: topos_redis
> [topos_redis_storage.c:253]: tps_redis_insert_dialog(): inserted dialog
> record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 26
> Mar 27 12:17:10 kam kamailio[1
On Tue, Mar 27, 2018 at 01:23:40PM +0200, Daniel Tryba wrote:
> I was testing topos with a mysql backend and all was fine.
Forgot to mention it is version 5.1.2.
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
ht
I was testing topos with a mysql backend and all was fine. But since the
purpose of this machine doesn't require a complete mysqlserver I tried
the redis backend. With redis the calls will drop after some timeout
(somewhere < 450 seconds (session timer set to 900)).
Config:
# topos
On Tue, Mar 27, 2018 at 03:27:27PM +0200, Karsten Horsmann wrote:
> loading the module and the dependencies - check.
> set and dlg_flag via module param - check.
>
> MUST i use a DB?! Or what is missing?
>
>
> kamctl stats | grep dialog
> "dialog:active_dialogs = 0",
>
On Wed, Jan 10, 2018 at 01:08:10PM +0100, Daniel-Constantin Mierla wrote:
> Hello,
>
> can you watch the traffic and see if this NOTIFY is part of a dialog?
> Respectively, if it has a tag parameter in the To header? Eventually you
> can paste here the output of ngrep command for the NOTIFY:
>
>
On Tue, Mar 27, 2018 at 06:06:58PM -0400, Fred Posner wrote:
> > Anyone has advice on kamailio on a VM, when it only handles sip?? ?
> >
> ESX/VMware has been great for me with kamailio, even with RTP and high
> volume.
ESX works fine indeed. But when you run something like rtpengine with a
On Wed, Mar 28, 2018 at 01:05:17PM +0200, Daniel Tryba wrote:
> I'm seeing the same spam in syslog, but in my case the source are
> OPTIONS (asterisk qualifies), so they aren't part of a dialog.
I was wrong, these are locally generated OPTIONS by the dispatcher
On Wed, Mar 28, 2018 at 04:30:45PM +0300, Atux Atux wrote:
> here are the outputs of the commands i issued:
You should really start read the output of your commands:
> Mar 28 16:07:18 debian kamailio[460]: CRITICAL: [core/cfg.y:3450]:
> yyerror_at(): parse error in config file /etc/kamaili
>
> My Question is now how the RTP Media Stream should/can flow. The clients
> are in different other networks. So P2P Media Stream isn't possible. Should
> I now run the RTP Stream Client - Asterisk or Client - Kamailio - Asterisk?
What do you want to accomplish?
Fact is that asterisk has to
On Thu, Mar 29, 2018 at 11:37:37AM +0200, Benjamin Marty wrote:
> I need the scenario for scalability on the asterisk side.
>
> The scenario I still have in my head is that the clients -could- do the RTP
> stream directly with the Asterisk Server. The Asterisk Instances have all
> their own NATed
On Wed, Mar 28, 2018 at 03:49:18PM +0200, Daniel-Constantin Mierla wrote:
> > I was wrong, these are locally generated OPTIONS by the dispatcher
> > module.
> >
> >
> just for clarificiation, the OPTIONS are generated by the same kamailio
> instance, right?
>
> If yes, then I will look at adding
On Thu, Mar 29, 2018 at 06:44:01PM +0300, Ali Taher wrote:
> I'm facing an issue , when I'm getting back the 183 message from the
> supplier(B-Party) and forwarding it to the customer(A-Party); the customer
> is not hearing the RBT and asking to send 180 rather than 183 . (the
> supplier is only
On Thu, Mar 15, 2018 at 03:32:00PM +, Seyed Mohammad Mirheydari wrote:
> I have 5 asterisk servers. And I want to use kamailio as load balancer.
> How can route sip calls from outside to asterisk servers?
>
> Note:
> I installed kamailio and mairadb , etc.
> I configured subscriber for
On Wed, Mar 14, 2018 at 05:30:23PM +0100, Daniel-Constantin Mierla wrote:
> I want to highlight that the last stable versions (for the latest 3
> release series: 4.4, 5.0 and 5.1) include fixes for two issues that can
> crash a running instance of Kamailio, therefore it is strongly
> recommended
On Sun, Mar 18, 2018 at 12:59:28PM +0100, Kjeld Flarup wrote:
> Does this mean that Kamailio doesn't reuse this connection, and instead
> tries to establish a new one.
>
> And is it possible to have two concurrent connections, especially when it is
> trying to send an invite to a client behind
On Wed, Mar 21, 2018 at 07:44:50PM +0100, Kjeld Flarup wrote:
> My own fault
>
> Because I needed to be able to forward the call to multiple GSM numbers at
> the same voip provider, I split the call onto several instances of Kamalio
> to be able to create new call id's
>
> As a result, the
On Wed, Mar 21, 2018 at 11:05:15AM -0400, Sergiu Pojoga wrote:
> When receiving an INVITE over a specific LTE carrier, I'm seeing 'c=IN IP4
> 192.0.0.4' in SDP, which isn't technically a RFC1918 or RFC6598 IP address
> and thus nat_uac_test(8) fails.
Looking at the table at the bottom of
On Mon, Mar 19, 2018 at 04:23:01PM +0100, Kjeld Flarup wrote:
> Interesting
>
> Just to be sure that I understand You correctly.
> When a Register is done, then an Invite, must create a new TCP connection.
That is not what I tried to say. All I wanted to say was:
uac ipA:portX -> syn ->
On Tue, Mar 20, 2018 at 03:17:36PM +0200, Zigelman, Amit wrote:
> How will the data look in the DB?
> Will it hold both entries pretty much the same with a difference is some
> fields such as the capture node?
There the capture node and timestamps will differ (node,date,micro_ts),
the rest is the
On Fri, Mar 23, 2018 at 10:43:05AM +0100, Daniel Tryba wrote:
> On Thu, Mar 22, 2018 at 04:23:18PM -0400, John Petrini wrote:
> > If you're asking if nat_uac_test should be updated to check for 192.0.0.0/29
> > I think that's a great idea.
...
> I'm to lazy^W^Wdon't have the time
On Thu, Mar 22, 2018 at 04:23:18PM -0400, John Petrini wrote:
> If you're asking if nat_uac_test should be updated to check for 192.0.0.0/29
> I think that's a great idea.
Don't know if I got every use in the source tree of private nets, but
following diff should do the trick. This net can be
On Thu, Mar 22, 2018 at 09:26:02PM +0100, Kjeld Flarup wrote:
> Thanks for Your suggestion. I'm however not sure that it will solve the
> problem which is a TCP connection problem.
>
> I solved the issue by sending the call back to the kamailio which handles
> the registrations, and just let the
On Wed, Feb 28, 2018 at 04:54:19PM +0530, Cibin Paul wrote:
> In my setup, Kamailio acts as a registration server and route the
> calls to different asterisk boxes. The clients using IPV6 private ip
> and IPV4 public ip address are rejected by the asterisk server. On
> printing the contact
On Wed, Feb 28, 2018 at 01:50:20PM +0200, ?? ??
wrote:
> - Is it possible to tell Kamailio which transport protocol to use other
> than by configuring auth_proxy column of uacreg table with domain name that
> has SRV record?
Don't know.
> - What are the priorities of
On Fri, Mar 02, 2018 at 11:14:15AM +0100, Syrine Riahi wrote:
> bonjour
> je suis une ??tudiante en mast??re professionnel des syst??mes des r??seaux et
> t??l??communication j'ai choisie comme un sujet de m??moire mise en oeuvre de
> solution des VoIP et apr??s des grands recherches j'ai
On Mon, Oct 08, 2018 at 07:16:43AM -0400, Alex Balashov wrote:
> The SDP-bearing INVITE and response are simply passed along as-is by
> Kamailio, and it is the SDP which specifies where the media goes. So, if
> endpoint A calls through Kamailio proxy B to Asterisk server C via SIP,
> A and C will
On Thu, Nov 01, 2018 at 09:20:46AM +, Toffi Bossol wrote:
> my use case will be:
>
>- We use two Kamailio instances (A and B)
>
>- A Client registers to a Kamailio A using TLS (SIP over TLS). The TLS
> session data shall be stored into an external DB.
>- Kamailio A is now
On Mon, Oct 29, 2018 at 05:27:27PM +0100, Joan Salvatella wrote:
> On this setup we are facing 2 issues:
>- *Diversion headers access: *Currently, Kamailio only supports access
>to the last diversion header but since we are receiving traffic from Twilio
>(it sets the last Diversion
On Mon, Nov 12, 2018 at 09:59:10AM +0100, Jos?? Antonio Guti??rrez Delgado
wrote:
> Hi, I would like to know if it is possible to make the last register of a
> user the active register and if possible the only one.
> I'm with an Android application and sometimes I can not deregister properly
>
On Mon, Nov 12, 2018 at 10:11:06PM +0100, Henning Westerholt wrote:
> > What is the difference between topoh and topos?
> > I am using topoh which works fine without DB and only needs the mask_key to
> > be the same on the nodes.
> > Is there a benefit for using a DB and the topos module?
> > [..]
On Fri, Sep 28, 2018 at 10:07:09AM +0200, Ivan Ribakov wrote:
> In a basic scenario of one2one call, I was able to make Kamailio to forward
> INVITE to callee over TCP by issuing a REGISTER request with
> ???transport=tcp??? parameter. Although it worked, as far as I understand
> that means all
On Mon, Sep 03, 2018 at 10:58:22AM +0100, David Villasmil wrote:
> Interesting approach! Though i don't think that will work for me. I've been
> looking into my requirements, and I'd need to do weight-based distribution
> instead of load-balance, also what i need is to add destinations to the
>
On Thu, Sep 20, 2018 at 12:54:30AM -0300, volga...@networklab.ca wrote:
> User location lookup looks like can't handle long $rU like
> 10102-ce72256df4945bc472ed9c27a1037f46.
> It always return -3 404 not found.
This username is only 38 chars long, I have a username in a 5.1.4
environment with
On Thu, Sep 27, 2018 at 10:03:24AM +0200, Pieter Muller wrote:
> VoIP Supplier (IP:161.161.252.20{Private Net}) ? Connect via routing to
> Kamailio Server(155.155.16.2 eno2{private Net}),Kamailio
> Server(187.221.197.252 eno1{public facing})?connect to PBX(public IP set in
> dispatcher) ? connect
On Fri, Sep 28, 2018 at 11:47:12AM +0200, Pieter Muller wrote:
> If I insert my public IP under:
> # - start RTPProxy:
> #rtpproxy -l _your_public_ip_ -s udp:localhost:7722
> # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING
>
> Must I also modify UDP part below:
>
>
On Wed, Jan 02, 2019 at 03:25:55PM +0100, Daniel-Constantin Mierla wrote:
> However, there was no follow up, Florian said he has to monitor after doing
> some fixes on the system and see how it goes. Since then I haven't see
> another update, so if you can get gdb backtrace, we can see if it is
>
On Sun, Jan 13, 2019 at 10:08:31PM +0300, Soltanici Ilie wrote:
> With Asterisk, we are able to get some peer round-trip connection statistic
> by setting qualify=yes for the specified peer.
> It sends periodic OPTIONS to the peer and calculates the time round trip time.
> It's something like -
On Tue, Jan 15, 2019 at 03:20:10PM +0300, Soltanici Ilie wrote:
> OK, that looks interesting - and I think I would able to generate such
> options from kamailio??- but how do??I?? measure the time for a response for
> this request?
> Is there any variables which can provide response time for
On Tue, Jan 15, 2019 at 03:57:54PM +0100, Daniel-Constantin Mierla wrote:
> some off-topic remarks, maybe you can do something about or others can
> confirm/infirm what I am seeing:
>
> ?? 1) I am subscribed to sr-users mailing list with two email accounts, I
> get your email on my secondary
On Mon, Jan 21, 2019 at 06:55:32PM +0530, Prabhat Kumar wrote:
> How can i modify username before consuming credentials set in *$au*
> variable. I tried $au=$_s("xyz:"+$au); but it says "*read only pvar in
> assignment left side*"
> ERROR: bad config file (1 errors)
$au is readonly. My workaround
On Tue, Jan 22, 2019 at 12:08:34PM +0530, Prabhat Kumar wrote:
> Is there any function for username authentication? as we have for password
> i.e. pv_www_authenticate(realm, passwd, flags)
I guess your only option for now is to use KEMI and your favorite
scripting language available there.
>
On Wed, Dec 26, 2018 at 06:08:20PM +, Wilkins, Steve wrote:
> 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
On Wed, Dec 05, 2018 at 09:40:38AM +0100, Kjeld Flarup wrote:
> Yes, the Phones may be on either local LAN (Wifi) and Internet via mobile
> data.
How about use different local address, 1 with an advertise for external
clients, 1 without. Have local DNS resolv to the 1 ip without advertise.
On Thu, Nov 29, 2018 at 02:32:02PM +0300, Soltanici Ilie wrote:
> We have a Kamailio Instance running on Public IP Address, one of our ISP
> cannot send ACK back to us because we are sending 100 Trying - without
> "received" parameter.
> Is there any way in Kamailio to force a "received"
On Tue, Nov 20, 2018 at 10:41:51PM +0100, ybouj...@by-research.be wrote:
>
> WHEN I MAKE AN INVITE FROM 801 TO 803, I HAVE THIS ERROR MESSAGE :
>
> KAMAILIO SERVICE (ERROR:
On Wed, Nov 21, 2018 at 06:24:22PM +0100, ybouj...@by-research.be wrote:
...
> xlog("NATMANAGE coei\n");
...
> Is it an issue with the public ip address configured in the rtpproxy
> (/etc/default/rtpproxy)?
There is no issue. Please take a look at
On Thu, Jan 10, 2019 at 08:53:43AM +0100, Jos?? Antonio Guti??rrez Delgado
wrote:
> Hello, what I need to know is when that user has been registered for the
> last time or, if possible, all the times he has been registered.
>
> If you could tell me how the RPC commands are used, I would be very
On Mon, Jan 07, 2019 at 04:21:35PM +, Floimair Florian wrote:
> This turned out to be unrelated to Kamailio itself in our case.
> The problem was that the systemd-journald of the systemd version shipped with
> Debian stretch was sometimes eating up our CPU time on a single-core VM.
> After
On Tue, Jan 08, 2019 at 11:40:56AM +0100, Daniel-Constantin Mierla wrote:
> > My crashes don't appear to be related, see
> > https://github.com/kamailio/kamailio/issues/1784
> > Mine are triggered by topos/redis, even though I have topos disabled
> > with an even route.
>
> I requested some extra
On Thu, Jan 03, 2019 at 07:06:51PM +, Duarte Rocha wrote:
> As far as i can tell, i have 3 ways to permanently change the memory
> settings in Kamailio :
>
> /etc/init.d/kamailio , /etc/default/kamailio and src/core/config.h.
>
> What's the priority between them? If i have different values
On Wed, Jan 09, 2019 at 05:30:52PM +0100, Jos?? Antonio Guti??rrez Delgado
wrote:
> Hi, I would like to know if there is a possibility to see a user's
> connection history, or at least their last connection. Thank you
Define connection?
If you want to see where INVITEs are coming from see the
On Tue, Oct 09, 2018 at 01:09:46PM +0200, Daniel-Constantin Mierla wrote:
> Hello,
>
> does it happen that you have the pcap with the sip trace for this call?
> If yes, can you send it to me (can be sent directly if you have some
> sensitive data there)?
>
Is there any progress on debugging
On Tue, Jan 08, 2019 at 12:08:23PM +0100, Daniel-Constantin Mierla wrote:
> > Yes. I initially had it enabled but later disabled it for testing with a
> > drop in the event_route and never removed it. In 5.1.3 there was no
> > problem, the moment I updated to 5.1.6 the segfaults began and
On Thu, Sep 13, 2018 at 08:19:00AM +, Nathanael Eneroth wrote:
> UA1<--->KAM1<>KAM2<>UA2
> int1int1 int3 int3
> int2 int2
>
> I would like KAM1 to forward all SIP requests destined to int3 via
> KAM2 and vice versa. I am aware of the routing logic in
>
> What does work is "time" instead of "time_hires". Since I have no idea
> what values are available (I guess the table names?) and time only has a
> 1 sec resolution, what is a correct version of time_hires?
To answer my own question:
with
modparam("acc", "time_mode", 2)
and
Under
https://www.kamailio.org/docs/modules/5.2.x/modules/db_redis.html#db_redis.sec.usage
there is an example for using db_redis for accounting. This doesn't
appear to work (5.2.2)
modparam("db_redis", "keys", "acc=entry:callid,time_hires:callid")
results in:
db_redis [redis_dbase.c:1886]:
> To answer my own question:
>
> with
> modparam("acc", "time_mode", 2)
> and
> modparam("db_redis", "keys", "acc=entry:callid,time_attr:callid")
> you'd get a millisec resolution timer in the key for redis
Well scratch that, I wasn't looking correctly at the logs. This doesn't
work.
On Mon, Apr 08, 2019 at 05:40:30PM +0530, vinod mn wrote:
> I have a cloud server, when I make call from a sip phone (registered with
> kamailio),
> in the INVITE header I am seeing the via header with public IP, is there
> any way that I can modify via header to send only the private IP.
> The
On Thu, Feb 28, 2019 at 09:03:30AM -0400, PICCORO McKAY Lenz wrote:
> hih thanks for your respond, but seems you dont paid attention to my
> problem, with ports opened and redirected to pulbic ip with the AWS
> firewalling (tech support) call have sound, but and later NAT traversal
> with rtpproxy
On Thu, Feb 21, 2019 at 04:55:40PM +0530, Prabhat Kumar wrote:
> When i try to login via SIP Client(zoiper) i am getting the following error.
>
> Feb 21 11:20:26 ip-10-0-0-121 /usr/local/sbin/kamailio[21131]: INFO:
> [core/parser/parse_fline.c:144]: parse_first_line():
> ERROR:parse_first_line:
On Thu, Feb 21, 2019 at 02:59:03PM +0100, Jan-Hendrik D??rner wrote:
> I would like to get parallel forking working on my Kamailio installation, but
> I have trouble to accomplish that.
[serial forking!]
> Is there any working Kamailio example-config, where I can actual see the
> parallel
On Tue, Mar 05, 2019 at 04:22:12PM +, Sergio Charrua wrote:
> Content-Length: 262
>
> v=0
> o=root 1219665045 1219665045 IN IP4 55.66.77.88
> s=SomeSIPGateway
> c=IN IP4 55.66.77.88
> t=0 0
> m=audio 14326 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101
On Fri, Mar 01, 2019 at 08:23:11AM -0600, JR Richardson wrote:
> My mind is not right on this one, need a pointer. Here is the simple scenario:
> Carrier> The end device is a static IP, the proxy dips database and knows where
> to send numbers destined for the end device, but if there is no
>
On Wed, Feb 27, 2019 at 04:04:45PM -0400, PICCORO McKAY Lenz wrote:
>
> N]OTE: the public ip are not a real interface in the kamailio/rtppropxy
> machine, are provided by the service AWS at amazon! a NAT kind i guess!
>
But how are you calling rtp(proxy|engine) from kamailio? I think you
need
On Thu, Mar 14, 2019 at 06:01:41PM +0200, Vitalii Aleksandrov wrote:
> > What is wrong with the default behavior? That adds ICE records and
> > rewrites SDP c=.
> When a call goes through multiple proxies and every proxy inserts itself SDP
> becomes really huge. What I like in "force-relay" is
On Thu, Mar 14, 2019 at 02:47:24PM +0200, Vitalii Aleksandrov wrote:
> Well it's mostly rtpengine question but didn't know where should I send it
> and probably the answer will be more or less useful for kamailio users.
I get the feeling. You could open an issue in github, but
sipwise/rtpengine
On Wed, Mar 13, 2019 at 06:01:39AM +0800, Isravel Raja Thangamani wrote:
> What I want is, I want to register the Kamailio Trunk in an asterisk with
> Username Password authentication,
>
> My current setup for making that,
>
> VoIP Provider -> Kamailio(Public) -> Asterisk(Randomly Changing
I have 3 rtp backends defined with equal weights (33 each). But when I
look at the number of calls being handled the spreaded load is always
5:4:3 for the machines in the orderd listed:
modparam("rtpengine", "rtpengine_sock", "udp:10.235.32.60:7723=33
udp:10.235.32.59:7723=33
On Fri, Feb 15, 2019 at 07:23:07PM +0100, Cristian Livadaru wrote:
> You are AWESOME!
> That's exactly what happened, the call went to first asterisk, that one sent
> it further to the client trunk which responded with SIP 500, kamailio sent
> it to the next one, sip 500 again and thus blocking
On Wed, Feb 13, 2019 at 04:03:52PM +, Jesse Strahn wrote:
> I have a server running Kamailio with dispatcher. I am trying to
> direct calls from kamailio to an SRV record but am receiving a 404
> error on the calls. If I instead place one of the hostnames from the
> SRV record in my
On Mon, Feb 11, 2019 at 09:07:04AM -0800, Julien Chavanton wrote:
> Hi Marco, not sure if it is the same issue, but I am looking at a problem I
> am facing where in-dialog requests are failing after 3 minutes.
>
> It seems you are also using topos_redis
>
> Tracing TOPOS traffic is seems some
On Fri, Feb 08, 2019 at 11:08:03AM -0800, Julien Chavanton wrote:
> The solution that worked for me was to use :
>
> trace_mode=1
>
> This is capturing both version of the message, I think this is about using
> a core event hook instead of a transaction callback
Although this is not the
On Fri, Feb 15, 2019 at 05:13:21PM +0100, Cristian Livadaru wrote:
> Hi,
> I have a Kamailio running as Load Balancer and it works great but since a
> couple of weeks I kept noticing 404s in Homer and when looked into it they
> came from Kamailio.
...
> The two asterisks behind it seem fine and I
On Thu, Jan 24, 2019 at 02:30:06PM -0300, Marcos Pytel wrote:
> I'm using Kamailio version: kamailio 5.1.6 (x86_64/linux).
>
> When I enabled topos, after a few minutes i get this erros in the log file
> and the LB Service goes down.:
>
> Jan 24 13:20:21 sipwise lb[30130]: ERROR: ndb_redis
On Fri, Jan 25, 2019 at 10:19:45AM -0300, Marcos Pytel wrote:
> Could you help me to catch the coredump file?
Please keep the mailinglist in the loop.
> When I execute bt full the systems shows "no symbols". What package i need to
> perform it?
I don't know what the sipwise config is, but it
On Thu, Jan 31, 2019 at 12:01:26PM +, YASIN CANER wrote:
>
> Kamailio gave this error that couldnt bind tcp connection other side.
>
No idea, what might be wrong with you conf. But you might start by
making a packetdump (of all traffic except eg ssh on all interfaces) to
see if the problem
On Fri, Feb 01, 2019 at 05:44:45PM +0100, Enrico Bandiera wrote:
> Hello, going back to 5.2.0 is actually not possible anymore if you didn't
> save locally the distro packages because right now on the repos only
> Kamailio 5.2.1 is available
I made the same conclusion with an upgrade to 5.1.6
On Mon, Feb 04, 2019 at 01:33:32PM +0100, Victor Seva wrote:
> Hi there,
>
> Is in my TODO list to integrate aptly into our deb build environment.
> https://github.com/sipwise/kamailio-deb-jenkins/issues/9
Okay, good to know that others see this as a "problem". I'll wait
patiently for this
On Fri, Mar 15, 2019 at 12:37:16PM +0200, Vitalii Aleksandrov wrote:
> Oh, it actually does. If you use ICE=force, rtpengine removes all ICE
> candidates and inserts its own and both call participant can't to talk to
> each other directly but still can use ICE to establish media streams to
>
On Fri, Mar 15, 2019 at 02:05:24PM +0100, Olivier wrote:
> I'm looking for a solution to integrate legacy devices to a SIP network.
> More precisely, I need to forward to and receive Clearmode RTP traffic (see
> [1]).
>
> 1. Do you know any Kamailio-compliant RTP engine (rtpproxy, rtp engine, ..)
On Thu, May 16, 2019 at 12:37:50AM +1000, Rhys Hanrahan wrote:
> Hi Daniel,
>
> Thanks for this, much appreciated. Was worried this approach was too much
> of a hack and not the right approach - but knowing someone else has gone
> this way gives me confidence it's a reasonable solution. Will try
On Wed, Jun 05, 2019 at 01:18:32PM +0300, wrote:
> When User and Group = kamailio I can't start kamailio.service at all. I get
> errors ()??
>
> >?? 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) ERROR:
> >[core/tcp_main.c:2855]: tcp_init(): bind(9,
On Mon, Jun 24, 2019 at 10:52:36AM +0300, Amar Tinawi wrote:
> Thank you Karsten
>
> i'll give a try
Karsten gives an excelent howto, but personally I use aptitude for these
situations. It gives you a list of all available package versions and
helps you with the conflicts that might surface.
On Mon, Jun 17, 2019 at 05:18:42PM +0200, Laura wrote:
> ++--+--+-+-++
> | Field | Type | Null | Key | Default |
> Extra
>
On Thu, May 09, 2019 at 10:15:16AM +0100, Mark Boyce wrote:
> We???ve been asked a few times recently if we can do screen-pop of
> incoming calls (SalesForce CRM / Zoho Support) so that customers
> details pop up on the display as calls are delivered. Similarly
> ???click to dial??? from such
On Tue, May 21, 2019 at 03:57:02PM +0200, Benoit Panizzon wrote:
...
> So if the Voice Switch is sending back the Register Contact in the
> INVITE, the PBX cannot use this field to determine which extension to
> ring.
>
> So it has to use the To: Header.
>
> Well, not in a forwarding scenario,
On Wed, May 15, 2019 at 09:38:29PM +1000, Rhys Hanrahan wrote:
> Just to add I've also tried adding multiple alias= definitions, but have
> the same issue - kamailio says "user@fqdn" not found in usrloc when doing
> lookup(). Maybe I need to modify my lookup() call to use a hardcoded URI?
> But I
On Fri, Apr 19, 2019 at 09:44:14AM +0300, Yu Boot wrote:
> Following code snippet from default kamailio.cfg never gives 403 if you
> smart enough to set "fromdomain" parameter on Asterisk to Kamailio's IP. How
> to fix it? I want password-based registration (which is OK now) and permit
> calls via
On Fri, Apr 19, 2019 at 11:38:50AM +0300, Yu Boot wrote:
> Added this before final "return", it still allow to call from any IP without
> registration. :(
>
> > if(!allow_source_address() || $au==$null)
> > {
> > sl_send_reply("403","Go away!");
> > }
This code is in no way related to
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