On Mon, Apr 10, 2017 at 12:34:51PM -0300, Diego Nadares wrote:
> Thanks for your response. Your suggestion helped me a lot. We don't use
> domains so instead a domain I set the ip address ($Ri).
Just took a look at the dns options, strangely enough there is no way to
disable it apparently.
> I a
On Sat, Apr 08, 2017 at 09:24:29PM -0300, Diego Nadares wrote:
> ERROR: [resolve.c:1694]: sip_hostport2su(): could not resolve hostn
> ame.
>
> Reconfigure those gw it's not possible. There are a lot of them and a lot
> of people that won't like that idea.
>
> Is possible to disable it? or What
On Thu, Apr 06, 2017 at 06:52:55PM +0200, DanB wrote:
> Many thanks again for premium support as always ;).
> Will get back to them since I understand Kamailio is RFC compatible on this
> one.
Out of curiosity, what type of Avaya PBX are youy having problems with?
I only have seen IP Offices on my
On Wed, Apr 05, 2017 at 02:25:29PM +0200, Abdoul Osséni wrote:
> The advantage of the SIP ping options is a bidirectional traffic through
> NAT. I think in this case, my issue will be solved.
Due to some limitations of the nat helper pinger (3 backends, one should
ping at hh:mm:00, the other at hh
On Fri, Mar 31, 2017 at 01:15:58PM +0900, Tahiro Hashizume wrote:
> > The hard part is how you will differentiate the REGISTERS from asterisk
> > to the sipserver.
> The imaginary solution to this in my head is telling Kamailio to
> listen to 10.1.1.1N:5061 (N=0-4) on dummyN as well, to which Aster
On Wed, Mar 29, 2017 at 09:57:02PM +0900, Tahiro Hashizume wrote:
> B.)Kamailio listens on five IP addresses on dummyN which has
> 10.1.1.1N/32 (N=0-4, so dummy0 has 10.1.1.10/32, for example) making
> five SIP+RTP sessions to the same remote host to appear as if they are
> from five different host
On Tue, Mar 07, 2017 at 11:06:20AM -0300, Diego Nadares wrote:
> Today I started testing dialog with db_mode = 3 (shutdown). The thing is
> that when I kill kamailio from terminal, the dialog module does not save
> any data. It only works when I execute kamctl stop.
You could ofcourse look at dbmo
On Tue, Mar 07, 2017 at 10:34:05AM -0300, Diego Nadares wrote:
> Is it possible to generate cdrs or to save dialogs to db when kamailio
> receives a kill signal from terminal?
Killing kamailio doesn't end running calls. So if you restart kamailio
before a BYE, accounting will be correct on teardow
On Thu, Mar 02, 2017 at 03:46:01PM +, José Seabra wrote:
> I changed the way that i was trying to get it done.
>
> Now I'm using the {line.sw,match} ( you had already mentioned this function
> on an old email),
Im not that Daniel :)
Or my memory starts to fail me.
> to split the string by
On Wed, Mar 01, 2017 at 04:58:00PM +, José Seabra wrote:
> #015#012Calling-Name-Status: available#015#012Calling-Name: "josé" <
> sip:52@10.10.10.10>#015#012Presentation-Indicator: allowed#015#012
>
> The Kamailio re.subst function is constructed as the following:
>
> $sht(cnam=>$ci::
On Wed, Mar 01, 2017 at 03:08:20PM +0100, Daniel-Constantin Mierla wrote:
> after a busy period of time with traveling and the release of kamailio
> v5.0.0, my current work involves a project with topos (I just pushed
> this morning a new module to master branch: topos_readis), so these days
> I wi
On Wed, Mar 01, 2017 at 09:55:52AM +0100, Igor Potjevlesch wrote:
> Again this morning. It looks to be during the beginning of the peak. I don't
> have any idea where to search the issue. Any ideas?
The root cause has always been the mysqlserver in my case. I've seen the
following jobs causing lo
On Mon, Feb 27, 2017 at 08:01:37PM +0100, Antony Stone wrote:
> Ah, okay - I had thought it would be easier to stick with something I know
> and
> just replace small bits of it with the unfamiliar Kamailio, adding to my
> knowledge one step at a time, but I can see that your suggestion makes sen
On Mon, Feb 27, 2017 at 07:11:28PM +0100, przeqpiciel wrote:
> Thanm you for respond. I would like to have a farm of asterisks for one
> domain and few single asterisk for dedicated domains. So probably i have to
> check destination domain and after that if i found this domain in my DB
> then i co
On Mon, Feb 27, 2017 at 06:25:40PM +0100, Antony Stone wrote:
> So, please can someone suggest a beginner's guide to introducing
> Kamailio into an Asterisk setup, keeping things as simple as possible
> to start with, so I can learn about Kamailio one bit at a time (since
> there do seem to be a lo
On Sun, Feb 26, 2017 at 06:33:08PM +0100, przeqpiciel wrote:
> Let's suppose that i have two machines with installed asterisk and one with
> kamailio. I would like have routing based on sip domain. for domain
> sip.domain1.com send sip signalling to asterisk#1 server and for
> sip.domain2.com send
On Thu, Feb 23, 2017 at 12:54:17PM +0530, vi...@advaitamtech.com wrote:
>
> It dint solve my problem. Am not able to get the proxy authenticate
> header values using the module you mentioned too. Does this allow to
> fetch the SIP header individually?
Yes, it use it to extract (part of) the co
On Wed, Feb 22, 2017 at 07:04:32PM +0530, vi...@advaitamtech.com wrote:
> So I will be receiving the "407 Proxy Authentication Required" response from
> the third party server for the first INVITE.
>
> I want to fetch the Proxy-Authenticate header present in the 407 response.
>
> I browsed and
On Thu, Feb 16, 2017 at 03:02:15PM +0100, przeqpiciel wrote:
> I'm trying to setup dispatcher module to working with mysql database but
> when i put to kamailio.cfg below lines then kamailio wont starts.
> i tried connect to mysql with credentials which you can see in mod_param
> line and i co
On Tue, Feb 14, 2017 at 08:32:27AM -0500, Annus Fictus wrote:
> In the parameter description, I read i have to use "lreq_callee_headers"
> dialog parameter to resolve this kind of issues, but I can't find this
> parameter on the module description.
>
> I'm using Kamailio 4.4.5
It is in the dialog
On Tue, Feb 14, 2017 at 02:27:59PM +0100, Igor Potjevlesch wrote:
> Looks to be a good idea.
> I use MySQL and MyISAM, so I understand that async is supported.
Async (db_insert_mode=2) is supported with any (acc)backend. With myisam you
can use delayed inserts (db_insert_mode=1). But without even
On Tue, Feb 14, 2017 at 09:54:42AM +0100, Igor Potjevlesch wrote:
[mysql query fails]
> These errors are repeated many times and cause Kamailio to hang and don't
> reply to SIP requests. A reboot of Kamailio solves the issue.
>
> Anyone has already had this issue and had solved it?
This is "norma
On Wed, Feb 08, 2017 at 12:28:36PM +0200, Arsen wrote:
> I am not sure that nat_uac_test can determine type of NAT device.
> and why you need all these checks if you always use rtpproxy? (another q
> from 2013 :)
The answer is: you don't have to.
> The idea is to reduce using of rtpproxy for bet
On Wed, Feb 08, 2017 at 01:12:05AM -0700, Arsen Semionov wrote:
> good question from 2013 :)
> Maybe someone has experience and can confirm this?
The answer to the 2013 question is: if you can depend on this (I have
never seen it) you can script kamailio to make use of it.
> My main question: is
On Thu, Feb 02, 2017 at 08:11:35AM +0100, Ján Füri wrote:
> I have the same experience with enabling end disabling rtpengines using
> commands like kamctl fifo nh_enable_rtpp udp:xyz.
> I tested in on kamailio 4.4.3
I have been able to produce crashes in any 4.4.x version so far tested
(making 4.4
On Sun, Jan 29, 2017 at 06:42:23PM +0100, Roman Dissauer wrote:
> When I get an INVITE with Diversion Header the Request is forwarded
> without Diversion Header and the Request User is taken from Diversion
> User. Problem is that on the Destination Host I cannot get original
> Request User what is
On Tue, Jan 24, 2017 at 03:50:24PM +, Pranathi Venkatayogi wrote:
> I am using Kamailio behind NAT, unable to figure how to make it put “public
> ip” in Record-route header, I am manually inserting the hard-coded header
> myself as below.
> However now I am having trouble choosing the rig
On Fri, Jan 20, 2017 at 05:11:32PM +0100, Daniel Tryba wrote:
> The host part of this contact is: $(T_rpl($ct){nameaddr.uri});
Which is not correct, {uri.host} should to the job (according to
https://www.kamailio.org/wiki/cookbooks/4.1.x/transformations#urih
On Fri, Jan 20, 2017 at 01:04:00PM -0300, Diego Nadares wrote:
> Anybody knows how to get the ip of the new contact of the redirect? I need
> it to update one xavp variable that contains the INVITE destination.
Don't know about an IP, but you get the contact in the (failure)
response with: $T_rpl(
On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote:
> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP,
> using raw socket capture, I think this may be a sipcapture issue,
> debuging kamailio shows normal startup and processing of UDP SIP
> packets, but does not show a
On Mon, Jan 09, 2017 at 08:44:25AM -0800, Thufir Hawat wrote:
> I'm ok with Linux and am dipping my toe into Asterisk by running an AWS EC2
> instance. What is the most frequent usage for kamailio? Would it mainly be
> for SIP to SIP? So that the URI would be for a specific domain?
To set your
On Mon, Jan 09, 2017 at 07:25:36AM -0800, anfecora wrote:
> Hi Guys, can any one please remind me how to change the request line to
> reply with a fqdn instead of an ip.
That should be $rd
https://www.kamailio.org/wiki/cookbooks/4.2.x/pseudovariables#rd_-_domain_in_r-uri
__
On Tue, Jan 03, 2017 at 04:07:20PM +0200, Vladyslav Zakhozhai wrote:
> Daniel, thank you for your answer.
>
> You did not understand me completely. This is my fault.
>
> Let me put it this way. I want kamailio to handle NAT (fixing nat only from
> client's side) not being registrar itself. This a
On Tue, Jan 03, 2017 at 12:58:20PM +0200, Vladyslav Zakhozhai wrote:
> The main question here is about nat pinning. Acctording to module doc (
> http://kamailio.org/docs/modules/4.4.x/modules/nathelper.html) I need
> nathelper module and usrloc module.
>
> So I can proxy REGISTER requests to frees
On Fri, Dec 30, 2016 at 12:48:16PM +0100, Daren FERREIRA wrote:
> Kamalio is very good for a lot of things, and would be perfect if it
> might better manage topology hiding, as TOPOS begin to be able to do.
Have you tried the older topoh module? It works fine on my standalone
kamailio setups, but
On Thu, Dec 29, 2016 at 09:45:39PM +0100, Daren FERREIRA wrote:
> I wonder if that is a bug, related to RE-INVITE, or if there is any variable
> i can change to say topos which domain to be advertised on Contact headers?
I encountered the same problem.
http://lists.sip-router.org/pipermail/sr-use
On Fri, Dec 23, 2016 at 01:22:42PM +0100, Olle E. Johansson wrote:
> It’s been another great year for Kamailio and I’m proud to be part of the
> Kamailio development community.
> We’ve made great releases, had a great conference and overall done good stuff
> :-)
I have to say kamailio was very
On Thu, Dec 22, 2016 at 12:23:52PM -0500, Alex Balashov wrote:
> That just sounds like the rtpproxy is not being engaged, i.e. that the
> rtpproxy_manage() call is failing. When that happens, the SDP from .2
> will be passed through unaltered.
>
> The Kamailio log should give you some idea of why
On Tue, Dec 20, 2016 at 08:59:23AM +, Phil Lavin wrote:
> Is it possible to set the natping_interval to a different value, per
> registration, at the time the user registers? For example, user A has
> 10 seconds and user B has 60 seconds.
I took a look at the pinger a couple of months ago for
On Thu, Dec 15, 2016 at 12:36:06AM -0800, Gonzalo Gasca Meza wrote:
> When PhoneB dials 111, Kamailio converts 111...@sp1.com to j...@sp2.com
> Now I need to provide VM services to PhoneA. (Opposite direction)
> PhoneA calls voicemail but the calling number is j...@sp2.com
> I need to have SP1
On Tue, Dec 13, 2016 at 03:31:04PM -0500, Satish Patel wrote:
> We have currently dispatcher running for single domain, but in future
> we have more domain coming so i want to do multi domain dispatching
> for example
>
> dispatcher redirect request for foo.com to foo.registar and bar.com to
> bar
On Tue, Dec 13, 2016 at 03:25:12PM -0500, Satish Patel wrote:
> It works! but it doesn't work when i tried to use with wildcard (*) like
>
> if(!($ua =~ "*Foo")){
This is not a valid regular expression. A regexp wildcard is any
character (.) zero or more times (*): .*
Though I suspected above co
On Mon, Nov 28, 2016 at 01:15:03PM +0100, Daniel Tryba wrote:
> > UAC == SIP/TLS ==> Kamailio == SIP/UDP ==> FreeSWITCH
> >
> solution is to use Path on the frontend/loadbalancer.
According to this closed bug report it should work for
Kamailio/Freeswitch:
https://f
On Mon, Nov 28, 2016 at 01:00:37PM +0200, Vladyslav Zakhozhai wrote:
> UAC == SIP/TLS ==> Kamailio == SIP/UDP ==> FreeSWITCH
>
> My main problem is in Contact header of SIP packet which passes through
> Kamailio SIP proxy and remains unmodified.
>
> For example, REGISTER request. There is FreeSWI
On Fri, Nov 25, 2016 at 06:55:34PM +0200, Sergey Basov wrote:
> Is it safe to use remove_hf("User-Agent") without check if this header
> exist?
> or better use if(is_present_hf("User-Agent")) { remove_hf("User-Agent"); } ?
Just remove_hf is enough. is_present_hf/remove_hf might be more readable
th
On Fri, Nov 25, 2016 at 02:08:07PM +0200, Sergey Basov wrote:
> Hello All.
>
> I have some troubles with upstream sip switch.
> It ignores SIP packets which contains:
>
> User-Agent: FPBX-2.11.0(11.17.1)
> or
> Server: User-Agent: FPBX-2.11.0(11.17.1)
>
> If space is present before first "(" the
On Mon, Nov 21, 2016 at 07:14:57PM +, Andrei Mahalean wrote:
> I have my pilot DID setup and registered with Kamailio ok and I can make
> calls in and out through my PSTN gateway, but I'm unsure how to group all
> the other DID's so they are matched against the pilot DID.
You can use dbaliases
On Thu, Nov 10, 2016 at 04:41:36PM +0100, andrzej.ciupek-asterisk.edu.pl wrote:
> I have some UAC like Panasonic PBX, that send traffic to port 6060 that I am
> listening on,
> but that port isn't included to R-URI.
> I can see that only at tcpdump, or sip_trace from sip_capture module.
> Kamailio
On Sun, Nov 06, 2016 at 02:22:06AM -0800, Gonzalo Gasca Meza wrote:
> Currently in sample configuration script, seems to be that value: $avp(oexten)
> is used to redirect to VM, but in my case this value is null.
> I didnt find any documentation for this.
>
> *Questions:*
> a) What is $avp(oexten)
How to use the sql transformation?
https://www.kamailio.org/wiki/cookbooks/4.4.x/transformations#sql_transformations
has the following example:
xlog("$$rm = $rm = $(rm{s.sql})");
But adding this to the request_route and starting kamailio will fail:
ERROR: pv [pv_trans.c:2351]: tr_parse_strin
On Wed, Oct 05, 2016 at 11:40:49AM +0530, Infinicalls Infinicalls wrote:
> Oct 5 06:05:03 infinicalls-proxy1 /usr/local/sbin/kamailio[4840]:
> ERROR: dispatcher [dispatcher.c:788]: ds_warn_fixup(): failover
> functions used, but required AVP parameters are NULL -- feature
> disabled
In an other m
On Mon, Oct 03, 2016 at 03:35:59AM -0700, ycaner wrote:
> it is simple way to solve this and it is softphone product problem. If i did
> configuration as you said , it consumes so much socket , CPU and etc. in
> future.
I can't imagine this will be a significant amount of cpu/traffic. I can
only s
On Mon, Oct 03, 2016 at 12:22:22PM +0300, Yasin CANER wrote:
> When i have a look ietf3261 , i couldnt find any thing this flow. Do you
> have any idea about it?
>
> I am looking forward to your suggestions.
It is simple, never trust the endpoint and always assume there is NAT or
a statef
On Fri, Sep 30, 2016 at 10:29:16PM +0530, Infinicalls Infinicalls wrote:
> Hi Daniel,
> This seems to be ideal setup for my requirement. Would you mind
> sharing the kamailio.cfg file with me (with all the important data
> masked) ?
Not much important stuff to censor.
LBIPADDR is the ipadress of
On Fri, Sep 30, 2016 at 09:08:50AM +0530, Infinicalls Infinicalls wrote:
> Edge proxies is a nice suggestion. Thanks.
>
> Can I implement, as suggested here,
> https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/
> to setup a proxy and direct requests to the registr
On Thu, Sep 29, 2016 at 01:53:14PM +0530, Infinicalls Infinicalls wrote:
> > You can either figure out which host was used to register by looking at
> > the socket value in the location table (using sqlops/avp_db_query) and
> > send the INVITE to the other host if socket doesn't contain a local
> >
On Mon, Sep 26, 2016 at 05:18:32PM +0530, Infinicalls Infinicalls wrote:
> I am sorry, I am not sure how to implement Brandon's method. Isn't
> there any simpler solution to achieve this? Could someone elaborate me
> more on this or point me to relevant links?
>
> My problem is simple. I have 2 h
On Fri, Sep 23, 2016 at 03:59:20PM +0200, Loic Chabert wrote:
> - I have some sbc for one destination. If one destination fail, kamailio
> try to second one etc..
> - If no SBC is available for this destination (for exemple, all are down),
> i want to send my call to an error server (asterisk), pla
On Thu, Sep 22, 2016 at 09:58:33AM -0400, Alex Balashov wrote:
> Normally, we just force_rport() on all incoming requests so that we reply to
> the real source port of the request, since most endpoints on this
> installation are NAT'd.
>
> However, occasionally we run into a scenario where an ALG
On Mon, Sep 19, 2016 at 11:29:47AM +, mjbemail2000-s...@yahoo.co.uk wrote:
> How do I access the Contact header in the reply message?
$T_rpl($ct)
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.o
With
kamctl fifo nh_enable_rtpp udp:rptipaddr:7723 1
kamailio will begin to crash:
Sep 19 16:00:08 sipcluster-backend2 kernel: [1658593.859081] kamailio[22992]:
segfault at 4 ip 7f5a4d44e11f sp
7ffcac5293e0 error 4 in rtpengine.so[7f5a4d446000+41000]
Sep 19 16:00:12 sipcluster-backend2 /
On Sun, Sep 18, 2016 at 11:51:42AM +0200, Torsten Hantzsche wrote:
> Bastian, thanks for the explanation. I have Kamailio running on FreeBSD
> and never thought of a difference in the routing table compared to Linux.
> The option WITH_IPAUTH did the trick.
There is no difference in routing to the
On Fri, Sep 16, 2016 at 05:46:35PM +0200, Daniel Pocock wrote:
> The database was not running and Kamailio refused to start.
>
> Would it be better for Kamailio to start anyway and go into a loop
> trying to connect to the database, just as if auto_reconnect was set?
I'd rather have kamailio down
On Fri, Sep 16, 2016 at 07:54:20AM +0100, Eric Koome wrote:
> Hi all - my Kamailio - 4.1.6 is receiving this particular structured
> INVITES from multiple IPs, and for some reason it is not requesting
> authentication. I have AUTH & IPAUTH modules in use for two years now,
> but this is bypassing t
On Tue, Sep 13, 2016 at 12:42:23PM -0300, Valter Nogueira wrote:
> I use Asterisk for SIP and Media Proxy. Despite the fact that Asterisk is
> not a SIP Proxy at all.
[...]
> Well, I understand that I have to use some kamailio modules, like auth,
> dialplan, rtpproxy and db_mysql.
>
> What make me
On Mon, Sep 05, 2016 at 09:15:41AM +0200, Daniel-Constantin Mierla
wrote:
> > System memory, the systems OOM killer is triggered. I have looked at
> > the "kamcmd mod.stats all shm" stats and they don't change much.
> > I'll have to look into memory debugging before I can give more info.
> >
> Inte
On Thu, Sep 08, 2016 at 04:50:50PM +0200, Federico Cabiddu wrote:
> The issue with dispatcher is that, in case of TCP transport, you cannot set
> the sending socket for the same reason.
> Basically, each time a client TCP socket is open by Kamailio, the SO select
> the port, due to the lack of supp
On Tue, Sep 06, 2016 at 12:53:35PM +0200, Daniel-Constantin Mierla wrote:
> > -Route to the callee has trailing nulls (starting from the first ACK to
> > the callee (packet 7 in pcap))
> > [...]
> Couldn't spot the real reason to have those zeors in the route value, I
> added some trimming before i
On Thu, Sep 08, 2016 at 03:38:32PM +0300, Yuriy Gorlichenko wrote:
> I didnt thought about keepalive. I suppose it can help.
Better than using qualify in asterisk is to use the dispatcher module in
kamailio. The idea is the same, but more configurable and it is just 1
keepalive mechanisme so the a
On Thu, Sep 08, 2016 at 02:43:03PM +0300, Yuriy Gorlichenko wrote:
> Like User sends registration, kamailio just Transcoding this request to TCP
> and then resend this registration packet to Asterisk.
> With this example asteisk must originate all PACKETS to TCP port of
> kamailio but it tries to s
On Thu, Sep 08, 2016 at 02:43:03PM +0300, Yuriy Gorlichenko wrote:
> yes. Thats will be great because in some system design it must use same
> port that listening for sendinf like in UDP for example for transcoding SIP
> over WebSocket to SIP over TCP and masking registration behind thanscoder.
>
On Thu, Sep 08, 2016 at 11:16:29AM +0200, Federico Cabiddu wrote:
> about this subject: linux kernel starting from 3.9 introduced SO_REUSEPORT
> which allows reusing TCP sockets.
> It could be interesting supporting this in Kamailio. I worked on a patch
> for this, I can open a PR and start a discu
On Wed, Sep 07, 2016 at 08:35:30PM +0300, Yuriy Gorlichenko wrote:
> Before to send to asteisk any packet i added
> $fs=ip.add.re.ss:port
>
> Also discribed
> listen=tcp:ip.add.re.ss:port
>
> But kamailio send outgoing packets from random prot throug TCP
> Presume i configured 5060 port but it se
On Tue, Sep 06, 2016 at 12:53:35PM +0200, Daniel-Constantin Mierla wrote:
> > -Route to the callee has trailing nulls (starting from the first ACK to
> > the callee (packet 7 in pcap))
> > [...]
> Couldn't spot the real reason to have those zeors in the route value, I
> added some trimming before i
On Mon, Sep 05, 2016 at 11:30:33AM +0200, Camille Oudot wrote:
> > Just out of curiosity, but is kamailio notified (some event in the
> > config) that a socket died when using tcp keepalives on the OS level?
>
> If the tcpops module is loaded, the tcp:closed event route will be
> called:
>
> http
On Mon, Sep 05, 2016 at 09:19:59AM +0200, Daniel-Constantin Mierla
wrote:
> > I would be interested in if it is possible to detect such situations
> > on Kamailio side when for example a message cannot be delivered
> > because of such network conditions like airplane mode?
> You can use failure_rou
On Fri, Sep 02, 2016 at 07:13:13PM +0200, Klaus Darilion wrote:
> I see 2 solutions:
> a) fix_nated_contact
...
> b) if the Contact header must be preserved (strict clients) or if the
> registrar does not apply NAT traversal, then it is necessary to do heavy
> message rewriting and registration in
On Fri, Sep 02, 2016 at 02:01:09PM +0200, Daniel-Constantin Mierla wrote:
> > -When enabling topos module, kamailio leaks memory like crazy at a rate
> > of about 1 GB in 2 hours without any calls (only OPTIONS and replies)
> >
> What kind of memory is leaking? Private (pkg), shared or system memor
On Thu, Sep 01, 2016 at 03:07:46PM +0200, Daniel-Constantin Mierla wrote:
> I pushed a new commit, aiming to avoid overwriting record route set with
> session updates. Let me know if it makes the difference.
The same scenario (callee wants sessiontimers refreshed by the uac)
works kind of. ACKs ar
On Wed, Aug 31, 2016 at 03:56:10PM +0200, Daniel Tryba wrote:
> Attached a pcap showing this problem (the initial INVITE to the callee
> is missing for some reason (fragmented?)) and the topos related
> debugging from the timeperiod.
How hard is it to not forget to attach them when wri
> I'll try to test all call scenarios tomorrow.
>
> Thanks for the quick fix.
Well, I found a new bug. If the callee wants session-timers and sets the
uac as refresher, the ACK on the 200 OK of the session-timer re-INVITE
isn't handled correctly. The ACK will be send directly to the contact of
th
On Wed, Aug 31, 2016 at 01:08:22AM +0200, Andrzej Kaczmarczyk wrote:
> I have common, cheap TL-WR1043ND V1 router.
> For private purposes I've set kamailio voip on my own server INSIDE my LAN.
> Is it possible to connect to that server from outside, through openWRT
> router?
> Many tries, no succes
On Tue, Aug 30, 2016 at 03:15:14PM +0200, Daniel-Constantin Mierla wrote:
> I pushed a new commit, can you give it another try?
>
A quick test confirms that the previous scenario is now handled
correctly. ACK and BYE are now routed correctly (not end to end based on
contact).
I'll try to test al
On Mon, Aug 29, 2016 at 05:23:46PM +0200, Daniel-Constantin Mierla wrote:
> can you try to change the module exports for uac module and allow
> uac_auth() for REQUEST_ROUTE or BRANCH_FAILURE_ROUTE (not sure right now
> by heart which one is required) and see if it actually works. Then the
> flags c
On Fri, Aug 26, 2016 at 07:04:20PM -0400, Mike Patterson wrote:
> As far as the register question, by default authentication is off.
Between the fully populating a subscriber table and disabling
authentication there is also
http://kamailio.org/docs/modules/4.4.x/modules/auth.html#auth.f.pv_www_au
On Fri, Aug 26, 2016 at 01:55:13PM +0200, Daniel-Constantin Mierla wrote:
> Finally some time to look at the source code for it ... can you run with
> higher debug level for topos and grab the message printed by the module
> that is like "... compacted headers - a_rr: ... b_rr: ..."?
Attached a p
On Fri, Aug 26, 2016 at 01:58:57PM +0200, Daniel-Constantin Mierla wrote:
> try to see if it works to do the authentication in the branch failure
> event route. There you can do processing as soon as the 401 arrives --
> it has to be tried to see if uac auth works fine there, if not probably
> need
On Fri, Aug 26, 2016 at 07:39:02AM -0400, Mike Patterson wrote:
> Thank you for your response. Can you tell me what value I need to put
> in is_in_subnet? Is this the local network of the Kamailio server or
> is it the network of the down-stream sip server?
It is the subnet (or ip) of your downs
On Thu, Aug 25, 2016 at 01:34:30PM -0400, Mike Patterson wrote:
> Thank you. I am reviewing the doc. I am new to Kamailio so it will
> take some time for me.
Good luck then :)
Here is my slightly redacted config for a setup similar to yours (which
is just a loadbalancer/proxy), which uses both
On Thu, Aug 25, 2016 at 09:38:12AM -0400, Mike Patterson wrote:
> I have configured Kamailio to pass a registration request to another VoIP
> provider. The reason I am doing this is to provide a sip port for users
> where the ISP is blocking SIP. I am able to pass the registrations to the
> VoIP
Trying to implement a way to authenticate outgoing INVITEs to endpoints
that need/want to. uac_auth in the initial failure route works fine when
there is only 1 registered user to send the INVITE to. But with at least
2 locations are available and 1 accepts the INVITE without
authentication, the 40
On Wed, Aug 24, 2016 at 03:36:16PM +, Phil Lavin wrote:
> We need to check if a 1XX reply is part of an INVITE transaction that
> was forked to 2 or more phones. We figure that we can do a location
> lookup, based on the to header of the 1XX... but is there a better
> way?
It looks like can te
On Mon, Aug 22, 2016 at 12:30:18PM +0200, Daniel Tryba wrote:
> > OK. I will have to look at the source code?
> >
> > Btw, did you try with longer timeout? Still the same? If you change the
> > modparam for the timeout, is it used? These will help to narrow down of
>
On Mon, Aug 22, 2016 at 09:27:25AM +0200, Daniel-Constantin Mierla wrote:
> > Ringing timeout, the task is to forward the call to an other destination
> > (or voicemail).
> >
> OK. I will have to look at the source code?
>
> Btw, did you try with longer timeout? Still the same? If you change the
>
On Mon, Aug 22, 2016 at 09:19:52AM +0200, Daniel-Constantin Mierla wrote:
> can you try with git master branch? There were some fixes there which
> were not ported yet (they will get into next 4.4.x release, which will
> be sometime in the next 2 weeks).
Just tried it, makes no difference. The pat
On Fri, Aug 19, 2016 at 09:37:15AM +0200, Daniel-Constantin Mierla wrote:
> > If an enduser INVITEted over TCP t_set_fr(t) with t < 30s have no
> > effect, the INVITE will always timeout after 30s. What is controlling
> > this 30s? The only default at 30s is the fr_timer, which should have no
> > p
Looking into the topos module of 4.4.2 I see something unexpected, ACK and
BYE are routed directly to the Contact in the 200 OK on an INVITE where
the 200 OK has a Record-Route.
Without the topos module the message flow is as expected:
109.235.32.57<->185.61.68.106<->109.235.32.55
With topos the A
On Thu, Aug 18, 2016 at 03:50:53PM +0200, Alex Hermann wrote:
> > I want to upgrade my cluster from 4.3 to 4.4, beginning with the spare
> > for live testing. But since there is a change to the trusted table
> > (which we don't use, only address) this isn't possible without some
> > tricks. There i
I want to upgrade my cluster from 4.3 to 4.4, beginning with the spare
for live testing. But since there is a change to the trusted table
(which we don't use, only address) this isn't possible without some
tricks. There is no override for skipping the version check on any
permissions table.
I need
If an enduser INVITEted over TCP t_set_fr(t) with t < 30s have no
effect, the INVITE will always timeout after 30s. What is controlling
this 30s? The only default at 30s is the fr_timer, which should have no
part in the timeout between an 1xx and a final answer for an INVITE.
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