> Or, to be put it another way: Just because you can, doesn't mean you should.
> There are quite a few things in OpenSIPS that prompt this thought.
please elaborate. many people have been wondering for some time now
what the technical differences between these projects might be.
there's also kamctl ul show.
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?
>
> Cheers,
> Daniel
>
>
> On Sun, Sep 8, 2013 at 11:00 PM, hiro <23h...@gmail.com> wrote:
>
>> Ok, I compiled the latest rtpproxy.so, the problem persists: the
>> interesting facts are: the callee only answers with sdp port once, in
>> the session
progress by
sending a prack to the callee itself followed by a 200 ok to the
caller that includes sdp but has CSeq: 893961 PRACK which never got
requested by the caller though.
On 9/4/13, hiro <23h...@gmail.com> wrote:
> For this installation I used the .deb from http://deb.kamailio.org/kamaili
EN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 17:01:35 Aug 19 2013 with gcc 4.7.2
On 9/4/13, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 8/29/13 10:22 PM, hiro wrote:
>>
rtpproxy port in session progress, but then forgets about it for
the 200 OK.
Attached is a tree overview and the conversations of each phone with kamailio.
hiro
|Time | 192.168.5.86 | 192.168.5.149
|
| | | kamailio-ip
I've been trying to make srtp->srtp work with kamailio, rtpproxy and
two symbian phones.
rtpproxy does not receive any rtp/sdp packets. I'm still trying to
debug what makes it fail.
Without tls and srtp everything works, with either results are varied.
Before I waste more time: Should kamailio+rtpp
had the same issue here. you have to manually set $du=$null, else it
doesn't get reset for the failure branch.
On 7/23/13, LAA wrote:
> Hi all,
>
> I'm running Kamailio 3.0.0, with SEMS integration as Media Server for Voice
> mail. I'm trying to get a configuration to forward calls on busy to voi
and check the RURI, and Route headers.
> Maybe the show some bugs. Also, if you manually apply routing to in-dialog
> requests (e.g. forcing the send socket) make sure to not make mistakes.
>
> regards
> Klaus
>
>
> On 12.07.2013 17:52, hiro wrote:
>>
>> hi
>>
hi
I have set up tls on kamailio successfully, but when I relay a TLS
client to an other proxy ($ru = "sip:" + $rU + "@" +
"127.0.0.1:5070;transport=udp";) via udp I get error messages.
>From the timing it seems like this warning always appears after I get
200OK from proxy, the Invite gets sent c
8 is a Nokia E72 calling in via kamailio. After the Invite I don't get
any other packets from the phone any more. It seems like the TCP
connection gets killed completely by something. It only starts working
again after I reboot the phone and it is completely reproducable.
14:06:51.457926 IP 82-171
same name are allowed in SIP
> URI, in headers they are not.
>
> Try to put:
>
> if(t_is_branch_route())
>
> as condition for add_rr_param(...)
>
> Cheers,
> Daniel
>
> On 6/5/13 7:18 PM, hiro wrote:
>> I'm just using the default kamailio.cfg for ka
rr_param() and see how many times it is
> executed for a branch (if executed in request_route, then it is for all
> branches at least one time).
>
> Cheers,
> Daniel
>
> On 6/4/13 10:57 PM, hiro wrote:
>> actually, I now see my last message is wrong.
>> I've co
at=yes two times here:
Record-Route:
Record-Route:
On 6/4/13, hiro <23h...@gmail.com> wrote:
> ok. right now from tcpdump I can see the session progress and OK
> messages are sent to the correct ip:port of my phone, but either the
> phone doesn't receive it or it doesn't p
y kamailio, but I'm not sure:
Call id and Cseq are the same as in RINGING, but contact header has
freeswitch's IP (on the same server as kamailio)
Contact:
Could that ever work this way?
On 6/4/13, Daniel Tryba wrote:
> On Tuesday 04 June 2013 12:07:35 hiro wrote:
>> Sometimes it
re" to
keep me going :P
I will test with xlog when I can test at home again which would at
least exclude NAT issues.
On 6/4/13, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 6/2/13 10:57 PM, hiro wrote:
>> I'm still thinking about this issue and wondering:
>> is
to clarify, it doesn't normally happen, just in some rare
non-reproducable cases after i played with the config in some very
random ways. have been playing to corner that case for many hours, but
going to sleep now.
On 6/2/13, hiro <23h...@gmail.com> wrote:
> I'm still thinki
I'm still thinking about this issue and wondering:
is it even compliant to the RFC to go directly from ringing to session
progress and then OK? Because that's what freeswitch is answering with
when I try to relay the call to it's voicemail when user is busy in
kamailio.
On 6/
ierla wrote:
>
> On 5/19/13 2:05 PM, hiro wrote:
>> i'm trying to use the example kamailio.cfg to route to voicemail
>> server on busy or decline.
>> Only thing I did was adding decline code to t_check_status("486|408"),
>> enabling the preprocessor variabl
i'm trying to use the example kamailio.cfg to route to voicemail
server on busy or decline.
Only thing I did was adding decline code to t_check_status("486|408"),
enabling the preprocessor variable for voicemail and changing the
voicemail host and port to my voicemail server.
No requests arrive on
makes sense, cool. now i just have to find a version of rtpproxy that
supports this stuff :)
On 5/14/13, Andres wrote:
> On 5/13/2013 2:17 PM, hiro wrote:
>> It doesn't seem to be the router/NAT's problem though, as the Nokia
>> itself binds to the right port at first
good, it would only be
problematic if there are multiple concurrent calls from the same
(perhaps NATted) IP, right?
On 5/13/13, Andres wrote:
> On 5/11/2013 4:29 PM, hiro wrote:
>> using kamailio-4.0.1_src.tar.gz with rtpproxy and a nokia e72 behind
>> NAT registered via UDP I get
broken by your E72.
>
> Leo
>
> On 12 May 2013, at 12:39, hiro <23h...@gmail.com> wrote:
>
>> On 5/12/13, Leo Brown wrote:
>>> Hi
>>>
>>> I don't know the e72, but if you do a `tcpdump -A -s0 port 5060` we can
>>> see
>>>
On 5/12/13, Leo Brown wrote:
> Hi
>
> I don't know the e72, but if you do a `tcpdump -A -s0 port 5060` we can see
> the SDP data to determine the issue
>
> RTP should be sent to the address defined in SDP unless a symmetric RTP
> setup is used for auto NAT traversal- but I believe only Asterisk pr
such packet? it doesn't happen
when i use sip over tcp. is my analysis right, is it a bug in
rtpproxy?
greetings
hiro
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