18 Apr
2012
18 Apr
'12
10:01 a.m.
Wait.. from your trace I see that you are running Kamailio behind NAT. Is
this the case? If yes, things have to be changed in Kamailio's config,
especially in record-routing and external IP address advertising. If this
is the case, my previous interpretation is somehow wrong..
So your Kamailio's IP is 10.62.99.103, NATed to 23.22.35.43.
In your conf, you should do the following:
- set advertised IP address (external address) :
advertised_address=23.22.35.43
- replace record_route( ) with record_route_preset("23.22.35.43")
(I'm not sure if there's anything else to configure as well)
There is also a weird IP address in your trace, 159.159.159.1598 (4 digits
in the last octet). You should take a look at that as well.
Reda
On Wed, Apr 18, 2012 at 09:42, Reda Aouad <reda.aouad(a)gmail.com> wrote:
Hi Saul,
X-Lite is sending 200OK with its IP @ in the contact header. Linphone is
sending the ACK directly to X-Lite without passing by Kamailio. I suspect
that since there is no direct TCP connection between the clients (each has
its own TCP connection to Kamailio), the ACK is discarded by the TCP/IP
stack of the X-Lite machine. X-Lite keeps then re-sending 200OK because it
doesn't receive an ACK. Kamailio is timing out and terminating the call.
(But I may be wrong about all this, it's just my interpretation)
I had the same problem over UDP, but I can't remember what I did to solve.
Maybe others can help? I'll take a look on my config later to see if I can
remember..
You can also make a call in the other direction (X-Lite > Linphone) to run
wireshark on the X-Lite machine and take a closer look at the ACK that it
is sending to see what is wrong with it..
Reda
On Tue, Apr 17, 2012 at 22:11, Saul Waizer <saulwaizer(a)gmail.com> wrote:
Thank You Reda,
Here is the entire trace of the 30 second call with ngrep, please let me
know if this helps:
http://pastebin.com/uJVJDuuv
On Tue, Apr 17, 2012 at 3:23 AM, Reda Aouad <reda.aouad(a)gmail.com> wrote:
Hi,
This is what I wrote on another thread for the same problem.
---------
Well, 30 seconds seems to be the timeout of your SIP client: I suspect
your client sends a 200OK but doesn't receive the ACK for it, and hence
times out 30 seconds after and terminates the call. You can run ngrep
to trace the call and see where the ACK is routed, and if it is routed to
the correct address, if anything is not correct in its header. Personally,
I run wireshark over an X11 tunnel for the GUI which helps a lot. If you
provide ngrep trace, we may help you better.
I had the exact same problem once, but don't remember what the reason
was.. maybe incorrect loose routing section or incorrect record-route or
incorrect address in contact header of 200OK..
---------
Reda
On Mon, Apr 16, 2012 at 20:51, Saul Waizer <saulwaizer(a)gmail.com> wrote:
Daniel,
thank you for the suggestion, I followed the tutorial and got it up and
running, however I am still experiencing the same issue, dropped calls
after 30 seconds. Furthermore the output of tshark -i eth0 -R sip does not
show any errors such as too many hops, everything looks very clean but I
still cant get more than 30 seconds out.
I have a default config with mysql enabled and nat. rtproxy is running
as well.
Any help is greatly appreciated.
On Mon, Apr 16, 2012 at 1:31 PM, Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
> Hello,
>
> ngrep trace (I haven't seen any yet in the thread) of such call can
> help seeing if Record-/Route and Contact headers are properly set and
> maintained during the call. It might be a broken RR handling in a device or
> a wrong update of contact address.
>
> On the other hand, in the config, I saw presence being loaded -- if
> you need that, then start with kamailio 3.2.x, here is a tutorial:
>
> http://www.kamailio.org/wiki/install/3.2.x/git
>
> It comes with a default config file where is very easy to enable nat
> traversal as well as presence handling -- just read the top of config file
> and add the appropriate #!define directives.
>
> 1.3.x is anyhow way too old...
>
> Cheers,
> Daniel
>
>
>
> On 4/16/12 7:16 PM, Stoyan Mihaylov wrote:
>
> I am using nathelper and rtpproxy.
> We had no serious problems with those modules.
> My idea was to overwrite wrong addresses in ACK and BYE packages,
> because they kept reentering kamailo, instead of being forwarded where they
> have to go.
>
> By the way - you can add:
> modparam("htable", "htable",
"forw=>size=8;autoexpire=7200;")
> modparam("htable", "htable",
"mustforw=>size=8;autoexpire=7200;")
>
> On Mon, Apr 16, 2012 at 8:03 PM, Saul Waizer <saulwaizer(a)gmail.com>wrote;wrote:
>
>> Thank you Stoyan
>> *
>> *
>> I added the routing config as you suggested but it still drops the
>> call after 30 seconds.
>>
>> Any other suggestions?
>>
>> Note: my nathelper module is commented out because I can't get it
>> to work, here is my mod list:
>>
>> ###### Modules Section ########
>>
>> #set module path
>> mpath="/usr/lib/openser/modules/"
>>
>> /* uncomment next line for MySQL DB support */
>> loadmodule "mysql.so"
>> loadmodule "sl.so"
>> loadmodule "tm.so"
>> loadmodule "rr.so"
>> loadmodule "maxfwd.so"
>> loadmodule "usrloc.so"
>> loadmodule "registrar.so"
>> loadmodule "textops.so"
>> loadmodule "mi_fifo.so"
>> loadmodule "uri_db.so"
>> loadmodule "uri.so"
>> loadmodule "xlog.so"
>> loadmodule "acc.so"
>>
>> loadmodule "auth.so"
>> loadmodule "auth_db.so"
>>
>> #loadmodule "domain.so"
>> loadmodule "presence.so"
>> #loadmodule "presence_xml.so"
>>
>> # !! Nathelper
>> #loadmodule "nathelper.so"
>> #loadmodule "nat_traversal.so"
>> #loadmodule "rtpproxy.so"
>> #loadmodule "dialog.so"
>>
>> Thank You
>>
>>
>> On Mon, Apr 16, 2012 at 12:03 PM, Stoyan Mihaylov <
>> stoyan.v.mihaylov(a)gmail.com> wrote:
>>
>>> Some time ago, I had similar problem.
>>> This was my solution:
>>> if(is_method("INVITE")){
>>> ds_select_dst("1","4");
>>> $sht(forw=>$ft)=$du;
>>> sl_send_reply("100","Trying");
>>> route(RELAY);
>>> exit();
>>> }
>>>
>>> if ( is_method("ACK|BYE") ) {
>>> if ( t_check_trans() ) {
>>> t_relay();
>>> exit;
>>> } else {
>>> if(($sht(forw=>$ft))=~$td){
>>> $du=$sht(forw=>$ft);
>>> }else if((($td=="sip.mydomain.com
>>>
")||($td=="ip.of.my.domain"))&&($si=="ip.of.my.domain")){
>>> $du=$sht(forw=>$ft);
>>> }
>>> t_relay();
>>> exit;
>>> }
>>>
>>>
>>>
>>> On Mon, Apr 16, 2012 at 6:18 PM, Saul Waizer
<saulwaizer(a)gmail.com>wrote;wrote:
>>>
>>>> Thank you Davy, I ran tshark and found the following: (masked ip
>>>> addresses to protect the innocent and trimmed the logs)
>>>>
>>>> 8.250307 300.300.300.300 -> 10.12.47.96 SIP Request: REGISTER
>>>> sip:sip.mydomain.com
>>>> 8.251503 10.12.47.96 -> 300.300.300.300 SIP Status: 200 OK (1
>>>> bindings)
>>>> 35.942381 300.300.300.300 -> 10.12.47.96 SIP Request: REGISTER
>>>> sip:sip.mydomain.com
>>>> 35.943011 10.12.47.96 -> 300.300.300.300 SIP Status: 200 OK (2
>>>> bindings)
>>>> 44.564112 300.300.300.300 -> 10.12.47.96 SIP Request: REGISTER
>>>> sip:sip.mydomain.com
>>>> 44.565402 10.12.47.96 -> 300.300.300.300 SIP Status: 200 OK (2
>>>> bindings)
>>>> 75.483859 400.400.400.400 -> 10.12.47.96 SIP Request: REGISTER
>>>> sip:sip.mydomain.com
>>>> 75.484572 10.12.47.96 -> 400.400.400.400 SIP Status: 200 OK (1
>>>> bindings)
>>>> 75.623702 400.400.400.400 -> 10.12.47.96 SIP Request: SUBSCRIBE
>>>> sip:dan@sip.mydomain.com
>>>> 75.624258 10.12.47.96 -> 400.400.400.400 SIP Request: SUBSCRIBE
>>>> sip:dan@400.400.400.400:23242;rinstance=f6f0a9561214235b
>>>> 75.763700 400.400.400.400 -> 10.12.47.96 SIP Status: 489 Event
>>>> Package Not Supported
>>>> 75.764128 10.12.47.96 -> 400.400.400.400 SIP Status: 489 Event
>>>> Package Not Supported
>>>> 75.802420 400.400.400.400 -> 10.12.47.96 SIP Request: SUBSCRIBE
>>>> sip:saul@sip.mydomain.com
>>>> 75.802904 10.12.47.96 -> 400.400.400.400 SIP Status: 404 Not Found
>>>> 93.834928 300.300.300.300 -> 10.12.47.96 SIP Request: REGISTER
>>>> sip:sip.mydomain.com
>>>> 93.835553 10.12.47.96 -> 300.300.300.300 SIP Status: 200 OK
>>>> (2 bindings)
>>>> 127.107903 400.400.400.400 -> 10.12.47.96 SIP/SDP Request: INVITE
>>>> sip:tom@sip.mydomain.com, with session description
>>>> 127.109021 10.12.47.96 -> 400.400.400.400 SIP Status: 100 Giving a
>>>> try
>>>> 138.138242 10.12.47.96 -> 300.300.300.300 SIP/SDP Request: INVITE
>>>> sip:tom@300.300.300.300:61162;transport=tcp;line=90fb23e32c7545e,
>>>> with session description
>>>> 138.306651 300.300.300.300 -> 10.12.47.96 SIP Status: 100 Trying
>>>> 138.392632 300.300.300.300 -> 10.12.47.96 SIP Status: 101 Dialog
>>>> Establishement
>>>> 138.393101 10.12.47.96 -> 400.400.400.400 SIP Status: 101 Dialog
>>>> Establishement
>>>> 138.460232 300.300.300.300 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.460780 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.461348 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.462717 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.463381 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.463959 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.464416 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.543567 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.547330 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.551563 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.551865 300.300.300.300 -> 10.12.47.96 SIP Status: 180 Ringing
>>>> 138.554340 10.12.47.96 -> 400.400.400.400 SIP Status: 180 Ringing
>>>> 138.560084 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.563560 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.575640 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.579557 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.587442 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.591558 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.596417 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.599565 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.604589 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.607561 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.613004 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 138.615556 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.255562 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.265531 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.267570 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.276852 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.279558 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.289309 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.291562 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.300859 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.303560 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.312556 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.315563 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.324327 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.327560 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.336426 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.339562 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.348431 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.351564 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.360179 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.363554 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.371931 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.487555 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.494448 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.495440 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.495487 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.495528 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.495557 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.495583 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.495628 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.495666 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.495787 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.495844 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.495888 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.496068 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.496097 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.496107 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.496438 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.496468 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.496471 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.496473 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.496567 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.732919 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.733655 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.735559 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.740030 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.743552 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 155.835477 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.835537 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.835988 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.836198 10.12.47.96 -> 500.500.500.500 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 155.836400 500.500.500.500 -> 10.12.47.96 SIP Request: OPTIONS
>>>> sip:500.500.500.500:5060
>>>> 156.146827 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.150092 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.162814 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.167551 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.168667 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.171363 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.171551 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.175549 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.181077 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.183553 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.186055 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.187549 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.189991 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.191553 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.193930 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.195550 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.198570 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.199550 500.500.500.500 -> 10.12.47.96 SIP Status: 483 Too Many
>>>> Hops
>>>> 156.204172 10.12.47.96 -> 500.500.500.500 SIP Status: 483 Too Many
>>>> Hops
>>>> 208.371477 300.300.300.300 -> 10.12.47.96 SIP Request: REGISTER
>>>> sip:sip.mydomain.com
>>>> 208.372297 10.12.47.96 -> 300.300.300.300 SIP Status: 200 OK (4
>>>> bindings)
>>>> 351.118527 300.300.300.300 -> 10.12.47.96 SIP Request: REGISTER
>>>> sip:sip.mydomain.com
>>>> 351.120474 10.12.47.96 -> 300.300.300.300 SIP Status: 200 OK (4
>>>> bindings)
>>>> 383.956615 400.400.400.400 -> 10.12.47.96 SIP Request: SUBSCRIBE
>>>> sip:saul@sip.mydomain.com
>>>> 383.957274 10.12.47.96 -> 400.400.400.400 SIP Status: 404 Not Found
>>>> 493.457746 300.300.300.300 -> 10.12.47.96 SIP Request: REGISTER
>>>> sip:sip.mydomain.com
>>>> 493.458864 10.12.47.96 -> 300.300.300.300 SIP Status: 200 OK (4
>>>> bindings)
>>>>
>>>> I am new to opensips so I am not too familiar with the routing
>>>> logics, a google search on that error suggests that there is a problem
with
>>>> the route config where its creating a loop exhausting the Max Hops. The
way
>>>> I configured my clients uses my server as a proxy.
>>>>
>>>> Any help is greatly appreciated!
>>>>
>>>> On Mon, Apr 16, 2012 at 1:59 AM, davy van de moere <
>>>> davy.van.de.moere(a)gmail.com> wrote:
>>>>
>>>>> 38 seconds sounds pretty close to 30 seconds. Could those 38
>>>>> seconds be a 30 seconds after the actual answer packet?
>>>>>
>>>>> If so, you might want to look at ACK and OK packets not arriving
>>>>> correctly because of NAT, wrong IP selection in openser , etc...
>>>>>
>>>>> A simple tshark might help you out to debug from a higher
>>>>> perspective : tshark -i eth0 -R sip
>>>>>
>>>>> good luck!
>>>>>
>>>>> Op 16 april 2012 05:11 schreef Saul Waizer
<saulwaizer(a)gmail.com>het volgende:
>>>>>
>>>>>> Greetings list,
>>>>>>
>>>>>> I am experiencing a strange behavior with openser 1.3.2 running
>>>>>> on ubuntu 10. I have a basic configuration (see bellow) and i am
using
>>>>>> Linphone for iPad as my client. I have 2 users registered and I
am able to
>>>>>> place calls no problem. The problem is that the calls (audio or
A/V) drop
>>>>>> after 38 seconds exactly, this behavior is pretty consistent, 38
seconds is
>>>>>> all I can get. There is no firewall in front of the clients.
>>>>>>
>>>>>> Here is my configuration, ip addresses changed to protect the
>>>>>> innocent:
>>>>>>
>>>>>> http://pastie.org/private/x1ck8rxjcxv6hl44hrmqg
>>>>>>
>>>>>> You can see the logs of the call here (the majority):
>>>>>>
>>>>>> http://pastie.org/private/4fj5efpbsrxan8plzqvfza
>>>>>>
>>>>>> Am I missing something or is there anything that needs to be
>>>>>> changed in the routing/configuration to achieve basic
functionality?
>>>>>>
>>>>>> Thank you in advance!
>>>>>>
>>>>>> _______________________________________________
>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>>>>> mailing list
>>>>>> sr-users(a)lists.sip-router.org
>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>>> list
>>>>> sr-users(a)lists.sip-router.org
>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>> list
>>>> sr-users(a)lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>> list
>>> sr-users(a)lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>> list
>> sr-users(a)lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla
> Kamailio Advanced Training, April 23-26, 2012, Berlin,
Germanyhttp://www.asipto.com/index.php/kamailio-advanced-training/
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users