Hey,
Here is an expert from the /var/log/messsages :
Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: New request -
M=PUBLISH RURI=sip:group1.105@mydomain.tld F=sip:group1.105@mydomain.tld
T=sip:group1.105@mydomain.tld IP=72.55.182.125
ID=7bddec01-28924(a)72.55.182.125
Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: Publish
rewritten forwarded for ext=group1.105 ruri=sip:group1.105@mydomain.tld
duri=<null> body=<?xml version="1.0"?> <dialog-info
xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full"
entity="sip:group1.105@mydomain.tld"> <dialog
id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld"
call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld"
local-tag="4fbrxu9548" remote-tag="as4fb63283"
direction="recipient"> <state>early</state> <remote>
<identity>sip:103@mydomain.tld</identity> <target
uri="sip:103@mydomain.tld"/> </remote> <local>
<identity>sip:group1.105@mydomain.tld</identity> <target
uri="sip:group1.105@mydomain.tld"/> </local> </dialog> </dialog-info>
Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: New request -
M=PUBLISH RURI=sip:group1.105@mydomain.tld F=sip:group1.105@mydomain.tld
T=sip:group1.105@mydomain.tld IP=72.55.182.125
ID=7bddec01-28924(a)72.55.182.125
Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: Publish accepted
for ext=group1.105 ruri=sip:group1.105@mydomain.tld duri=<null>
body=<?xml version="1.0"?> <dialog-info
xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full"
entity="sip:105@mydomain.tld"> <dialog
id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld"
call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld"
local-tag="4fbrxu9548" remote-tag="as4fb63283"
direction="recipient"> <state>early</state> <remote>
<identity>sip:103@mydomain.tld</identity> <target
uri="sip:103@mydomain.tld"/> </remote> <local>
<identity>sip:105@mydomain.tld</identity> <target
uri="sip:105@mydomain.tld"/> </local> </dialog> </dialog-info>
Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]:
INFO:presence:send_notify_request: NOTIFY sip:group1.101@mydomain.tld
via sip:group1.101@phone.ip:57234 on behalf of sip:105@mydomain.tld for
event dialog
But the phone's syslog shows this :
Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060
Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060
Aug 14 10:16:21 linksys NOTIFY sip:group1.101@phone.ip:57234 SIP/2.0^M
Via: SIP/2.0/UDP 72.55.182.125;branch=z9hG4bK0b0a.9725e23.0^M To:
sip:group1.101@mydomain.tld;tag=1c32db3af4d889ac^M From:
sip:105@mydomain.tld;tag=6d6077e15fecee48b2721a5e94bf8883-91a8^M CSeq:
12 NOTIFY^M Call-ID: 984d4cd2-37b04f4(a)192.168.1.104^M Content-Length:
655^M User-Agent: Kamailio (1.5.2-tls (i386/linux))^M Max-Forwards: 70^M
Event: dialog^M Contact: <sip:mydomain.tld:5060>^M Subscription-State:
active;expires=1470^M Content-Type: application/dialog-info+xml^M ^M
<?xml version="1.0"?> <dialog-info
xmlns="urn:ietf:params:xml:ns:dialog-info" version="11"
state="full" entity="group1.105(a)mydomain.tld"> <dialog
id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld"
call-id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld"
local-tag="loto7h3i9a" remote-tag="as4f284f05"
direction="recipient"> <state>early</state> <remote>
<identity>sip:103@mydomain.tld</identity> <target uri="
Everything is good, but notice the entity is somehow back to group1.105.
Where do you want me to run the traceroute? On the server?
David
Klaus Darilion wrote:
>
>
> David schrieb:
>> Hey,
>>
>> I tried that, but it did not seem to change anything. I saw the
>> message change on the second loop, but it seems that the presence
>> module figure out my tactic and fixed the message.
>
> That's not possible. The presence module does not alter the received
> body. I think the problem is somewhere else.
>
> Can you send me a trace?
>
> ngrep -t -P "" -W byline -d any port 5060
>
> regards
> klaus
>
>
>>
>> David
>>
>> Klaus Darilion wrote:
>>> That does not work: presence_server always uses the original
>>> received body, the modifications applied by you are not seen by the
>>> presence server.
>>>
>>> The workaround would be to "loop" the modified PUBLISH to Kamailio
>>> again, e.g.:
>>>
>>> listen=.......:5060
>>> listen=.......:5050
>>>
>>> route{
>>> ...
>>> if (is_method("PUBLISH")) {
>>> if (dst_port==5060) {
>>> subst_body('/group1[.]//');
>>> t_relay("udp:127.0.0.1:5050);
>>> } else if (dst_port==5050) {
>>> handle_publish();
>>> t_release();
>>> }
>>> }
>>> ...
>>> }
>>>
>>>
>>> klaus
>>> David schrieb:
>>>> Hey,
>>>>
>>>> I tried doing a subst or replace_body_all to change the entity=""
>>>> value,
>>>>
>>>> if (is_method("PUBLISH") )
>>>> {
>>>> subst_body('/group1[.]//');
>>>> handle_publish();
>>>> t_release();
>>>> }
>>>>
>>>> It did not work, the $rb variable still shows the old body and the
>>>> NOTIFY received on the phone is the old body as well.
>>>>
>>>> I thought about doing a string replace on the outbound NOTIFY but
>>>> it never hits my routing script, so I can't.
>>>>
>>>> I read textops and it does not list any limitations relating to me.
>>>>
>>>> Any ideas?
>>>>
>>>> David
>>>>
>>>> Klaus Darilion wrote:
>>>>> I think the problem lies within the dialog module. Dialog module
>>>>> stores from/to URI. Both, from and to URI are derived from the
>>>>> headers. We could extend the dialog module to store request-URI in
>>>>> the dialog-info structure too, and add an option to pua_dialoginfo
>>>>> to use RURI instead of To-URI.
>>>>>
>>>>> regards
>>>>> klaus
>>>>>
>>>>> David schrieb:
>>>>>> Hey,
>>>>>>
>>>>>> I think I am doing something wrong, because here is the NOTIFY
>>>>>> that comes to the phone :
>>>>>>
>>>>>> NOTIFY sip:group1.101@my.home.ip:57234 SIP/2.0
>>>>>> Via: SIP/2.0/UDP my.kamailio.ip;branch=z9hG4bK3cab.78e494a4.0
>>>>>> To: sip:group1.101@my.kamailio.domain.name;tag=b2514519391e8b4d
>>>>>> From:
>>>>>> sip:102@my.kamailio.domain.name;tag=6d6077e15fecee48b2721a5e94bf8883-f712
>>>>>>
>>>>>> CSeq: 12 NOTIFY^M Call-ID: 24db3209-611c031d(a)192.168.1.104
>>>>>> Content-Length: 298
>>>>>> User-Agent: MyServer 1.0
>>>>>> Max-Forwards: 70
>>>>>> Event: dialog
>>>>>> Contact: <sip:my.kamailio.domain.name:5060>
>>>>>> Subscription-State: active;expires=1570
>>>>>> Content-Type: application/dialog-info+xml
>>>>>>
>>>>>> <?xml version="1.0"?> <dialog-info
>>>>>> xmlns="urn:ietf:params:xml:ns:dialog-info" version="11"
>>>>>> state="full" entity="group1.102(a)my.kamailio.domain.name">
>>>>>> <dialog id="7aaa32e84362a246716009175ad670be(a)domain.tld"
>>>>>> direction="recipient"> <state>early</state> </dialog>
>>>>>> </dialog-info>
>>>>>>
>>>>>> The phone ( I tested with a GXP2000, GXP2020 and SPA962 + SPA932)
>>>>>> does not flash lights or anything. Since you suggested that a
>>>>>> solid kamailio would work out of the box, I suspect that either I
>>>>>> miscommunicated my setup or did something really wrong. The
>>>>>> notify definitely gets to the phone and the phone replies 200/OK
>>>>>> when it receives the NOTIFY. But I think that the telephone is
>>>>>> not understanding the request because it subscribed to '102' but
>>>>>> received a notification for 'group1.102'. The funny thing is, the
>>>>>> From header matches the subscribe To header, it's just the XML
>>>>>> has the full name instead of the shortened name.
>>>>>>
>>>>>> To:
>>>>>> <sip:102@my.kamailio.domain.name>;tag=6d6077e15fecee48b2721a5e94bf8883-f712
>>>>>>
>>>>>>
>>>>>> I see this in my ( on kamailio ) /var/log/messages :
>>>>>>
>>>>>> Aug 13 09:55:24 kamailio-dev /usr/sbin/kamailio[25449]:
>>>>>> INFO:presence:send_notify_request: NOTIFY
>>>>>> sip:group1.101@my.kamailio.domain.name via
>>>>>> sip:group1.101@my.home.ip.where.phone.is:57234 on behalf of
>>>>>> sip:102@my.kamailio.domain.name for event dialog
>>>>>>
>>>>>> I should also mention that the NOTIFY sent out by presence
>>>>>> bypasses my routing scripts. So I have the PUBLISH come through (
>>>>>> which I leave alone ) and the NOTIFY is sent according to the
>>>>>> location table without ever consulting my routing script.
>>>>>>
>>>>>> So everything amazingly worked out well, except that the lights
>>>>>> are not changing status which I think is related to the XML
>>>>>> document dialog-info entity attribute containing the group name,
>>>>>> sent from my server.
>>>>>>
>>>>>> Any ideas or suggestions ?
>>>>>>
>>>>>> Thanks,
>>>>>>
>>>>>> David
>>>>>>
>>>>>> Other information :
>>>>>>
>>>>>> if ( is_method("SUBSCRIBE") )
>>>>>> {
>>>>>> avp_db_query("select groupname from sometable ",
>>>>>> "$avp(s:groupname)");
>>>>>> avp_printf("$ru",
>>>>>> "sip:$avp(s:zone).$tU@my.kamailio.domain.name");
>>>>>> xlog("L_INFO", "Subscribe rewritten from $tU to $ru
>>>>>> - M=$rm \n");
>>>>>> xlog("L_INFO", "Handling SUBSCRIBE - $fU -
>>>>>> $avp(s:zone) - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
>>>>>> handle_subscribe();
>>>>>> t_release();
>>>>>> exit;
>>>>>> }
>>>>>>
>>>>>> Kamailio compiled from sources :
>>>>>>
>>>>>> Path: /usr/src/kamailio
>>>>>> URL:
>>>>>> https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5
>>>>>> Repository Root: https://openser.svn.sourceforge.net/svnroot/openser
>>>>>> Repository UUID: 689a6050-402a-0410-94f2-e92a70836424
>>>>>> Revision: 5910
>>>>>> Node Kind: directory
>>>>>> Schedule: normal
>>>>>> Last Changed Author: henningw
>>>>>> Last Changed Rev: 5910
>>>>>> Last Changed Date: 2009-08-06 13:08:30 -0400 (Thu, 06 Aug 2009)
>>>>>>
>>>>>>
>>>>>> Klaus Darilion wrote:
>>>>>>> notifies are generated by the presence module, based on the
>>>>>>> subscription. So, there is nothing you have to do with NOTIFY.
>>>>>>> Also PUBLISHs are created internally. Depending on your concrete
>>>>>>> setup (e.g. format of the SIP user names) it should work out of
>>>>>>> the box.
>>>>>>>
>>>>>>> klaus
>>>>>>>
>>>>>>> dlublink schrieb:
>>>>>>>> What about the notifies?
>>>>>>>>
>>>>>>>> Klaus Darilion wrote:
>>>>>>>>> dlublink wrote:
>>>>>>>>>> Hello,
>>>>>>>>>>
>>>>>>>>>> I have three different groups of extensions on my kamailio I
>>>>>>>>>> want to be able to separate them, so I prefixed a name to the
>>>>>>>>>> extensions, so I have :
>>>>>>>>>>
>>>>>>>>>> 1. group1.101
>>>>>>>>>> 2. group1.102
>>>>>>>>>> 3. group2.101
>>>>>>>>>> 4. group2.102
>>>>>>>>>> 5. group3.102
>>>>>>>>>> 6. group3.103.
>>>>>>>>>>
>>>>>>>>>> The phones from different groups can not call each other, I
>>>>>>>>>> found a pseudo variable that I use to rewrite the destination
>>>>>>>>>> url, so if user group1.101 dials 102 I rewrite it to group1.102.
>>>>>>>>>>
>>>>>>>>>> I want to do the same thing for presence_dialog info, how can
>>>>>>>>>> I rewrite the subscribe, presence and and notify messages to
>>>>>>>>>> append the appropriate prefix ?
>>>>>>>>>
>>>>>>>>> Just apply the same rewrite you have already done for the
>>>>>>>>> INVITE also for the SUBSCRIBE
>>>>>>>>>
>>>>>>>>> regards
>>>>>>>>> klaus
>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Thanks,
>>>>>>>>>>
>>>>>>>>>> David
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Kamailio (OpenSER) - Users mailing list
>>>>>>>>>> Users(a)lists.kamailio.org
>>>>>>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>>>>>>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>>>>>>>
>>>>>>
>>>>
>>
Hello,
I have three different groups of extensions on my kamailio I want to be
able to separate them, so I prefixed a name to the extensions, so I have :
1. group1.101
2. group1.102
3. group2.101
4. group2.102
5. group3.102
6. group3.103.
The phones from different groups can not call each other, I found a
pseudo variable that I use to rewrite the destination url, so if user
group1.101 dials 102 I rewrite it to group1.102.
I want to do the same thing for presence_dialog info, how can I rewrite
the subscribe, presence and and notify messages to append the
appropriate prefix ?
Thanks,
David
Hi,
I searched around the web to load balance asterisk servers and found
Kamailio for possible solution. Let's say I have two identical asterisk
servers with same dialplan and configuration and I want both servers look
like they have same IP address from clients. Based on my understanding, we
need 1 Kamailio as load balancer and 2 Asterisk servers as the real servers.
Let's say the setup is :
Kamailio load balancer 192.168.2.1
Asterisk Server #1 192.168.2.2
Asterisk Server #2 192.168.2.3
My question is, X-Lite softphone Configuration should be set to domain
192.168.2.1, right?
I also want to know the step by step configuration to set kamailio as load
balancer. I have not used Kamailio before. However after searching the
documentation, the step (based on my understanding) is somewhat like this :
1. Install Kamailio. I will use the step by step here :
http://kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn
2. Then, using dipatcher module, I will configure it using this guide here :
http://kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha
Then modify the dispatcher.list file to match the IP address of my asterisk
servers :
*1 sip:192.168.2.2:5060
1 sip:192.168.2.3:5060*
Am I missing some steps?
Do I also need to configure dialplan or any other file at Kamailio load
balancer? Or those two steps basically done it all for simple load balancing
configuration? Thanks for your responses.
Best Regard,
Kurt Weasel.
Hello everyone,
I want to port openSER on an embedded board,I need configure the openser
without MySQL due to space constraint.
So can you let me know whether this is possible also can you give any
reference document on how to port on ARM 9 based boad.
Regards,
Rathndran
<users(a)lists.kamailio.org>
.. fwd to list ... reply all button mismatch ...
---------- Forwarded message ----------
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: Wed, Aug 12, 2009 at 9:20 AM
Subject: Re: [Kamailio-Users] dialog info + Grandstreams = freezing
To: David <kamailio.org(a)spam.lublink.net>
Hello,
On Tue, Aug 11, 2009 at 2:18 PM, David <kamailio.org(a)spam.lublink.net>wrote:
> Hey,
>
> I am not good at diagrams, but here goes :
>
>
> [ Asterisk] [ Asterisk ] ================> [ MySQL ]
> / \ / \
> [ Kamailio ] ============/\
> / \ / \ / \
> [ Extension ] [ Extension ] [ Extension ]
>
> Here is a simple flow :
>
> 1. Extension 1 is plugged in
> 2. It registers to Kamailio and Kamailio stores to Location
> 3. Extension 1 subscribes to all the other extensions
> 4. Kamailio calls handle_publish() and t_release()
> 5. Extension 1 dials '102'
> 6. Kamailio receives the INVITE chooses an asterisk server and relays the
> INVITE ( t_relay )
> 7. Asterisk does some checking, starts monitor ( if needed ) and some other
> stuff
> 8. Asterisk sends INVITE to Kamailio for testspace.102
> 9. Kamailio finds testspace.102 in database and forwards INVITE to the
> appropriate extension
> 10. Extension replies 'OK'
> 11. Kamailio, using reply-route, sends it back to asterisk
> 12. Asterisk bridges the call with 101
> 13. Asterisk sends an OK back through Kamailio to Extension 1.
> 14. Asterisk tries to reinvite the audio ( if not monitoring )
> 15. Audio bypasses Asterisk, but the SIP still is making the full trip.
>
> Does this answer your question?
Do you do call tracking with dialog for both legs (inbond: phone to
asterisk; outbound: asterisk to phone)? You should do it for one, outbound
is the best, in case asterisk auto-answers the inbound leg.
Cheers,
Daniel
>
> David
>
>
>
> Daniel-Constantin Mierla wrote:
>
>> Hello,
>>
>> I am using the module with snom 370 and works ok. The BLF is on for
>> Polycoms but they do not implement the call pickup properly (they have
>> direct call pickup as far as I could get).
>>
>> Please take latest kamailio 1.5 from svn branch 1.5 as there were
>> committed some updates (maybe they don't affect you, being mainly to rls,
>> but is better to have the latest stable so we refer to same source code).
>>
>> What would be good is to have a diagram of the signaling, since you say
>> there are lot of messages sent around.
>> http://callflow.sourceforge.net/
>>
>> Cheers,
>> Daniel
>>
>>
>> On 11.08.2009 5:03 Uhr, David wrote:
>>
>>> Hello,
>>>
>>> I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04,
>>> 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS
>>> 1.5.2-tls compilde on Ubuntu 8.04
>>>
>>> I am trying to setup presence_dialoginfo with my Grandstreams, Snom and
>>> Linksys. I have a 4 phones on the server.
>>>
>>> 101 - Linksys SPA962 ( 6.1.5a )
>>> 102 - Grandstream GXP2020 ( 1.2.1.4 )
>>> 103 - Grandstream GXP2000 ( 1.2.1.4 )
>>> 104 - Grandstream GXP2000 ( 1.1.6.46 )
>>> 105 - Snom 360 ( 7.3.23 )
>>>
>>> My Kamailio deals with registrations, NAT and BLF everything else is sent
>>> to one of two asterisk boxes. I use the dispatcher module for this. This
>>> means that when I call one extension to the other, both call legs from
>>> asterisk are going through Kamailio as separate calls. But to divide my
>>> customers, the usernames are different from the URL that the user types. For
>>> example the customer dials '101' but it is changed to testspace.101 when it
>>> comes back from asterisk. So Kamailio would have two calls in the event that
>>> 101 dials 102.
>>>
>>> sip:testspace.101@myserver to 102 ( this is sent to asterisk )
>>> sip:testspace.101@myserver to testspace.102 ( this is coming back from
>>> asterisk )
>>>
>>> Something is horribly wrong. I have the following problems :
>>>
>>> 1. If 102 calls 103, when 103 answers both phones hang for about 2
>>> minutes
>>> 2. If 105 calls 101, 101 BLF comes back to the inactive state ( green on
>>> the Linksys and dark on the Snom), but the orange light stays on on the Snom
>>> and it thinks the call is still active ( the light is on, but the call is
>>> over )
>>> 3. If any extension calls any extension and I try a call pickup, it
>>> fails. It looks like the Linksys is sending a NOTIFY to pickup the call ( I
>>> thought it was supposed to send an invite... ? )
>>>
>>> Looking at the logs it looks like Kamailio is sending out so many NOTIFYs
>>> that it is crashing the Grandstreams, and causing the Snom to act funny.
>>>
>>> Here are some experts from my config file :
>>>
>>> root@kamailio-dev:/etc/kamailio# grep dialog *
>>> kamailio.cfg:# * avp value for dialogs is still not correct
>>> kamailio.cfg:loadmodule "dialog.so"
>>> kamailio.cfg:loadmodule "presence_dialoginfo.so"
>>> kamailio.cfg:loadmodule "pua_dialoginfo.so"
>>> kamailio.cfg:#modparam("pua_dialoginfo", "include_localremote", 0)
>>> kamailio.cfg:#modparam("pua_dialoginfo", "include_tags", 0)
>>> kamailio.cfg:#modparam("pua_dialoginfo", "include_callid", 0)
>>> kamailio.cfg:modparam("dialog", "dlg_flag", 4)
>>> kamailio.cfg:modparam("dialog", "db_mode", 1)
>>> kamailio.cfg:modparam("dialog", "timeout_avp", "$avp(i:10)") # I still
>>> haven't figured out how to set $avp(i:10)
>>> kamailio.cfg:modparam("pua_dialoginfo", "override_lifetime", 300)
>>> kamailio.cfg:modparam("presence_dialoginfo", "force_single_dialog", 1)
>>> kamailio.cfg:modparam("pua_dialoginfo", "caller_confirmed", 1)
>>> kamailio.cfg:modparam("auth_db|usrloc|acc|domain|avpops|presence|presence_xml|pua|dialog",
>>> "db_url",
>>> kamailio.cfg:# Flag 4 = Mark the current request for a dialog
>>> kamailio.cfg: # sequential request withing a dialog should
>>>
>>> the set flag looks like this :
>>> if ( ds_is_from_list() )
>>> {
>>> xlog("L_INFO", "Coming from asterisk");
>>> if ( is_method("INVITE"))
>>> {
>>> setflag(4);
>>> }
>>> }
>>>
>>> So the dialog flag is only set for the leg coming back from asterisk.
>>>
>>> When a notify comes in :
>>>
>>> if(is_method("NOTIFY") )
>>> {
>>> if (! t_newtran())
>>> {
>>> sl_reply_error();
>>> exit;
>>> };
>>>
>>> t_reply("200", "OK");
>>> t_release();
>>> exit ;
>>> }
>>>
>>> Publish and subscribe are like this :
>>>
>>>
>>> if( is_method("PUBLISH") || is_method("SUBSCRIBE") )
>>> {
>>> route(5);
>>> exit;
>>> }
>>>
>>> route[5]
>>> {
>>> # absorb retransmissions
>>> if (! t_newtran())
>>> {
>>> xlog("L_INFO", "Ignoring PUBLISH/SUBSCRIBE on retransmition -
>>> M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
>>> sl_reply_error();
>>> exit;
>>> };
>>>
>>> append_to_reply("Contact: <sip:myserver.tld:5060>\r\n");
>>>
>>> if(is_method("PUBLISH"))
>>> {
>>> handle_publish();
>>> t_release();
>>>
>>> } else if( is_method("SUBSCRIBE")) {
>>> handle_subscribe();
>>> t_release();
>>> }
>>> else
>>> {
>>> }
>>>
>>> exit;
>>> }
>>>
>>> I also have NAT checking for those telephones where stun isn't enough.
>>> Before I reach publish/subscribe/invite/notify, I also call setbflag() and
>>> sometimes call fix_nated_contact(). Additionnally, I have a block if code
>>> before my presence stuff if ( has_totag() && loose_route()) { t_relay(); }.
>>>
>>> If sip.conf:canreinvite=yes, the grandstreams freeze so long that the
>>> server times out, and the BLFs get really messed up. if sip:canreinvite=no
>>> the grandstreams only freeze for about 30 seconds.
>>>
>>> Obviously I am doing something wrong, but despite having searched google
>>> for endless hours, and poured over documentation, I can not seem to find
>>> what I did wrong.
>>>
>>> I would really appreciate if someone could shed light on my problem.
>>>
>>> I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04,
>>> 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS
>>> 1.5.2-tls compilde on Ubuntu 8.04
>>>
>>> Thanks,
>>>
>>> David
>>>
>>> _______________________________________________
>>> Kamailio (OpenSER) - Users mailing list
>>> Users(a)lists.kamailio.org
>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users(a)lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
--
Daniel-Constantin Mierla
http://www.asipto.com
--
Daniel-Constantin Mierla
http://www.asipto.com
Hello to all
I enabled the acc module but it isn't writing to "acc" or
"missed_calls" table in mysql, and to a log file too.
my config:
(...)
loadmodule "db_mysql.so"
loadmodule "tm.so"
loadmodule "rr.so"
(...)
# ----- acc params -----
/* what sepcial events should be accounted ? */
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
/* by default ww do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
modparam("acc",
"log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
/* uncomment the following lines to enable DB accounting also */
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("acc", "db_url", "mysql://xxx:xxxx@localhost/kamailio")
modparam("acc", "db_extra",
"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
Flags 1 and 2 set just like in a sample configuration file.
(...)
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
}
route(1);
(...)
# account only INVITEs
if (is_method("INVITE")) {
setflag(1); # do accounting
}
Mysql tables are created:
| acc | CREATE TABLE `acc` (
`id` int(10) unsigned NOT NULL auto_increment,
`method` varchar(16) NOT NULL default '',
`from_tag` varchar(64) NOT NULL default '',
`to_tag` varchar(64) NOT NULL default '',
`callid` varchar(128) NOT NULL default '',
`sip_code` char(3) NOT NULL default '',
`sip_reason` varchar(32) NOT NULL default '',
`time` datetime NOT NULL default '0000-00-00 00:00:00',
`src_ip` varchar(64) NOT NULL default '',
`dst_user` varchar(64) NOT NULL default '',
`dst_domain` varchar(128) NOT NULL default '',
`src_user` varchar(64) NOT NULL default '',
`src_domain` varchar(128) NOT NULL default '',
`cdr_id` int(11) NOT NULL default '0',
`dst_ouser` varchar(64) NOT NULL default '',
PRIMARY KEY (`id`),
KEY `acc_callid` (`callid`)
) ENGINE=MyISAM DEFAULT CHARSET=latin1 |
| missed_calls | CREATE TABLE `missed_calls` (
`id` int(10) unsigned NOT NULL auto_increment,
`method` varchar(16) NOT NULL default '',
`from_tag` varchar(64) NOT NULL default '',
`to_tag` varchar(64) NOT NULL default '',
`callid` varchar(128) NOT NULL default '',
`sip_code` char(3) NOT NULL default '',
`sip_reason` varchar(32) NOT NULL default '',
`time` datetime NOT NULL default '0000-00-00 00:00:00',
`src_ip` varchar(64) NOT NULL default '',
`dst_user` varchar(64) NOT NULL default '',
`dst_domain` varchar(128) NOT NULL default '',
`src_user` varchar(64) NOT NULL default '',
`src_domain` varchar(128) NOT NULL default '',
`cdr_id` int(11) NOT NULL default '0',
`dst_ouser` varchar(64) NOT NULL default '',
PRIMARY KEY (`id`),
KEY `mc_callid` (`callid`)
) ENGINE=MyISAM DEFAULT CHARSET=latin1
There are no errors in log, and nothing about acc module in debug mode.
Cant get where is my mistake.
Hi!
Can any of you explain how select framework is working?
Where can i find any documentation about it.
I am experiencing problem when i am try using it.
Any help highly appreciated!
Regards,
Misi
here is correct reqwest and reqwest via kamailio
is there any way to rewrite To: field in kamailio
because now it nto work
correct reqwest(asterisk)
RECEIVING FROM: 69.70.173.195:5060
CANCEL sip:34249@193.110.78.12:8484 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK4c97bd4a
From: "Unknown"
<sip:Unknown@voip1.bravotelecom.com<sip%3AUnknown(a)voip1.bravotelecom.com>
>;tag=as366a0ba0
To: <sip:34249@193.110.78.12:8484>
Call-ID: 322db75c71be4ba45a61c7666b93e6be(a)voip1.bravotelecom.com
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
incorrect reqwest(multi-homed kamailio, asterisk originate to internal eth)
RECEIVING FROM: 69.70.173.195:5060
CANCEL sip:34249@193.110.78.12:8484 SIP/2.0
Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK1b1b.3c42cea6.0
From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195>
>;tag=as6075df60
Call-ID: 4d784fa972c4e9c9795032417f46cb26(a)voip1.bravotelecom.com
To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>>
CSeq: 102 CANCEL
Max-Forwards: 70
asterisk
Content-Length: 0
--
Merkulov Alexander