Re: [SR-Users] Evapi module is not broadcasting any data
Hi Daniel,
sorry for not being clear.
I have all three evapi event routes in my config. When I telnet to port
8448, xlog shows that evapi:connection-new is exectued. When I send
6:efelin, to telnet, the evapi:message-received is executed. When I close
the telnet session, the evapi:connection-closed is executed. This works
just fine. However, when I execute evapi_async_relay or evapi_relay nothing
is sent to my telnet session or CGRateS application (This was also observed
by ngrep and tcpdump).
My configuration is default from CGRateS:
https://github.com/cgrates/cgrates/tree/master/data/tutorials/kamevapi/kamailio/etc/kamailio
(kamailio.cfg and included kamailio-cgrates.cfg) with few xlogs for further
debuging +
modparam("evapi", "workers", 2)
modparam("evapi", "bind_addr", "0.0.0.0:8448")
I have just downloaded the fresh config in order to evade some typo, but
the problem persists.
Thanks
Jan
2016-08-22 9:35 GMT+02:00 Daniel-Constantin Mierla :
>
>
> On 22/08/16 09:31, Efelin Novak wrote:
>
> [...]
>
> When I connect to evapi using telnet, I receive NO answer:
>
> root@PC:~$ telnet 192.168.30.29 8448
> Trying 192.168.30.29...
> Connected to 192.168.30.29.
> Escape character is '^]'.
>
>
> However it reacts on my netstring messages and executes the
> event_route[evapi:message-received].
>
>
> To clarify -- do you have event_route[evapi:connection-new] {...} block
> defined in your config and it is not executed?
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierlahttp://www.asipto.com -
> http://www.kamailio.orghttp://twitter.com/#!/miconda -
> http://www.linkedin.com/in/miconda
>
>
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Re: [SR-Users] Evapi module is not broadcasting any data
Hi Daniel, kamctl ps: root@cgrates:/usr/share/cgrates/tutorials/kamevapi/cgrates/etc/init.d# kamctl ps Process:: ID=0 PID=14529 Type=main process - attendant Process:: ID=1 PID=14530 Type=udp receiver child=0 sock=192.168.30.29:5060 Process:: ID=2 PID=14531 Type=udp receiver child=1 sock=192.168.30.29:5060 Process:: ID=3 PID=14532 Type=udp receiver child=2 sock=192.168.30.29:5060 Process:: ID=4 PID=14533 Type=udp receiver child=3 sock=192.168.30.29:5060 Process:: ID=5 PID=14534 Type=slow timer Process:: ID=6 PID=14535 Type=timer Process:: ID=7 PID=14536 Type=secondary timer Process:: ID=8 PID=14537 Type=MI FIFO Process:: ID=9 PID=14538 Type=ctl handler Process:: ID=10 PID=14539 Type=TIMER NH Process:: ID=11 PID=14540 Type=EvAPI Dispatcher Process:: ID=12 PID=14541 Type=EvAPI Worker Process:: ID=13 PID=14542 Type=EvAPI Worker Process:: ID=14 PID=14543 Type=Dialog Clean Timer Process:: ID=15 PID=14544 Type=tcp receiver (generic) child=0 Process:: ID=16 PID=14545 Type=tcp receiver (generic) child=1 Process:: ID=17 PID=14546 Type=tcp receiver (generic) child=2 Process:: ID=18 PID=14547 Type=tcp receiver (generic) child=3 Process:: ID=19 PID=14548 Type=tcp main process When I connect to evapi using telnet, I receive NO answer: root@PC:~$ telnet 192.168.30.29 8448 Trying 192.168.30.29... Connected to 192.168.30.29. Escape character is '^]'. However it reacts on my netstring messages and executes the event_route[evapi:message-received]. Thanks for a reply. Jan 2016-08-22 9:25 GMT+02:00 Daniel-Constantin Mierla : > Hello, > > did you get the evapi message on your application or telnet connection? > > Can you give the output of: > > kamctl ps > > to see if the evapi processes are running? > > Cheers, > Daniel > > On 19/08/16 18:45, Efelin Novak wrote: > > Hi folks, > > I came into a problem with evapi module, while I was trying to make the > CGRateS tutorial work. It does not send any messages at all. > > Taken steps: > > I start kamailio with debug=3. > > I start: > CGRates (on the kamailio machine) > telnet 192.168.30.29 8448 (from my PC) > tcpdump port 8448 -A -s0 -i any (on the kamailio machine) > > Telnet command successfully connects to kamailio. Kamailio yells: > > root@cgrates:/opt/kamailio/src/kamailio-4.4/kamailio# 11(12205) DEBUG: > evapi [evapi_dispatch.c:453]: evapi_accept_client(): new connection - > pos[1] from: [192.168.30.1:40312] > 11(12205) INFO:
Re: [SR-Users] Evapi module is not broadcasting any data
Hi, thanks for a reply. Command "ngrep -d any -t -W byline port 8448" prints # when CGRateS and telnet connect (TCP handshakes) and stays quiet during the rest of the test. When I type something into telnet, I can see incoming packets: T 2016/08/22 09:18:21.115922 192.168.30.1:46378 -> 192.168.30.29:8448 [AP] efelin. Unfortunetely I am no closer to resolving my issue. Netstat shows: root@cgrates:/usr/share/cgrates/tutorials/kamevapi/cgrates/etc/init.d# netstat -tlpnueea | grep kamailio tcp0 0 0.0.0.0:84480.0.0.0:* LISTEN 0 78179 14540/kamailio tcp0 0 192.168.30.29:5060 0.0.0.0:* LISTEN 0 78149 14548/kamailio tcp0 0 127.0.0.1:8448 127.0.0.1:51976 ESTABLISHED 0 78552 14540/kamailio tcp0 0 192.168.30.29:8448 192.168.30.1:46432 ESTABLISHED 0 78493 14540/kamailio udp0 0 192.168.30.29:5060 0.0.0.0:* 0 78147 14529/kamailio Thanks. Jan 2016-08-20 4:22 GMT+02:00 Infinicalls Infinicalls : > Hi, > Use "ngrep -t -W byline port 8448" to sniff the port and see if it helps. > > regards > Ganesh Kumar > > > On 8/19/16, Efelin Novak wrote: > > Hi folks, > > > > I came into a problem with evapi module, while I was trying to make the > > CGRateS tutorial work. It does not send any messages at all. > > > > Taken steps: > > > > I start kamailio with debug=3. > > > > I start: > > CGRates (on the kamailio machine) > > telnet 192.168.30.29 8448 (from my PC) > > tcpdump port 8448 -A -s0 -i any (on the kamailio machine) > > > > Telnet command successfully connects to kamailio. Kamailio yells: > > > > root@cgrates:/opt/kamailio/src/kamailio-4.4/kamailio# 11(12205) DEBUG: > > evapi [evapi_dispatch.c:453]: evapi_accept_client(): new connection - > > pos[1] from: [192.168.30.1:40312] > > 11(12205) INFO:
[SR-Users] Evapi module is not broadcasting any data
Hi folks, I came into a problem with evapi module, while I was trying to make the CGRateS tutorial work. It does not send any messages at all. Taken steps: I start kamailio with debug=3. I start: CGRates (on the kamailio machine) telnet 192.168.30.29 8448 (from my PC) tcpdump port 8448 -A -s0 -i any (on the kamailio machine) Telnet command successfully connects to kamailio. Kamailio yells: root@cgrates:/opt/kamailio/src/kamailio-4.4/kamailio# 11(12205) DEBUG: evapi [evapi_dispatch.c:453]: evapi_accept_client(): new connection - pos[1] from: [192.168.30.1:40312] 11(12205) INFO:
Re: [SR-Users] restore_uri_reply() called twice in 200OK to BYE
Hi Daniel,
thank you for a reply. I have managed to solve it in a different way.
In a first message I forgot to add that I'm calling uac_replace_from before
mentioned code. I have moved uac_replace_from function, so it is called
after the dlg_manage function and right now only one callback is
registered. This fixed it.
Isn't it possible to un-register the first callback when the second one is
registered?
Kind regards
Efelin
2015-11-06 8:27 GMT+01:00 Daniel-Constantin Mierla :
> Hello,
>
> try without restore_mode auto if you want that restore_dlg to be 1.
>
> Cheers,
> Daniel
>
>
> On 05/11/15 18:00, Efelin Novak wrote:
>
> Hi Folks,
>
> I have a problem with To and From headers in 200 OK to BYE request. Their
> URI is doubled when leaving Kamailio.
>
> Incoming 200 OK:
> From : ;tag=as017a2986
> To : ;tag=K7ryDgH2g7gFm
> Outgoing 200 OK:
> From : ;tag=as017a2986
> To : ;tag=K7ryDgH2g7gFm
>
> I'm using uac module to replace these headers in auto and dialog mode:
>
> modparam("uac", "restore_mode", "auto")
> modparam("uac", "restore_dlg", 1)
>
> append_fromtag is also set:
>
> modparam("rr", "append_fromtag", 1)
>
> When I run kamailio in debug mode I can see the function
> restore_uri_reply() is called twice:
>
> .
> .
> .
> 0(3133) DEBUG: [parser/parse_via.c:1284]: parse_via_param(): Found
> param type 234, = ; state=6
> 0(3133) DEBUG: [parser/parse_via.c:1284]: parse_via_param(): Found
> param type 232, = ; state=6
> 0(3133) DEBUG: [parser/parse_via.c:1284]: parse_via_param(): Found
> param type 235, = <5060>; state=16
> 0(3133) DEBUG: [parser/parse_via.c:2672]: parse_via(): end of
> header reached, state=5
> 0(3133) DEBUG: [parser/msg_parser.c:513]: parse_headers():
> parse_headers: Via found, flags=62
> 0(3133) DEBUG: [parser/msg_parser.c:526]: parse_headers():
> parse_headers: this is the second via
> 0(3133) DEBUG: [parser/parse_addr_spec.c:176]: parse_to_param():
> DEBUG: add_param: tag=teHD72X67aQUm
> 0(3133) DEBUG: [parser/parse_addr_spec.c:898]: parse_addr_spec():
> end of header reached, state=29
> 0(3133) DEBUG: [parser/msg_parser.c:190]: get_hdr_field(): DEBUG:
> get_hdr_field: [60]; uri=[sip:421222@IP_ADDRESS:5060]
> 0(3133) DEBUG: [parser/msg_parser.c:192]: get_hdr_field(): DEBUG:
> to body []
> 0(3133) DEBUG: [parser/msg_parser.c:170]: get_hdr_field():
> get_hdr_field: cseq : <103>
> 0(3133) DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG:
> t_reply_matching: hash 10806 label 0 branch 0
> 0(3133) DEBUG: tm [t_lookup.c:1004]: t_reply_matching(): DEBUG:
> t_reply_matching: reply matched (T=0x220071b4)!
> 0(3133) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG:
> trans=0x220071b4, callback type 2, id 0 entered
> 0(3133) DEBUG: [parser/parse_addr_spec.c:176]: parse_to_param():
> DEBUG: add_param: tag=as46bccb83
> 0(3133) DEBUG: [parser/parse_addr_spec.c:898]: parse_addr_spec():
> end of header reached, state=29
> 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <
> >
> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <
> >
> 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <
> >
> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <
> >
> 0(3133) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG:
> trans=0x220071b4, callback type 2, id 0 entered
> 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <
> >
> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <
> >
> 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <
> >
> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <
> >
> .
> .
> .
>
> I think the problem is in registering a function restore_uris_reply() in
> callbacks in functions (both in replace.c of uac module)
> * void rr_checker(struct sip_msg *msg, str *r_param, void *cb_param)
> * static void replace_callback(struct dlg_cell *dlg, int type, struct
> dlg_cb_params *_params)
>
> Both of these callbacks are registered for BYE and ACK requests (INVITE
> request is using different callback), therefore running restore_uri_reply()
> twice.
> For testing purposes after I have removed the callback in rr_checker and
> recompile the uac module, To and From headers were OK.
>
> In my configuration I'm calling uac_replace_to in RELAY route:
> route[RELAY]
> {
>
> if ( !has_totag() ) {
> dlg_manage();
> uac_replace_to("$ru");
> }
>
>
[SR-Users] restore_uri_reply() called twice in 200OK to BYE
Hi Folks,
I have a problem with To and From headers in 200 OK to BYE request. Their
URI is doubled when leaving Kamailio.
Incoming 200 OK:
From : ;tag=as017a2986
To : ;tag=K7ryDgH2g7gFm
Outgoing 200 OK:
From : ;tag=as017a2986
To : ;tag=K7ryDgH2g7gFm
I'm using uac module to replace these headers in auto and dialog mode:
modparam("uac", "restore_mode", "auto")
modparam("uac", "restore_dlg", 1)
append_fromtag is also set:
modparam("rr", "append_fromtag", 1)
When I run kamailio in debug mode I can see the function
restore_uri_reply() is called twice:
.
.
.
0(3133) DEBUG: [parser/parse_via.c:1284]: parse_via_param(): Found
param type 234, = ; state=6
0(3133) DEBUG: [parser/parse_via.c:1284]: parse_via_param(): Found
param type 232, = ; state=6
0(3133) DEBUG: [parser/parse_via.c:1284]: parse_via_param(): Found
param type 235, = <5060>; state=16
0(3133) DEBUG: [parser/parse_via.c:2672]: parse_via(): end of
header reached, state=5
0(3133) DEBUG: [parser/msg_parser.c:513]: parse_headers():
parse_headers: Via found, flags=62
0(3133) DEBUG: [parser/msg_parser.c:526]: parse_headers():
parse_headers: this is the second via
0(3133) DEBUG: [parser/parse_addr_spec.c:176]: parse_to_param():
DEBUG: add_param: tag=teHD72X67aQUm
0(3133) DEBUG: [parser/parse_addr_spec.c:898]: parse_addr_spec():
end of header reached, state=29
0(3133) DEBUG: [parser/msg_parser.c:190]: get_hdr_field(): DEBUG:
get_hdr_field: [60]; uri=[sip:421222@IP_ADDRESS:5060]
0(3133) DEBUG: [parser/msg_parser.c:192]: get_hdr_field(): DEBUG:
to body []
0(3133) DEBUG: [parser/msg_parser.c:170]: get_hdr_field():
get_hdr_field: cseq : <103>
0(3133) DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG:
t_reply_matching: hash 10806 label 0 branch 0
0(3133) DEBUG: tm [t_lookup.c:1004]: t_reply_matching(): DEBUG:
t_reply_matching: reply matched (T=0x220071b4)!
0(3133) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG:
trans=0x220071b4, callback type 2, id 0 entered
0(3133) DEBUG: [parser/parse_addr_spec.c:176]: parse_to_param():
DEBUG: add_param: tag=as46bccb83
0(3133) DEBUG: [parser/parse_addr_spec.c:898]: parse_addr_spec():
end of header reached, state=29
0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing
<>
0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting
<>
0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing
<>
0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting
<>
0(3133) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG:
trans=0x220071b4, callback type 2, id 0 entered
0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing
<>
0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting
<>
0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing
<>
0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting
<>
.
.
.
I think the problem is in registering a function restore_uris_reply() in
callbacks in functions (both in replace.c of uac module)
* void rr_checker(struct sip_msg *msg, str *r_param, void *cb_param)
* static void replace_callback(struct dlg_cell *dlg, int type, struct
dlg_cb_params *_params)
Both of these callbacks are registered for BYE and ACK requests (INVITE
request is using different callback), therefore running restore_uri_reply()
twice.
For testing purposes after I have removed the callback in rr_checker and
recompile the uac module, To and From headers were OK.
In my configuration I'm calling uac_replace_to in RELAY route:
route[RELAY]
{
if ( !has_totag() ) {
dlg_manage();
uac_replace_to("$ru");
}
if (!t_relay())
{
sl_reply_error();
}
}
So I think there is a bug in uac module, when one function is registered
twice in callbacks for NON-INVITE requests.
Or is there a problem with my configuration script?
Can anyone point me to the direction how to solve this issue?
Thank you
Efelin
Kamailio version: kamailio 4.4.0-dev6 (i386/linux) a66e22-dirty
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Re: [SR-Users] Missing Route headers in locally generated BYE
Hi Daniel, thanks for an answer & sorry for the delay (Christmas time). Here we go: Scenario: phone1(192.168.9.3, 192.168.10.75 (nat)) <--TCP--> kamailio1(192.168.10.2) <--UDP--> kamailio2(192.168.5.3) <--UDP--> phone2(192.168.12.2) dialog:: hash=1456:1090244220 state:: 4 timestart:: 1388767148 timeout:: 70022 callid:: 7fbca98b-22ea049@192.168.9.3 from_uri:: sip:pho...@example.com from_tag:: 17cfe8a323b2ac11o1 caller_contact:: sip:phone1@192.168.10.75:5075;transport=tcp caller_cseq:: 102 caller_route_set:: caller_bind_addr:: tcp:192.168.10.2:5060 to_uri:: sip:pho...@example.com to_tag:: a94c095b773be1dd6e8d668a785a9c84ec059e7d callee_contact:: sip:phone2@192.168.12.2:5060 callee_cseq:: 102 callee_route_set:: , callee_bind_addr:: udp:192.168.10.2:5060 Efelin 2013/12/11 Alex Balashov > On 12/11/2013 03:41 AM, Daniel-Constantin Mierla wrote: > > there must not be any route for proxy itself, even when BYE is sent by >> end UA the Route is consumed by proxy, so it is no difference from this >> perspective. >> > > Oh, yeah. [slaps head] That's true. You and Carsten are correct. > > > -- Alex > > -- > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Decatur, GA 30030 > United States > Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Missing Route headers in locally generated BYE
Hi, I locally generate BYE using dlg_end_dlg. When I want to end a call that is "transport layer" bridged, the BYE is not sent to first hop in route_set but directly to the endpoint. In such BYE there are no Route headers. In non-bridging calls Routes are correctly placed and the message is routed to the first "hop". When the error happens, this is written to a log: WARNING: rr [loose.c:821]: after_loose(): no socket found for match second RR Here ([SR-Users] no socket found for match second RR) I have read this is only a warning, but in my configuration it seriously influences the message routing. My setup is phone1(192.168.10.3) <--TCP--> kamailio1(192.168.10.2) <--UDP--> kamailio2(192.168.5.3) <--UDP--> phone2 On kamailio1 I generate dlg_end_dlg and the BYE is sent to phone1 and phone2 directly. I'm using Kamailio 4.0.4 on Debian machines. How can I make the Kamailio1 to send the BYE to kamailio2 in the transport layer bridging scenario? Do I have some misconfiguration or this is not a correct behaviour? Thanks for answer Efelin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] t_reply in failure route with dialog module
params->param = &(cbp->param);
(gdb)
290cbp->callback( trans, type, params );
(gdb)
292cbp=cbp->next;
(gdb)
283while(cbp){
(gdb)
286if ( (cbp->types)&type ) {
(gdb)
292cbp=cbp->next;
(gdb)
283while(cbp){
(gdb)
286if ( (cbp->types)&type ) {
(gdb)
287DBG("DBG: trans=%p, callback type %d, id %d entered\n",
(gdb)
290cbp->callback( trans, type, params );
(gdb)
289params->param = &(cbp->param);
(gdb)
290cbp->callback( trans, type, params );
(gdb)
292cbp=cbp->next;
(gdb)
283while(cbp){
Regards
Efelin
2013/11/20 Daniel-Constantin Mierla
> Hello,
>
> I will investigate more -- meanwhile had some traveling. It would speed up
> if you can send the backtrace of one process that blocks when you applied
> the patch.
>
> You need to connect with gdb to it:
>
> gdb /path/to/kamailio _PID_
>
> replace _PID_ with the pid of blocked kamailio process.
>
> Cheers,
> Daniel
>
>
> On 11/15/13 5:50 PM, Efelin Novak wrote:
>
>Hi Daniel,
>
> thanks for a reply. I applied the patch and now the Kamailio just prints
>
> WARNING: tm [t_lookup.c:1564]: t_unref(): WARNING: script writer didn't
> release transaction
>
> and than freezes without any log. It does not resend the incoming
> "winning" failure reply neither response to any other messages, not even to
> a new calls. It just freezes. All kamailio processes are running and eating
> the whole 4-core processor.
>
> Restart of Kamailio solves this problem.
>
> Any ideas how to continue with debug?
>
> Thanks
>
> Efelin
>
>
> 2013/11/15 Daniel-Constantin Mierla
>
>> Hello,
>>
>> can you try attached patch?
>>
>> Let me know if all goes fine and I will commit it to the repository.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 11/15/13 10:25 AM, Efelin Novak wrote:
>>
>> Hi,
>>
>> when I use t_reply("505", "Error"); in my failure route, the response is
>> not forwarded and following is written into a log:
>>
>> kamailio[26216]: WARNING: tm [t_lookup.c:1559]: t_unref(): WARNING:
>> script writer didn't release transaction
>>
>> plus next line is written exactly 416000 times into a log afterwards:
>>
>> kamailio[32685]: CRITICAL: dialog [dlg_hash.c:794]: log_next_state_dlg():
>> bogus event 4 in state 5 for dlg 0xb4af6588 [2575:7017] with clid
>> '121d44f0-6555f4c8' and tags 'd12546d053aadc68o2' ''
>>
>> My point is to change the incoming code from users and append a Q.850
>> reason code.
>> Is there any other way how to do this or a way how to fix this?
>> I'm using Kamilio 4.0.4 on Debian 7.1
>>
>>
>> The code is as follows:
>> failure_route[MANAGE_FAILURE]
>> {
>> if (t_is_canceled()) {
>> exit;
>> }
>> if($T_reply_code == 408 && isflagset(10))
>> {
>> xlog("Ringing timeout");
>> append_to_reply("Reason: Q.850;cause=28\r\n");
>> t_reply("505", "Error");
>> }
>> }
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierla -
>> http://www.asipto.comhttp://twitter.com/#!/miconda -
>> http://www.linkedin.com/in/miconda
>> Kamailio Advanced Trainings - Berlin, Nov 25-28
>> - more details about Kamailio trainings at http://www.asipto.com -
>>
>>
>> ___
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>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda
> - http://www.linkedin.com/in/miconda
> Kamailio Advanced Trainings - Berlin, Nov 25-28
> - more details about Kamailio trainings at http://www.asipto.com -
>
>
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[SR-Users] Append to reply in default action of failure_route
Hi,
I would like to append a header to a 'winning' negative reply in
failure_route and let the Kamailio do the default action (state fully
forward the winning reply).
When I use append_to_reply("Foo: bar\r\n"); and then call exit; in
failure_route nothing is appended.
When I use same append_to_reply then t_relay("505","Error"); and exit; the
header is appended.
When I use append and t_reply with dialog modul turned on I got a bug I'm
solving here '[SR-Users] t_reply in failure route with dialog module'.
So my question is how to put a header into a reply when I don't want to
alter its code or text?
I'm using Kamilio 4.0.4 on Debian 7.1
Thanks for an answer
Efelin
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Re: [SR-Users] t_reply in failure route with dialog module
Hi Daniel,
thanks for a reply. I applied the patch and now the Kamailio just prints
WARNING: tm [t_lookup.c:1564]: t_unref(): WARNING: script writer didn't
release transaction
and than freezes without any log. It does not resend the incoming "winning"
failure reply neither response to any other messages, not even to a new
calls. It just freezes. All kamailio processes are running and eating the
whole 4-core processor.
Restart of Kamailio solves this problem.
Any ideas how to continue with debug?
Thanks
Efelin
2013/11/15 Daniel-Constantin Mierla
> Hello,
>
> can you try attached patch?
>
> Let me know if all goes fine and I will commit it to the repository.
>
> Cheers,
> Daniel
>
>
> On 11/15/13 10:25 AM, Efelin Novak wrote:
>
> Hi,
>
> when I use t_reply("505", "Error"); in my failure route, the response is
> not forwarded and following is written into a log:
>
> kamailio[26216]: WARNING: tm [t_lookup.c:1559]: t_unref(): WARNING: script
> writer didn't release transaction
>
> plus next line is written exactly 416000 times into a log afterwards:
>
> kamailio[32685]: CRITICAL: dialog [dlg_hash.c:794]: log_next_state_dlg():
> bogus event 4 in state 5 for dlg 0xb4af6588 [2575:7017] with clid
> '121d44f0-6555f4c8' and tags 'd12546d053aadc68o2' ''
>
> My point is to change the incoming code from users and append a Q.850
> reason code.
> Is there any other way how to do this or a way how to fix this?
> I'm using Kamilio 4.0.4 on Debian 7.1
>
>
> The code is as follows:
> failure_route[MANAGE_FAILURE]
> {
> if (t_is_canceled()) {
> exit;
> }
> if($T_reply_code == 408 && isflagset(10))
> {
> xlog("Ringing timeout");
> append_to_reply("Reason: Q.850;cause=28\r\n");
> t_reply("505", "Error");
> }
> }
>
>
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>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda
> - http://www.linkedin.com/in/miconda
> Kamailio Advanced Trainings - Berlin, Nov 25-28
> - more details about Kamailio trainings at http://www.asipto.com -
>
>
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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[SR-Users] t_reply in failure route with dialog module
Hi,
when I use t_reply("505", "Error"); in my failure route, the response is
not forwarded and following is written into a log:
kamailio[26216]: WARNING: tm [t_lookup.c:1559]: t_unref(): WARNING: script
writer didn't release transaction
plus next line is written exactly 416000 times into a log afterwards:
kamailio[32685]: CRITICAL: dialog [dlg_hash.c:794]: log_next_state_dlg():
bogus event 4 in state 5 for dlg 0xb4af6588 [2575:7017] with clid
'121d44f0-6555f4c8' and tags 'd12546d053aadc68o2' ''
My point is to change the incoming code from users and append a Q.850
reason code.
Is there any other way how to do this or a way how to fix this?
I'm using Kamilio 4.0.4 on Debian 7.1
The code is as follows:
failure_route[MANAGE_FAILURE]
{
if (t_is_canceled()) {
exit;
}
if($T_reply_code == 408 && isflagset(10))
{
xlog("Ringing timeout");
append_to_reply("Reason: Q.850;cause=28\r\n");
t_reply("505", "Error");
}
}
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Re: [SR-Users] Call_control after executing dlg_manage()
Thanks Daniel, I'm confused a little bit by the dialog module documentation. Please correct me if I'm wrong For initial invite I should create a dialog with dlg_manage "just" before t_relay. All dialog variables would be set correctly? How about dialog:failed route, would it work correctly? For in dialog requests (has_totag) I should call dlg_manage at the beginning of the script to have all dialog variables available. So far seems to be fine. However uac_replace_from auto restore feature does not work with this scenario. Should I move it right after dlg_manage like this:? dlg_manage(); uac_replace_from(); t_relay; I think e.g. mediaproxy also requires the dialog to be started before enagage_mediaproxy()? Aren't there any other functions bound to the dialog creation? I apologize for so many questions and if any of these questions is inappropriate I'm sorry. Thanks for clarifying answers. Efelin 2013/10/14 Daniel-Constantin Mierla > Hello, > > > On 10/14/13 10:07 AM, Efelin Novak wrote: > > > Hi folks, > > i have a problem with call_control module. I get a following log from > kamailio > > kamailio[15066]: WARNING: call_control [call_control.c:1159]: > postprocess_request(): dialog to trace controlled call was not created. > discarding callcontrol. > > and following log from call_control python application > > Call id of 1...@domain.com to sip:003300@domain.comcanceled by > user > > As far as I can see the call_control module runs a postprocess_request() > function which checks whether the dialog was created by checking > FL_USE_CALL_CONTROL variable. > > This variable is set at the time of the registration of call_controll to > the dialog module DLGCB_CREATED. > > Therefore if I call the call_control() function in kamailio after the > dlg_manage, the FL_USE_CALL_CONTROL cannot be set and the call_control > won't work. > > In my configuration I would like to call the dlg_manage at the beginning > of the script for example because of uac_replace_from as I have > AUTO_RESTORE mode ON. > > I have kamailio 4.0.2 and call_control 2.0.15. > > Is there anything I can fix in a script or configuration or this is an > implementation problem? > > > you can call dlg_manage() inside if conditions for initial requests and > within dialog requests. So for initial you have it towards the end of the > script. Anyhow, in both cases, do it as much as possible before relaying. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda > - http://www.linkedin.com/in/miconda > Kamailio Advanced Trainings - Berlin, Nov 25-28; Miami, Nov 18-20, 2013 > - more details about Kamailio trainings at http://www.asipto.com - > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Call_control after executing dlg_manage()
Hi folks, i have a problem with call_control module. I get a following log from kamailio kamailio[15066]: WARNING: call_control [call_control.c:1159]: postprocess_request(): dialog to trace controlled call was not created. discarding callcontrol. and following log from call_control python application Call id of 1...@domain.com to sip:0033000...@domain.com canceled by user As far as I can see the call_control module runs a postprocess_request() function which checks whether the dialog was created by checking FL_USE_CALL_CONTROL variable. This variable is set at the time of the registration of call_controll to the dialog module DLGCB_CREATED. Therefore if I call the call_control() function in kamailio after the dlg_manage, the FL_USE_CALL_CONTROL cannot be set and the call_control won't work. In my configuration I would like to call the dlg_manage at the beginning of the script for example because of uac_replace_from as I have AUTO_RESTORE mode ON. I have kamailio 4.0.2 and call_control 2.0.15. Is there anything I can fix in a script or configuration or this is an implementation problem? Thanks for the reply Efelin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Dispatcher module without OPTIONS keepalive pinging
Hi. Is there a possibility to turn off the pinging in dispatcher module for selected gateways? I want them to be active (to use them for dispatching) but I don't want to monitor them because one of my gateways does not support OPTIONS or INFO correctly. Kind regards Efelin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Acknowlidging BYE in advance using t_reply
Hi,
I'd like to ask you whether it is a good idea to acknowledge a BYE
message in advance using t_reply("200", "OK") on my border proxy. Not
acknowledged BYE messages by clients cause me problems in the middle
of my network.
Is there any problem with this behavior from Kamailio side, from
script side or from SIP side? Do you have any experience with such
logic manipulation? I am aware that this is not a proxy function, but
it would solve many problems.
My script is as follows:
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
t_newtran();
t_reply("200", "OK");
}
route(RELAY);
}
My test shows that 200 OK from client are absorbed. Also if there is
no reply from a client, Kamailio does not generate 408 Request
Timeout.
I'm using older version kamailio 3.1.0 (i386/linux) 21a375.
Thanks for an advice
Efelin
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[SR-Users] t_lookup.c: WARNING: script writer didn't release transaction - problem
Hi folks,
we have a serious issue with tm module. In Fail route we would like to
send a reply 404 when a cr_route function fails(no destination found).
We use t_reply("404", "No route to destination"). However after the
call is disconnected with a 404 reply the tm module writes a following
warning
WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction
Very soon after this error message is shown a segfault occurs. Log is
as follows:
Mar 3 15:26:46 hostname kamailio[791]: ERROR: carrierroute
[cr_func.c:594]: desired routing domain doesn't exist, prefix
00393, carrier 14, domain 2
Mar 3 15:26:46 hostname kamailio[791]: WARNING: tm [t_lookup.c:1543]:
WARNING: script writer didn't release transaction
//ACK is received
Mar 3 15:26:52 hostname kernel: [771134.093250] kamailio[796]:
segfault at 4 ip 0815c182 sp bf8ace00 error 6 in
kamailio[8048000+1ea000]
Mar 3 15:26:53 hostname kamailio[803]: : [pass_fd.c:293]:
ERROR: receive_fd: EOF on 15
Mar 3 15:26:53 hostname kamailio[790]: ALERT: [main.c:785]:
child process 796 exited by a signal 11
Mar 3 15:26:53 hostname kamailio[790]: ALERT: [main.c:788]:
core was generated
Mar 3 15:26:53 hostname kamailio[790]: INFO: [main.c:800]:
INFO: terminating due to SIGCHLD
Mar 3 15:26:53 hostname kamailio[802]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[799]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[801]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[798]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[800]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[797]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[792]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[803]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[795]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[791]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[794]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[793]: INFO: [main.c:851]:
INFO: signal 15 received
Mar 3 15:26:53 hostname kamailio[790]: INFO:
[mem/f_malloc.c:532]: freeing a free fragment (0x886b3ad8/0x886b3ae0)
- ignore
Mar 3 15:26:53 hostname kernel: [771135.681720] kamailio[790]:
segfault at 7dd193e0 ip 0816f5cd sp bf8acc20 error 4 in
kamailio[8048000+1ea000]
Currently we have fixed it using a custom route
route(T_REPLY);
.
.
.
route[T_REPLY] {
sl_send_reply("404","No route to destination");
}
This works correctly with no error or a warning message or a segfault.
We are using kamilio version: kamailio 3.3.4 (i386/linux) f8c8f2
Is this behavior a bug or should I dig deeper into the tm documentation?
Kind regards
Efelin
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[SR-Users] Changing outgoing/source port
Hi, I'd like to ask whether I am able to change an outgoing port. I'm using t_relay for outgoing INVITEs & Kamailio uses 5060 port defined in listen. When the call fails I am rerouting it using append_branch and I'd like to change an outgoing port to 5061. The reason I'd like to do this is a Loop detection of my buggy VoIP/analog gateway. Am I able to specify an outgoing interface/port for t_relay in Kamailio or should I find an another gateway? :) Thanks for help Kind regards Efelin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Multiple radius responses on a missed call
I'm little bit confused about a usage of radius acc_radius module.
Parameters from acc_radius module can also be found in acc module
after recompiling it with ENABLE_RADIUS_ACC=true.
Which one should I use? Which one is more up to date? What is the difference?
Cheers,
Efelin
2013/2/8 Efelin Novak :
> Thanks Daniel,
>
> I'll check it and post results here.
>
> Efelin
>
> 2013/2/8 Daniel-Constantin Mierla :
>> Hello,
>>
>> there are different event logged for accounting:
>> - missed call event which is sent from the point of view of callee
>> - transaction answered which is sent from the point of view of caller
>>
>> In db (eg mysql) accounting, each event in stored in a different table. I
>> don't know how they are in radius. You probably get two missed call events
>> and one transaction answered.
>>
>> If you want only one, do not set the flag for missed calls.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 2/7/13 9:23 PM, Efelin Novak wrote:
>>>
>>> Hi folks,
>>>
>>> I'd like to ask you how can I solve an issue when I'm receiving three
>>> RADIUS requests on one missed call. My scenario is as follows:
>>>
>>> A calls to B. B returns 503 error message, INIVITE is then sent to C
>>> and C sends 486.
>>>
>>> After first 503 I receive a first RADIUS request. This is correct.
>>>
>>> However after 486 I receive three RADIUS requests with different IDs.
>>> As you can see in a DEBUG one RADIUS is send after the SIP reply
>>> message is received and second and third RADIUS messages are sent
>>> after the SIP reply message is forwarded.
>>>
>>> I'd like to receive only two RADIUS messages. One after 503 and second
>>> after 486.
>>>
>>>
>>> FLT_ACC is defined as 1. My config is as follows:
>>> modparam("acc", "failed_transaction_flag", FLT_ACC)
>>> modparam("acc", "report_cancels", 1)
>>> modparam("acc", "report_ack", 0)
>>> modparam("acc", "detect_direction", 0)
>>> modparam("acc", "early_media",0)
>>> modparam("acc", "log_level", 5)
>>> modparam("acc", "log_flag", 1)
>>> modparam("acc", "log_missed_flag",1)
>>> modparam("acc_radius", "radius_config",
>>> "/etc/radiusclient-ng/radiusclient.conf") # This is the
>>> location of the configuration file of radius client
>>> modparam("acc_radius", "radius_flag",FLT_ACC)
>>> modparam("acc_radius", "radius_missed_flag", FLT_ACC)
>>>
>>> In a route I set setflag("FLT_ACC") only once per call.
>>>
>>> Here is the debug for debug level 7:
>>> 0(5890) DEBUG: [parser/msg_parser.c:634]: SIP Reply (status):
>>> 0(5890) DEBUG: [parser/msg_parser.c:636]: version:
>>> 0(5890) DEBUG: [parser/msg_parser.c:638]: status: <486>
>>> 0(5890) DEBUG: [parser/msg_parser.c:640]: reason:
>>> 0(5890) DEBUG: [parser/parse_via.c:1286]: Found param type
>>> 232, = ; state=6
>>> 0(5890) DEBUG: [parser/parse_via.c:1286]: Found param type
>>> 234, = <192.168.21.101>; state=16
>>> 0(5890) DEBUG: [parser/parse_via.c:2561]: end of header reached,
>>> state=5
>>> 0(5890) DEBUG: [parser/msg_parser.c:511]: parse_headers: Via
>>> found, flags=2
>>> 0(5890) DEBUG: [parser/msg_parser.c:513]: parse_headers: this
>>> is the first via
>>> 0(5890) DEBUG: [receive.c:149]: After parse_msg...
>>> 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=6
>>> global id=5 T start=0x
>>> 0(5890) DEBUG: [parser/parse_via.c:1286]: Found param type
>>> 232, = ; state=16
>>> 0(5890) DEBUG: [parser/parse_via.c:2561]: end of header reached,
>>> state=5
>>> 0(5890) DEBUG: [parser/msg_parser.c:511]: parse_headers: Via
>>> found, flags=22
>>> 0(5890) DEBUG: [parser/msg_parser.c:524]: parse_headers: this
>>> is the second via
>>> 0(5890) DEBUG: [parser/parse_via.c:1286]: Found param type
>>> 232, = ; state=6
>>> 0(5890) DEBUG: [parser/parse_via.c:1286]: Found param type
>>> 235, = <5060>; state=16
>>> 0(5890) DEBUG: [parser/parse_via.c:2561]: end of header reached,
>>> state=5
>>
Re: [SR-Users] Multiple radius responses on a missed call
Thanks Daniel,
I'll check it and post results here.
Efelin
2013/2/8 Daniel-Constantin Mierla :
> Hello,
>
> there are different event logged for accounting:
> - missed call event which is sent from the point of view of callee
> - transaction answered which is sent from the point of view of caller
>
> In db (eg mysql) accounting, each event in stored in a different table. I
> don't know how they are in radius. You probably get two missed call events
> and one transaction answered.
>
> If you want only one, do not set the flag for missed calls.
>
> Cheers,
> Daniel
>
>
> On 2/7/13 9:23 PM, Efelin Novak wrote:
>>
>> Hi folks,
>>
>> I'd like to ask you how can I solve an issue when I'm receiving three
>> RADIUS requests on one missed call. My scenario is as follows:
>>
>> A calls to B. B returns 503 error message, INIVITE is then sent to C
>> and C sends 486.
>>
>> After first 503 I receive a first RADIUS request. This is correct.
>>
>> However after 486 I receive three RADIUS requests with different IDs.
>> As you can see in a DEBUG one RADIUS is send after the SIP reply
>> message is received and second and third RADIUS messages are sent
>> after the SIP reply message is forwarded.
>>
>> I'd like to receive only two RADIUS messages. One after 503 and second
>> after 486.
>>
>>
>> FLT_ACC is defined as 1. My config is as follows:
>> modparam("acc", "failed_transaction_flag", FLT_ACC)
>> modparam("acc", "report_cancels", 1)
>> modparam("acc", "report_ack", 0)
>> modparam("acc", "detect_direction", 0)
>> modparam("acc", "early_media",0)
>> modparam("acc", "log_level", 5)
>> modparam("acc", "log_flag", 1)
>> modparam("acc", "log_missed_flag",1)
>> modparam("acc_radius", "radius_config",
>> "/etc/radiusclient-ng/radiusclient.conf") # This is the
>> location of the configuration file of radius client
>> modparam("acc_radius", "radius_flag",FLT_ACC)
>> modparam("acc_radius", "radius_missed_flag", FLT_ACC)
>>
>> In a route I set setflag("FLT_ACC") only once per call.
>>
>> Here is the debug for debug level 7:
>> 0(5890) DEBUG: [parser/msg_parser.c:634]: SIP Reply (status):
>> 0(5890) DEBUG: [parser/msg_parser.c:636]: version:
>> 0(5890) DEBUG: [parser/msg_parser.c:638]: status: <486>
>> 0(5890) DEBUG: [parser/msg_parser.c:640]: reason:
>> 0(5890) DEBUG: [parser/parse_via.c:1286]: Found param type
>> 232, = ; state=6
>> 0(5890) DEBUG: [parser/parse_via.c:1286]: Found param type
>> 234, = <192.168.21.101>; state=16
>> 0(5890) DEBUG: [parser/parse_via.c:2561]: end of header reached,
>> state=5
>> 0(5890) DEBUG: [parser/msg_parser.c:511]: parse_headers: Via
>> found, flags=2
>> 0(5890) DEBUG: [parser/msg_parser.c:513]: parse_headers: this
>> is the first via
>> 0(5890) DEBUG: [receive.c:149]: After parse_msg...
>> 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=6
>> global id=5 T start=0x
>> 0(5890) DEBUG: [parser/parse_via.c:1286]: Found param type
>> 232, = ; state=16
>> 0(5890) DEBUG: [parser/parse_via.c:2561]: end of header reached,
>> state=5
>> 0(5890) DEBUG: [parser/msg_parser.c:511]: parse_headers: Via
>> found, flags=22
>> 0(5890) DEBUG: [parser/msg_parser.c:524]: parse_headers: this
>> is the second via
>> 0(5890) DEBUG: [parser/parse_via.c:1286]: Found param type
>> 232, = ; state=6
>> 0(5890) DEBUG: [parser/parse_via.c:1286]: Found param type
>> 235, = <5060>; state=16
>> 0(5890) DEBUG: [parser/parse_via.c:2561]: end of header reached,
>> state=5
>> 0(5890) DEBUG: [parser/msg_parser.c:511]: parse_headers: Via
>> found, flags=22
>> 0(5890) DEBUG: [parser/parse_to.c:178]: DEBUG: add_param:
>> tag=as493d6134
>> 0(5890) DEBUG: [parser/parse_to.c:802]: end of header reached,
>> state=29
>> 0(5890) DEBUG: [parser/msg_parser.c:188]: DEBUG:
>> get_hdr_field: [42]; uri=[sip:5556003900...@dr.vm]
>> 0(5890) DEBUG: [parser/msg_parser.c:190]: DEBUG: to body
>> []
>> 0(5890) DEBUG: [parser/msg_parser.c:168]: get_hdr_field: cseq
>> : <102>
>> 0(5890) DEBUG: tm [t_lookup.c:965]: DEBUG: t_reply_matching: hash
>> 51672 label 59018528
[SR-Users] Kamailio ignores some ACK
Hi folks, I have a strange problem when Kamailio ignores ACKs in a specific scenario. The call flow is as follows: A -> INVITE -> kamailio -> INVITE -> B [omitting 100 and 180] A <- 200 OK <- kamailio <- 200 OK <- B A -> ACK -> kamailio There are INVITE Xlogs, Reply ROUTE xlogs and media-proxy logs in the syslog. However there is no information about these ACKs. No XLOGs are printed even if there is one on the top of the main route. "tcpdump -A -s0 -i any -n port 5060" receives this message correctly: 14:47:01.246153 IP 111.111.11.11.5060 > 80.80.80.80.60442: SIP, length: 915 SIP/2.0 200 OK Via: SIP/2.0/UDP 111.111.11.11:5060;rport=60442;x-route-tag="tgrp:A";branch=z9hG4bK1634E6A88 Record-Route: Contact: To: "test_account";tag=cb7dd641 From: ;tag=599248D4-260 Call-ID: 9AFCFC51.11.50 CSeq: 101 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length:263 v=0 o=- 492575093 492575093 IN IP4 111.111.11.60 s=test_device i=(o=IN IP4 192.168.1.10) c=IN IP4 111.111.11.71 t=0 0 m=audio 16416 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 14:47:01.254511 IP 111.111.11.50.60442 > 111.111.11.11.5060: SIP, length: 521 ACK sip:80.80.80.80:65002;transport=udp SIP/2.0 Via: SIP/2.0/UDP 111.111.11.50:5060;x-route-tag="tgrp:A";branch=z9hG4bK1634E7DE8 From: ;tag=599248D4-260 To: "test_account";tag=cb7dd641 Call-ID: 9AFCFC51.11.50 Route: Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 My Kamailio version is kamailio 3.1.0 (i386/linux) 1e204f. Does anybody knows where can be a problem? How can I check whether Kamailio receives something? ... Jan ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Topoh hiding prefixes
I just want to know whether installing of the TOPOH modul would somehow affect users of my proxy? When the UAC creates (at least blink does so and few hardware phones I have) the seq. request, it places a received contact from 200OK (INVITE transaction) into a Request URI of the seq.request. So when the UAC replies back with the ACK its Request URI is "hidden contact URI from 200OK". As the TOPOH module do not translates this ACK Request URI back, it is being sent to UAS unchanged (hidden contact of UAS). It works actually in the testing topology. Are there any clients that should have a problem with this? Is this behavior described in some RFC? Thanks Efelin 2012/2/17 Alex Balashov : > On 02/17/2012 11:30 AM, Efelin Novak wrote: > >> But the contact from the UAS is hidden to the UAC. So the UAS have to >> put a "hidden" contact to the R-URI. Should the TOPOH module also >> "fallback" the request uri? I dont think so. So the client has to >> receive ACK with a different r-uri that the contact he sent in 200OK. > > > I am unable to understand this statement. > > However, the client does not receive ACKs. > > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Topoh hiding prefixes
But the contact from the UAS is hidden to the UAC. So the UAS have to put a "hidden" contact to the R-URI. Should the TOPOH module also "fallback" the request uri? I dont think so. So the client has to receive ACK with a different r-uri that the contact he sent in 200OK. Efelin 2012/2/17 Alex Balashov : > On 02/17/2012 10:33 AM, Efelin Novak wrote: > >> However don't you have any experinece with clients that have >> problems with seq. requests being sent to some "random" string >> expect of the propagated contact? > > > Not sure I follow. The initiating UA is obligated to use whatever contact > is provided by the UAS for sequential requests. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Topoh hiding prefixes
I have to apologize. The Topoh module hides the whole contact not just a domain part. So The problem is fixed. However don't you have any experinece with clients that have problems with seq. requests being sent to some "random" string expect of the propagated contact? UA ---invite ---> kamailio(topoh) ---invite ---> UA UA <---200ok--- kamailio(topoh) <---200ok--- UA (sent contact= 100@192.168.1.101) UA ---ACK---> kamailio(topoh) ---ACK---> UA (received ru= sip:;line=sr--random_string SIP/2.0 ) Also fix_nated_contact() destroys the contact so I suppose there should be no problem whit this. Thank anyway Efelin 2012/2/17 Efelin Novak : > Hi Alex, > > thanks for the answer. > > My customer is on the SIP side. So my customer sees a prefix of my > PSTN gateway in the contact header. Sequential requests are being sent > to @. However I think when I remove the > prefix from the contact, the gateway would have no problem in > responding to this seq. request (Isn't the call defined by to_tag, > from_tag, call_id?). > > So I want to hide the contact prefix from the UA. > > Efelin > > 2012/2/17 Alex Balashov : >> Efelin, >> >> There is probably some confusion here. The 200 OK message is from your >> gateway (in your call flow), and it can put whatever it wants in the Contact >> URI; you have no control over that on the Kamailio side. You might be able >> to influence it with configuration settings on the GW side, but most likely >> not. >> >> However, much more importantly: the contact URI in the replies from the >> gateway only serves to target your UA's sequential requests. It has nothing >> to do with the other side of your GW. If it's a SIP -> TDM or analog GW, >> then SIP details are totally irrelevant. If it's a SIP-to-SIP media >> gateway, it's a back-to-back user agent, meaning it generates a whole new >> logical call leg on the other side and bridges it to the incoming one from >> the UA. The parameters in that new leg are completely independent of any >> attributes of the old one. >> >> Regardless, the contact URI is something sent back to the UA to tell it how >> to communicate with the GW. In the overall scheme of your call flow, no >> matter the media, it is a "purely internal" attribute. >> >> -- >> This message was painstakingly thumbed out on my mobile, so apologies for >> brevity and errors. >> >> Alex Balashov - Principal >> Evariste Systems LLC >> 260 Peachtree Street NW >> Suite 2200 >> Atlanta, GA 30303 >> Tel: +1-678-954-0670 >> Fax: +1-404-961-1892 >> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >> >> On Feb 17, 2012, at 9:20 AM, Efelin Novak wrote: >> >>> Hi folks, >>> >>> I have a topology like this: >>> >>> UA -> kamailio -> PSTN GW. >>> >>> When I'm placing calls to a PSTN I have to append prefixes so the PSTN >>> GW knows how to route the call. So the request-uri is >>> @ e.g. 999123456@192.168.1.1. When the UA >>> calls to the GW a "200 OK" message contains a contact header with >>> 999123456@192.168.1.1. This is a problem because I don't want my >>> customers to see gateway prefixes. >>> >>> The Topoh module only hides IP addresses or the domain part. Is there >>> any way to remove prefixes. This document >>> (http://www.kamailio.org/events/2011-Cluecon/DCM-kamailio-security.pdf) >>> stated "encoding IP and prefixes can be set via parameters". Can I >>> remove prefixes from the contact in any other way than some contact >>> header substitutions? >>> >>> Thanks >>> >>> Efelin >>> >>> ___ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Topoh hiding prefixes
Hi Alex, thanks for the answer. My customer is on the SIP side. So my customer sees a prefix of my PSTN gateway in the contact header. Sequential requests are being sent to @. However I think when I remove the prefix from the contact, the gateway would have no problem in responding to this seq. request (Isn't the call defined by to_tag, from_tag, call_id?). So I want to hide the contact prefix from the UA. Efelin 2012/2/17 Alex Balashov : > Efelin, > > There is probably some confusion here. The 200 OK message is from your > gateway (in your call flow), and it can put whatever it wants in the Contact > URI; you have no control over that on the Kamailio side. You might be able > to influence it with configuration settings on the GW side, but most likely > not. > > However, much more importantly: the contact URI in the replies from the > gateway only serves to target your UA's sequential requests. It has nothing > to do with the other side of your GW. If it's a SIP -> TDM or analog GW, > then SIP details are totally irrelevant. If it's a SIP-to-SIP media gateway, > it's a back-to-back user agent, meaning it generates a whole new logical call > leg on the other side and bridges it to the incoming one from the UA. The > parameters in that new leg are completely independent of any attributes of > the old one. > > Regardless, the contact URI is something sent back to the UA to tell it how > to communicate with the GW. In the overall scheme of your call flow, no > matter the media, it is a "purely internal" attribute. > > -- > This message was painstakingly thumbed out on my mobile, so apologies for > brevity and errors. > > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com > > On Feb 17, 2012, at 9:20 AM, Efelin Novak wrote: > >> Hi folks, >> >> I have a topology like this: >> >> UA -> kamailio -> PSTN GW. >> >> When I'm placing calls to a PSTN I have to append prefixes so the PSTN >> GW knows how to route the call. So the request-uri is >> @ e.g. 999123456@192.168.1.1. When the UA >> calls to the GW a "200 OK" message contains a contact header with >> 999123456@192.168.1.1. This is a problem because I don't want my >> customers to see gateway prefixes. >> >> The Topoh module only hides IP addresses or the domain part. Is there >> any way to remove prefixes. This document >> (http://www.kamailio.org/events/2011-Cluecon/DCM-kamailio-security.pdf) >> stated "encoding IP and prefixes can be set via parameters". Can I >> remove prefixes from the contact in any other way than some contact >> header substitutions? >> >> Thanks >> >> Efelin >> >> ___ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Topoh hiding prefixes
Hi folks, I have a topology like this: UA -> kamailio -> PSTN GW. When I'm placing calls to a PSTN I have to append prefixes so the PSTN GW knows how to route the call. So the request-uri is @ e.g. 999123456@192.168.1.1. When the UA calls to the GW a "200 OK" message contains a contact header with 999123456@192.168.1.1. This is a problem because I don't want my customers to see gateway prefixes. The Topoh module only hides IP addresses or the domain part. Is there any way to remove prefixes. This document (http://www.kamailio.org/events/2011-Cluecon/DCM-kamailio-security.pdf) stated "encoding IP and prefixes can be set via parameters". Can I remove prefixes from the contact in any other way than some contact header substitutions? Thanks Efelin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to test registration setting
Thank you Andreas,
it worked like a charm.
Efelin
2011/12/7 Andreas Granig :
> Hi,
>
> On 12/07/2011 09:45 AM, Efelin Novak wrote:
>> So is there any other common practice how to test registers?
>> Is there any flag that can prevent storing location?
>> Can I send 200 OK to these user in any other way?
>
> You can do sl_send_reply("200", "Auth test ok") after
> www_authenticate(...) succeeds.
>
> Andreas
>
>
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[SR-Users] How to test registration setting
Hi folks, I would like to know what to use for the following scenario. I want users to be able to test their registration by sending REGISTER and Kamailio sends them 200 OK if credentials are OK. However I don't want the aor to be stored in location table. This REGISTER should be only used for testing the settings. After some time I would allow the registration to be stored and users will normally receive calls. I tried to implement this by not doing save(location) when the flag from load_credential is set. Unfortunately it is the function save who sends 200 OK. When it is not being called, a user agent keeps sending REGISTER until SIP triggers fail. So is there any other common practice how to test registers? Is there any flag that can prevent storing location? Can I send 200 OK to these user in any other way? Thanks Efelin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] lookup for specified user
Thanks,
the branch route worked well.
Efelin
2011/8/12 Klaus Darilion :
>
>
> Am 12.08.2011 14:31, schrieb Efelin Novak:
>> Hi thanks for answers,
>>
>> unfortunately I need to send original RURI username (rU) to the user.
>> Setting up aliases_db I came to this situation:
>>
>
> 0. $var(temp) = $rU;
>> 1. RURI: j...@domain.com
>> 2. alias_db_lookup("dbaliases")
>> 3. RURI: x...@domain.com
>> 4. lookup("location")
>> 5. RURI: X@192.168.1.2
> 6. $rU = $var(temp);
> 7. RURI: john@192.168.1.2
>
> 6. only modifies the main branch unless you do it in a branch route.
>
>
> regards
> Klaus
>
>>
>> I would like to have john@192.168.1.2 at step 5.
>>
>> Is it possible somehow?
>>
>> 2011/8/11 Klaus Darilion :
>>>
>>>
>>> Am 10.08.2011 15:17, schrieb Juha Heinanen:
>>>> Efelin Novak writes:
>>>>
>>>>> I would like to change uri like lookup() function does it, but for the
>>>>> specific user. Problem is I want to have one registration (X) for
>>>>> several numbers. When this number is called I have to perform lookup()
>>>>> for this one common registration (X) and send this call there.
>>>>
>>>> perhaps you could define the other numbers as aliases for X.
>>>
>>> some more hints:
>>> Use this function before lookup() to map the user's "aliases" to the
>>> user's main identity (under which the registration will be done).
>>>
>>> http://www.kamailio.org/docs/modules/3.1.x/modules_k/alias_db.html#id2916034
>>>
>>> regards
>>> klaus
>>>
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>>>
>>
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Re: [SR-Users] lookup for specified user
Hi thanks for answers,
unfortunately I need to send original RURI username (rU) to the user.
Setting up aliases_db I came to this situation:
1. RURI: j...@domain.com
2. alias_db_lookup("dbaliases")
3. RURI: x...@domain.com
4. lookup("location")
5. RURI: X@192.168.1.2
I would like to have john@192.168.1.2 at step 5.
Is it possible somehow?
2011/8/11 Klaus Darilion :
>
>
> Am 10.08.2011 15:17, schrieb Juha Heinanen:
>> Efelin Novak writes:
>>
>>> I would like to change uri like lookup() function does it, but for the
>>> specific user. Problem is I want to have one registration (X) for
>>> several numbers. When this number is called I have to perform lookup()
>>> for this one common registration (X) and send this call there.
>>
>> perhaps you could define the other numbers as aliases for X.
>
> some more hints:
> Use this function before lookup() to map the user's "aliases" to the
> user's main identity (under which the registration will be done).
>
> http://www.kamailio.org/docs/modules/3.1.x/modules_k/alias_db.html#id2916034
>
> regards
> klaus
>
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[SR-Users] lookup for specified user
Hi folks, I discussed this issue in previous email ( [SR-Users Modifying $rU after lookup() ) but the question is a bit different now. I would like to change uri like lookup() function does it, but for the specific user. Problem is I want to have one registration (X) for several numbers. When this number is called I have to perform lookup() for this one common registration (X) and send this call there. To do this I changed $rU just before lookup() for X, and after lookup() I changed it back for the original number (both changes were done using variables $rU). However when there are two registrations (X1, X2), only X1 can be changed for the original number and the second one (X2) is sent to user without modification. I also found function reg_fetch_contacts(domain, uri, profile) but in documentation there is $ulc variable mentioned. This variable is not documented. Is there any way how to perform lookup for specific user? Thanks for the answer. Regards, Efelin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Modifying $rU after lookup()
Hi Folks,
I would like to ask how can I change the $rU (user part of uri) in multiple
appended branches?
In my scenario I do lookup("location") and after this I try to modify an
username using the $rU variable. This works fine until there are several
records in the location table for the given user (registered from different
locations). In this case only one branch is changed and unchanged uri is
sent to an another location.
Is there a way to change it in all branches together?
Thanks for the answer.
Regards,
Efelin
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Re: [SR-Users] Registrar module creates an extra branch
Hi Klaus,
thanks for a prompt answer.
Third INVITE (extra) is sent to the blink, but only if the Blnik is second
in the location table. When the Linksys is second in the list, only two
INVITES are being sent.
No nothing is looping, it acts like there are 3 separate rows in the
location table.
Kamailio creates 3 branches from two AOR. Extra INVITE differs from the
original one only in via branch x.{0,1,2}.
Regards,
Efelin Novak
2011/3/14 Klaus Darilion
> Where is the 3rd INVITE sent to? to blink, to linksys or somewhere else?
>
> Is something looping? (ngrep -d lo port 5060)
>
> klaus
>
> Am 14.03.2011 18:38, schrieb Efelin Novak:
> > Hi Folks,
> > I have a problem with registrar module (I suppose). When A calls to B,
> > is some specific situation, B receives two INVITEs. This mostly (not
> > only) happens when one account is registered on two or more UAs. In this
> > situation extra INVITE is sent only to the one that is second in the
> > list (kamctl ul show ACCOUNT).
> >
> > kamctl ul show Alice:
> > Contact:: > <http://sip:orlvuzyd@192.168.28.72:52129
> >>;q=0;expires=1916;flags=0x0;cflags=0x40;socket= > <http://8.8.8.8:5060>>;methods=0x;received= > <http://8.8.8.8:52129>>;user_agent=
> > Contact:: > <http://sip:Alice@192.168.28.51:5060
> >>;q=;expires=3061;flags=0x0;cflags=0x40;socket= > <http://8.8.8.8:5060>>;methods=0x129F;received= > <http://8.8.8.8:5060>>;user_agent=
> >
> >
> > My situation.
> > 1. Alice calls to BOB.
> > 2. kamailio looks up in location table for BOBs registrations
> > 3. in DB there are two rows (two devices(linksys PAP2T and Blink(blink
> > is second one)))
> > 4. kamailio send 3 INVITES ( linksys, blink and blink). Every INVITE has
> > different via.branch
> > (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.1)
> > (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.2)
> > (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.3)
> >
> > All my registrar parameters are set to default.
> >
> > I have no idea whether this is a bug or my missconfiguration.
> >
> > Thanks for replies.
> >
> > Kind regards,
> >
> > Efelin Novak
> >
> >
> >
> >
> >
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[SR-Users] Registrar module creates an extra branch
Hi Folks, I have a problem with registrar module (I suppose). When A calls to B, is some specific situation, B receives two INVITEs. This mostly (not only) happens when one account is registered on two or more UAs. In this situation extra INVITE is sent only to the one that is second in the list (kamctl ul show ACCOUNT). kamctl ul show Alice: Contact:: ;q=0;expires=1916;flags=0x0;cflags=0x40;socket=;methods=0x;received=;user_agent= Contact:: ;q=;expires=3061;flags=0x0;cflags=0x40;socket=;methods=0x129F;received=;user_agent= My situation. 1. Alice calls to BOB. 2. kamailio looks up in location table for BOBs registrations 3. in DB there are two rows (two devices(linksys PAP2T and Blink(blink is second one))) 4. kamailio send 3 INVITES ( linksys, blink and blink). Every INVITE has different via.branch (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.1) (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.2) (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.3) All my registrar parameters are set to default. I have no idea whether this is a bug or my missconfiguration. Thanks for replies. Kind regards, Efelin Novak ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Calls are killed during kamailio restart
Hi, I'd like to ask whether my situation is normal. During kamailio restart calls are dropped from mediaproxy. Those two programs are connected using kamailio.sock. Why RTP strem is dropped when SIP proxy is restarted? I would expect just undelivered BYE or something. Thanks for answer, Efelin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] handle_new_connect ... Too many open files
Hi, today during two hours kamailio made 2G of a syslog messages. This message was repeated for 10 000 000 times. /sbin/kamailio[24272]: ERROR: [tcp_main.c:3959]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files It was preceded by following error. /sbin/kamailio[24265]: ERROR: [tcp_read.c:269]: error reading: Connection reset by peer (104) /sbin/kamailio[24265]: ERROR: [tcp_read.c:882]: ERROR: tcp_read_req: error reading Finally it stopped without any given reason. I'm using kamailio 3.1.0 (i386/linux) 1e204f, callcontrol 2.0.11 and clustered mysql. Was this some kind of attack, bug, error, misconfiguration or what? Thank for answer, Efelin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Billing party and radius accounting
Unfortunately I was wrong about variables ($var). They are same across whole kamailio process. Neither $avp is good for me as they are transaction aware. I need to store variable in the current dialog. I was looking into dialog module, but no luck. I wasn't able to make dialog variables $dlg and $dlg-ctx work. Are they appropriate in this situation? Can you please give here some examples how to use them in configuration file? Regards, Efelin 2010/10/28 Efelin Novak > O, actually I have just found out that variables are stored across whole > call. Thank you for your answers. > > Regards, > Efelin > > 2010/10/28 Juha Heinanen > >> Efelin Novak writes: >> >> >> > The request uri after redirect will be us...@another_domain. I have to >> store >> > the original request_uri(us...@my_domain) and after redirect, insert it >> to >> > Billing Party header in radius message. >> >> yes, i had a typo in my answer. you, of course, need to use request uri >> before redirect as your billing uri. i wonder why you asked, when you >> knew the answer yourself. >> >> -- juha >> > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Billing party and radius accounting
O, actually I have just found out that variables are stored across whole call. Thank you for your answers. Regards, Efelin 2010/10/28 Juha Heinanen > Efelin Novak writes: > > > The request uri after redirect will be us...@another_domain. I have to > store > > the original request_uri(us...@my_domain) and after redirect, insert it > to > > Billing Party header in radius message. > > yes, i had a typo in my answer. you, of course, need to use request uri > before redirect as your billing uri. i wonder why you asked, when you > knew the answer yourself. > > -- juha > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Billing party and radius accounting
The request uri after redirect will be us...@another_domain. I have to store the original request_uri(us...@my_domain) and after redirect, insert it to Billing Party header in radius message. I need UserC not to be aware of the redirect. In this case I have to make a new call to UserC. This call is to the another network which I have to pay. I have to bill it to UserB. Regards, Efelin 2010/10/28 Juha Heinanen > Efelin Novak writes: > > > thanks for answer. Yes, my server is proxy. Situation is like this: > > > > us...@domain --> us...@my_domain (redirect) > > > us...@another_domain > > > > I have to bill call to another_domain to UserB. > > then you must make sure that your billing uri in the accounting record > gets populated by uri of UserB, i.e., the request uri after redirect. > > -- juha > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Billing party and radius accounting
Hello Juha thanks for answer. Yes, my server is proxy. Situation is like this: us...@domain --> us...@my_domain (redirect) > us...@another_domain I have to bill call to another_domain to UserB. Regards, Efelin 2010/10/27 Juha Heinanen > Efelin Novak writes: > > > I have a problem with setting billing party in my radius messages. My > > accounting is done by CDRTool via radius messages to radius server. The > > thing I don`t know is how to set billing party variable (by default > > $avp(s:billing_party) with reliable value. I can try adding Diversion > header > > and fill the variable with it, but can I trust the Diversion header? When > > UAC sets redirect(302) and Invite comes, he replies 302 and Kamailio > passes > > this message to origin. However is it possible to change Diversion header > by > > UAC? This could be way how to make free calls and bill it to someone > > else. > > you bill the uac who sends an invite that results to 200 ok reply, > not the party that replies with 302 (unless your proxy converts 302 to > invite in which case you bill the uas). > > -- juha > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
