[OpenSIPS-Users] MOH / rtpengine
Hi
when call is being trasfered to another number MS Teams sends new Invite
with SDP as 'a=inactive'. How can I put ringback ton as MOH for this
sitation?
I tried with:
if(is_audio_on_hold()) {
xlog("L_INFO", "onHOLD");
rtpengine_play_media("file=/home/ringback.wav");
}
From logs i can see that due to a=inactive rtpengine will not play
media. I tried also to replace inactive with sendonly with function
body_replace before I call rtpengine_play_media but it does not help.
thank you
miha
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Re: [OpenSIPS-Users] replace_body() issue
I have without ^ and $ before, i have now tried again like you suggested
but it does not work.
It goes in the if loop ("search_body") but it does not replace inactive
with sendonly.
I have also look at return code from replace_body and it gives me "-1".
Stas Kobzar je 6/18/2021 ob 1:04 PM napisal:
Hello,
Just do not use ^ and $ in the search pattern. It is probably trying
to match the whole SDP packet, not single line.
On Fri, Jun 18, 2021 at 5:09 AM Miha via Users
mailto:users@lists.opensips.org>> wrote:
Hello
have issue with replace_body as it does not change SDP.
My code looks like this:
if (has_body("application/sdp")){
if(search_body("a=inactive")){
*replace_body("^a=inactive$", "a=sendonly");*
}
$var(rtpengine_flags) ="trust-address
replace-origin replace-session-connection ICE=remove RTP/AVP
rtcp-mux-demux";
rtpengine_offer("$var(rtpengine_flags)");
if(is_audio_on_hold()) {
rtpengine_play_media("callee
file=/home/ringback.wav");
}
t_on_reply("1");
}
What could be wrong that inactive is not replaced by sendonly?
On a leg I can see "a=inactive" and also on b leg "a=inactive".
thank you
miha
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[OpenSIPS-Users] replace_body() issue
Hello
have issue with replace_body as it does not change SDP.
My code looks like this:
if (has_body("application/sdp")){
if(search_body("a=inactive")){
*replace_body("^a=inactive$", "a=sendonly");*
}
$var(rtpengine_flags) ="trust-address replace-origin
replace-session-connection ICE=remove RTP/AVP rtcp-mux-demux";
rtpengine_offer("$var(rtpengine_flags)");
if(is_audio_on_hold()) {
rtpengine_play_media("callee file=/home/ringback.wav");
}
t_on_reply("1");
}
What could be wrong that inactive is not replaced by sendonly?
On a leg I can see "a=inactive" and also on b leg "a=inactive".
thank you
miha
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[OpenSIPS-Users] ACK with wrong RURI
Hello when call is not pickup on teams and just canceled I get 603 Declined which opensips send it further to SBC (our main sbc). When we get back ACK and from SBC, there is wrong RURI and opensips does not relay this ACK to MS teams. Is anything can be done on opensips side in this situation so that i will not have this issue? SIP/2.0 603 Decline FROM: ;tag=3832669067-409068083 TO: ;tag=7cb08507d0944d7a838dfdc418227564 CSEQ: 1 INVITE CALL-ID: 901735-3832669067-641737...@sbc1.test.com VIA: SIP/2.0/UDP SBC:5060;branch=z9hG4bK029c59d110eb800c4e70bac6d27181cb REASON: Q.850;cause=21;text="84fbb35a-99e1-4ae7-b6cd-12d428c96190;CallEndReasonLocalUserInitiated" RECORD-ROUTE: ,, CONTACT: CONTENT-LENGTH: 0 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY SERVER: Microsoft.PSTNHub.SIPProxy v.2021.5.28.7 i.EUWE.6 ACK sip:87654321@OPENSIPS_IP SIP/2.0 Max-Forwards: 69 To: ;tag=7cb08507d0944d7a838dfdc418227564 From: ;tag=3832669067-409068083 Call-ID: 901735-3832669067-641737...@sbc1.softnet.si CSeq: 1 ACK Via: SIP/2.0/UDP SBC:5060;branch=z9hG4bK029c59d110eb800c4e70bac6d27181cb Contact: Content-Length: 0 thank you miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MS teams, reinvite after ACK
yes. Thank you.
is there any way to get also attended transfer working?
Johan De Clercq je 6/2/2021 ob 11:21 AM napisal:
remove Refer from your supported methods.
Do note that attended transfer will not work in this case.
wkr,
Op wo 2 jun. 2021 om 10:15 schreef Miha via Users
mailto:users@lists.opensips.org>>:
Hello
i manage to fix this. I did not do t_relay() also seq Invites,
after this everything works ok.
Just on question, regarding transfers, i see that MS Teams send
REFER in which trafter is defined. How do you deal with this? You
do not allow REFER from MS teams and hope that MS teams will send
new INVITE?
thank you
miha
Jeff Pyle je 6/1/2021 ob 3:26 PM napisal:
Miha,
First, do you need to use "mtsbc.test.com:5060
<http://mtsbc.test.com:5060>" in the first record_route_preset()
param? Can you use the IP address of your proxy instead? FQDNs
are legal of course, but outside of MS Teams' implementation,
they're rarely required. It's just another thing to go wrong.
Especially while testing.
The ACK to the 200 OK is a sequential (in-dialog) request. It's
not part of the original INVITE transaction. Your script will
have a section like
if (has_totag()) {
if (loose_route()) {
t_relay();
}
}
for sequential requests through a loose-routing proxy. This is
very oversimplified and yours will have more. In this section,
however, is where you'll process the ACK because it has a to-tag
(line 293) and a route header (line 298) so the conditions match.
Use xlogs or the debug tool of your choice to diagnose what's
happening in this section with the ACK. In my scripts, I use
global flag 0 to indicate when I want logging. So, I might have
something like this:
if (has_totag()) {
if (is_gflag(0)) xlog("L_NOTICE", "...in-dialog
$rm request\n");
# ...do all the things...maybe more logging like the line above...
- Jeff
On Tue, Jun 1, 2021 at 4:57 AM Miha via Users
mailto:users@lists.opensips.org>> wrote:
Hello
I have an issue and I am unable to find out what is wrong.
Incoming calls are working but when doing outbound call after
200OK, which is send to Teams I get back ACK and after that
Teams do again initial. I guess this is not ok.
I am doing this for outband calls:
xlog("L_INFO", "rtp rtps record route");
record_route_preset("mtsbc.test.com:5060
<http://mtsbc.test.com:5060>","mtsbc.test.com
<http://mtsbc.test.com>:5061;transport=tls");
add_rr_param(";r2=on");
I am pasting here trace. Opensips is in the middle.
Thank you for help!
https://pastebin.com/qM0dMiCc <https://pastebin.com/qM0dMiCc>
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Re: [OpenSIPS-Users] MS teams, reinvite after ACK
ok, it does new seq invite, so not is is working.
thank you for help.
miha
Miha via Users je 6/2/2021 ob 10:11 AM napisal:
Hello
i manage to fix this. I did not do t_relay() also seq Invites, after
this everything works ok.
Just on question, regarding transfers, i see that MS Teams send REFER
in which trafter is defined. How do you deal with this? You do not
allow REFER from MS teams and hope that MS teams will send new INVITE?
thank you
miha
Jeff Pyle je 6/1/2021 ob 3:26 PM napisal:
Miha,
First, do you need to use "mtsbc.test.com:5060
<http://mtsbc.test.com:5060>" in the first record_route_preset()
param? Can you use the IP address of your proxy instead? FQDNs are
legal of course, but outside of MS Teams' implementation, they're
rarely required. It's just another thing to go wrong. Especially
while testing.
The ACK to the 200 OK is a sequential (in-dialog) request. It's not
part of the original INVITE transaction. Your script will have a
section like
if (has_totag()) {
if (loose_route()) {
t_relay();
}
}
for sequential requests through a loose-routing proxy. This is very
oversimplified and yours will have more. In this section, however,
is where you'll process the ACK because it has a to-tag (line 293)
and a route header (line 298) so the conditions match.
Use xlogs or the debug tool of your choice to diagnose what's
happening in this section with the ACK. In my scripts, I use global
flag 0 to indicate when I want logging. So, I might have something
like this:
if (has_totag()) {
if (is_gflag(0)) xlog("L_NOTICE", "...in-dialog $rm
request\n");
# ...do all the things...maybe more logging like the line above...
- Jeff
On Tue, Jun 1, 2021 at 4:57 AM Miha via Users
mailto:users@lists.opensips.org>> wrote:
Hello
I have an issue and I am unable to find out what is wrong.
Incoming calls are working but when doing outbound call after
200OK, which is send to Teams I get back ACK and after that Teams
do again initial. I guess this is not ok.
I am doing this for outband calls:
xlog("L_INFO", "rtp rtps record route");
record_route_preset("mtsbc.test.com:5060
<http://mtsbc.test.com:5060>","mtsbc.test.com
<http://mtsbc.test.com>:5061;transport=tls");
add_rr_param(";r2=on");
I am pasting here trace. Opensips is in the middle.
Thank you for help!
https://pastebin.com/qM0dMiCc <https://pastebin.com/qM0dMiCc>
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Re: [OpenSIPS-Users] MS teams, reinvite after ACK
Hello
i manage to fix this. I did not do t_relay() also seq Invites, after
this everything works ok.
Just on question, regarding transfers, i see that MS Teams send REFER in
which trafter is defined. How do you deal with this? You do not allow
REFER from MS teams and hope that MS teams will send new INVITE?
thank you
miha
Jeff Pyle je 6/1/2021 ob 3:26 PM napisal:
Miha,
First, do you need to use "mtsbc.test.com:5060
<http://mtsbc.test.com:5060>" in the first record_route_preset()
param? Can you use the IP address of your proxy instead? FQDNs are
legal of course, but outside of MS Teams' implementation, they're
rarely required. It's just another thing to go wrong. Especially
while testing.
The ACK to the 200 OK is a sequential (in-dialog) request. It's not
part of the original INVITE transaction. Your script will have a
section like
if (has_totag()) {
if (loose_route()) {
t_relay();
}
}
for sequential requests through a loose-routing proxy. This is very
oversimplified and yours will have more. In this section, however, is
where you'll process the ACK because it has a to-tag (line 293) and a
route header (line 298) so the conditions match.
Use xlogs or the debug tool of your choice to diagnose what's
happening in this section with the ACK. In my scripts, I use global
flag 0 to indicate when I want logging. So, I might have something
like this:
if (has_totag()) {
if (is_gflag(0)) xlog("L_NOTICE", "...in-dialog $rm
request\n");
# ...do all the things...maybe more logging like the line above...
- Jeff
On Tue, Jun 1, 2021 at 4:57 AM Miha via Users
mailto:users@lists.opensips.org>> wrote:
Hello
I have an issue and I am unable to find out what is wrong.
Incoming calls are working but when doing outbound call after
200OK, which is send to Teams I get back ACK and after that Teams
do again initial. I guess this is not ok.
I am doing this for outband calls:
xlog("L_INFO", "rtp rtps record route");
record_route_preset("mtsbc.test.com:5060
<http://mtsbc.test.com:5060>","mtsbc.test.com
<http://mtsbc.test.com>:5061;transport=tls");
add_rr_param(";r2=on");
I am pasting here trace. Opensips is in the middle.
Thank you for help!
https://pastebin.com/qM0dMiCc <https://pastebin.com/qM0dMiCc>
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[OpenSIPS-Users] MS teams, reinvite after ACK
Hello
I have an issue and I am unable to find out what is wrong. Incoming
calls are working but when doing outbound call after 200OK, which is
send to Teams I get back ACK and after that Teams do again initial. I
guess this is not ok.
I am doing this for outband calls:
xlog("L_INFO", "rtp rtps record route");
record_route_preset("mtsbc.test.com:5060","mtsbc.test.com:5061;transport=tls");
add_rr_param(";r2=on");
I am pasting here trace. Opensips is in the middle.
Thank you for help!
https://pastebin.com/qM0dMiCc
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Re: [OpenSIPS-Users] TLS to UDP, record route
Thank you I will check. Br miha On 18 May 2021, 13:08 +0200, John Quick , wrote: > The client I was working with used this: > https://docs.microsoft.com/en-us/microsoftteams/direct-routing-sbc-multiple-tenants > > I touched on the topic in my article about MS Teams, under the heading > "Terminology and multi-tenant solutions" > > John Quick > Smartvox Limited > > > From: Miha > > Sent: 18 May 2021 11:59 > > To: john.qu...@smartvox.co.uk; users@lists.opensips.org > > Subject: Re: [OpenSIPS-Users] TLS to UDP, record route > > > > btw what is the trick if you have multiple trunks to sbc teams (inbound, > > outbound)? multiple companies? > > > > miha > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] TLS to UDP, record route
btw what is the trick if you have multiple trunks to sbc teams (inbound,
outbound)? multiple companies?
br
miha
John Quick je 5/18/2021 ob 11:15 AM napisal:
Miha
Altering the text in the Record-Route headers with subst() function is not
the correct approach.
I believe the problem is that you are not inserting the correct RR headers
in the first place.
If you get the RR headers right then it will also fix the problem with ACK
using the wrong protocol.
This pseudo-code snippet illustrates what is required when adding RR headers
to the initial INVITE request:
if (INVITE-from-Teams-Proxy-to-us) {
record_route_preset("IP:port", "SBC_FQDN:5061;transport=tls");
add_rr_param(";r2=on");
} else if (INVITE-from-us-to-Teams-Proxy) {
record_route_preset( "SBC_FQDN:5061;transport=tls", "IP:port");
add_rr_param(";r2=on");
} else
record_route();
Don't insert any RR headers when handling loose-routed requests.
I tried to explain all this stuff in a number of articles. Here are the
links:
https://kb.smartvox.co.uk/opensips/opensips-as-ms-teams-sbc/
https://kb.smartvox.co.uk/opensips/nat-contact-and-via-fixing-in-sip-part-4/
https://kb.smartvox.co.uk/opensips/contact-and-record-route-headers-explaine
d/
John Quick
Smartvox Limited
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Re: [OpenSIPS-Users] TLS to UDP, record route
hello John
i have found what was causing the issue. is was topology hiding when ACK
was received by opensips.
thank you for all your help and time :)
br
miha
John Quick je 5/18/2021 ob 11:15 AM napisal:
Miha
Altering the text in the Record-Route headers with subst() function is not
the correct approach.
I believe the problem is that you are not inserting the correct RR headers
in the first place.
If you get the RR headers right then it will also fix the problem with ACK
using the wrong protocol.
This pseudo-code snippet illustrates what is required when adding RR headers
to the initial INVITE request:
if (INVITE-from-Teams-Proxy-to-us) {
record_route_preset("IP:port", "SBC_FQDN:5061;transport=tls");
add_rr_param(";r2=on");
} else if (INVITE-from-us-to-Teams-Proxy) {
record_route_preset( "SBC_FQDN:5061;transport=tls", "IP:port");
add_rr_param(";r2=on");
} else
record_route();
Don't insert any RR headers when handling loose-routed requests.
I tried to explain all this stuff in a number of articles. Here are the
links:
https://kb.smartvox.co.uk/opensips/opensips-as-ms-teams-sbc/
https://kb.smartvox.co.uk/opensips/nat-contact-and-via-fixing-in-sip-part-4/
https://kb.smartvox.co.uk/opensips/contact-and-record-route-headers-explaine
d/
John Quick
Smartvox Limited
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Re: [OpenSIPS-Users] TLS to UDP, record route
thank you.
Anther thing, when you got ACK, how opensips route it, how it will
choose right interface?
Initial invite is send via tls to ms teams by opensips. But ACK which I
get from another SBC is routed via UDP and not TLS? Where i should look
for this issue?
thank you
miha
U Opensips:5060 -> Another_SBC(Not opensips):5060 #5
SIP/2.0 200 OK.
FROM: opensips)>;tag=3830244708-1542214291.
TO:
;tag=797d0c0c74c94df5abf29cc5ba182311.
CSEQ: 1 INVITE.
CALL-ID: 16826581-3830244708-1607573...@sbc2.test.com.
VIA: SIP/2.0/UDP Another_SBC(Not
opensips):5060;branch=z9hG4bK4c6433227c52d1863c051b23a170706e.
RECORD-ROUTE:
,,.
CONTACT:
.
Content-Length: 309.
SUPPORTED: timer.
CONTENT-TYPE: application/sdp.
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY.
REQUIRE: timer.
SESSION-EXPIRES: 3600;refresher=uac.
SERVER: Microsoft.PSTNHub.SIPProxy v.2021.5.5.15 i.EUWE.3.
U Another_SBC(Not opensips):5060 -> Opensips:5060 #6
ACK
sip:52.114.75.24:5061;x-i=d1df8c6c-2e6c-44b8-b670-de5b94e4b93e;x-c=e6e92b7127765ed3843b370d59046d54/s/1/6cff2c6f65024a6b86a5c5f3794202a5
SIP/2.0.
Max-Forwards: 68.
Route: .
Route: .
Route: .
To:
;tag=797d0c0c74c94df5abf29cc5ba182311.
From: opensips)>;tag=3830244708-1542214291.
Call-ID: 16826581-3830244708-1607573...@sbc2.test.com.
CSeq: 1 ACK.
Via: SIP/2.0/UDP Another_SBC(Not
opensips):5060;branch=z9hG4bK58b13e244cbca78925b45e9bcc69d3a2.
Contact: .
Content-Length: 0.
.
U Opensips:5060 -> 52.114.75.24:5061 #7
ACK
sip:52.114.75.24:5061;x-i=d1df8c6c-2e6c-44b8-b670-de5b94e4b93e;x-c=e6e92b7127765ed3843b370d59046d54/s/1/6cff2c6f65024a6b86a5c5f3794202a5
SIP/2.0.
Max-Forwards: 67.
Route: .
Route: .
Route: .
To:
;tag=797d0c0c74c94df5abf29cc5ba182311.
From: opensips)>;tag=3830244708-1542214291.
Call-ID: 16826581-3830244708-1607573...@sbc2.test.com.
CSeq: 1 ACK.
Via: SIP/2.0/UDP Opensips:5060;branch=z9hG4bKed06.0b1dc487.2.
Via: SIP/2.0/UDP Another_SBC(Not
opensips):5060;branch=z9hG4bK58b13e244cbca78925b45e9bcc69d3a2.
Contact: .
Content-Length: 0.
Callum Guy je 5/17/2021 ob 10:15 AM napisal:
subst_uri only works on the request uri, try again with subst()!
On Mon, 17 May 2021 at 08:58, Miha via Users <mailto:users@lists.opensips.org>> wrote:
Hello
i need a little help how to chnage RR in responses to UDP GW
(requestes goes via TLS to MS teams).
So in reply i have like this:
RECORD-ROUTE:
,.
But i should have like this: RECORD-ROUTE:
,.
I tried to do it like: subst_uri('/mtsbc.test.com
<http://mtsbc.test.com>:5061;transport=tls/xxx.xxx.xxx.:5060/i');
but it does not match.
thank you
miha
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[OpenSIPS-Users] TLS to UDP, record route
Hello
i need a little help how to chnage RR in responses to UDP GW (requestes
goes via TLS to MS teams).
So in reply i have like this:
RECORD-ROUTE:
,.
But i should have like this: RECORD-ROUTE:
,.
I tried to do it like:
subst_uri('/mtsbc.test.com:5061;transport=tls/xxx.xxx.xxx.:5060/i');
but it does not match.
thank you
miha
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Re: [OpenSIPS-Users] MS team issue
hello
i tried to put this in address table:
"*.pstnhub.microsoft.com" but it does not work.
On Tue, 11 May 2021 09:13:37 +0200
Johan De Clercq wrote:
> the pstnhub's can change their ip address.
> Therefore you need to use the fqdn.
>
> Op ma 10 mei 2021 om 21:33 schreef Miha via Users
> >:
>
> > found an issue. It was missing ip in addresses. Is
> there
> > any easier way to put all servers from Ms to addresses,
> > maybe just domain with "*."?
> >
> >
> > thank you
> >
> >
> > On Mon, 10 May 2021 19:23:20 +0200
> > Miha via Users wrote:
> > > Hello
> > >
> > > it seems for me that this works now. I only do not
> know
> > > why
> > > after 200 ok, opensips sends OPTIONS also to it self,
> > > which
> > > is quite wird.
> > >
> > > pasting logs.
> > >
> > > y 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:tm:print_request_uri:
> sip:sip.pstnhub.microsoft.com
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:tm:run_local_route: building sip_msg from buffer
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_msg:
> > > SIP Request:
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_msg:
> > > method:
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_msg:
> > > uri:
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_msg:
> > > version:
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: flags=
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_via_param: found param type 232,
>
> > > =
> > > ; state=16
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_via:
> > > end of header reached, state=5
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: via found,
> flags=
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: this is the first via
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:_parse_to:
> > > end of header reached, state=9
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:_parse_to:
> > > display={}, ruri={sip:sip.pstnhub.microsoft.com}
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:get_hdr_field: [31];
> > > uri=[sip:sip.pstnhub.microsoft.com]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:get_hdr_field: to body
> > > [sip:sip.pstnhub.microsoft.com#015#012
> > <http://sip.pstnhub.microsoft.com#015%23012>]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:get_hdr_field: cseq : <14>
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:get_hdr_field: content_length=0
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:get_hdr_field: found end of header
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: flags=
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: flags=78
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: flags=
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: flags=
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_to_param:
> > > tag=a665d66adab06c7308a33b8567de92d6-7c10
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:_parse_to:
> > > end of header reached, state=29
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:_parse_to:
> > > display={}, ruri={sip:prober@localhost}
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:w_sip_capture: src_ip:
> [xxx.xxx.xxx.xxx]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:w_sip_capture: dst_ip:
> [xxx.xxx.xxx.xxx]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:w_sip_capture: dst_port: [5061]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:w_sip_capture: src_port: [5061]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:w_sip_capture: DONE
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:db_sync_store: storing info...
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:db_mysql:db_mysql_do_prepared_query:
> > > conn=0x7f60225e6108 (tail=140050870197640)
> > > MC=
Re: [OpenSIPS-Users] MS team issue
found an issue. It was missing ip in addresses. Is there
any easier way to put all servers from Ms to addresses,
maybe just domain with "*."?
thank you
On Mon, 10 May 2021 19:23:20 +0200
Miha via Users wrote:
> Hello
>
> it seems for me that this works now. I only do not know
> why
> after 200 ok, opensips sends OPTIONS also to it self,
> which
> is quite wird.
>
> pasting logs.
>
> y 10 19:20:26 mtsbc opensips[7582]:
> DBG:tm:print_request_uri: sip:sip.pstnhub.microsoft.com
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:tm:run_local_route: building sip_msg from buffer
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:parse_msg:
> SIP Request:
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:parse_msg:
> method:
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:parse_msg:
> uri:
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:parse_msg:
> version:
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: flags=
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_via_param: found param type 232,
> =
> ; state=16
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:parse_via:
> end of header reached, state=5
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: via found, flags=
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: this is the first via
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:_parse_to:
> end of header reached, state=9
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:_parse_to:
> display={}, ruri={sip:sip.pstnhub.microsoft.com}
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:get_hdr_field: [31];
> uri=[sip:sip.pstnhub.microsoft.com]
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:get_hdr_field: to body
> [sip:sip.pstnhub.microsoft.com#015#012]
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:get_hdr_field: cseq : <14>
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:get_hdr_field: content_length=0
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:get_hdr_field: found end of header
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: flags=
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: flags=78
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: flags=
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: flags=
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_to_param:
> tag=a665d66adab06c7308a33b8567de92d6-7c10
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:_parse_to:
> end of header reached, state=29
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:_parse_to:
> display={}, ruri={sip:prober@localhost}
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:w_sip_capture: src_ip: [xxx.xxx.xxx.xxx]
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:w_sip_capture: dst_ip: [xxx.xxx.xxx.xxx]
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:w_sip_capture: dst_port: [5061]
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:w_sip_capture: src_port: [5061]
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:w_sip_capture: DONE
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:db_sync_store: storing info...
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:db_mysql:db_mysql_do_prepared_query:
> conn=0x7f60225e6108 (tail=140050870197640)
> MC=0x7f60225e6218
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:db_mysql:db_mysql_do_prepared_query: new
> query=|insert
> into sip_capture
>
(date,micro_ts,method,reply_reason,ruri,ruri_user,from_user,from_tag,to_user,to_tag,pid_user,contact_user,auth_user,callid,callid_aleg,via_1,via_1_branch,cseq,reason,content_type,auth,user_agent,source_ip,source_port,destination_ip,destination_port,contact_ip,contact_port,originator_ip,originator_port,proto,family,rtp_stat,type,node,correlation_id,from_domain,to_domain,ruri_domain,msg,custom_field1,custom_field2,custom_field3
> ) values
>
(?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?)|
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:db_mysql:re_init_statement: query is sip_capture
>
(date,micro_ts,method,reply_reason,ruri,ruri_user,from_user,from_tag,to_user,to_tag,pid_user,contact_user,auth_user,callid,callid_aleg,via_1,via_1_branch,cseq,reason,content_type,auth,user_agent,source_ip,source_port,destination_ip,destination_port,contact_ip,contact_port,originator_ip,originator_port,proto,family,rtp_stat,type,node,correlation_id,from_domain,to_domain,ruri_domain,msg,custom_field1,custom_field2,custom_field3
> ) values
>
(?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?)>,
> ptr=(nil)
> May 10 19:
Re: [OpenSIPS-Users] MS team issue
te#015#012FROM:
;tag=dd7d4e13-28ca-405e-9108-d83d4ea81f40#015#012TO:
;tag=bd3a.8217c8258ec89ff4f5427ff311954104#015#012CSEQ:
1 OPTIONS#015#012CALL-ID:
bf155f68-8a4c-4f18-b5f5-1340da0d618d#015#012VIA:
SIP/2.0/TLS
52.114.75.24:5061;branch=z9hG4bK9fe9fb55#015#012Server:
OpenSIPS (3.1.1 (x86_64/linux))#015#012Content-Length:
0#015#012#015#012
May 10 19:20:26 mtsbc opensips[7575]:
DBG:core:destroy_avp_list: destroying list (nil)
On Mon, 10 May 2021 11:45:32 -0300
Carlos Eduardo wrote:
> Thank you Nick.
>
> I've read these docs lots of times and didn't pay
> attention on it.
>
>
> Em seg., 10 de mai. de 2021 às 11:44, Nick Altmann
>
> escreveu:
>
> > Yes. You can use avp for this.
> >
>
https://opensips.org/docs/modules/3.1.x/tls_mgm.html#param_client_sip_domain_avp
> >
> > --
> > Nick
> >
> > пн, 10 мая 2021 г. в 16:09, Carlos Eduardo
> :
> >
> >> Hey all,
> >>
> >> About using the right certificate, is it possible to
> ensure opensips is
> >> going to use the right one when multiple are set in
> tls_mgm?
> >>
> >> Em seg., 10 de mai. de 2021 às 04:41, Răzvan Crainea
>
> >> escreveu:
> >>
> >>> Hi, Miha!
> >>>
> >>> According to your logs, opensips is 100% sending the
> OPTIONS through
> >>> tls, but I am not sure it is using the right
> certificate.
> >>> You can try to setup sip trace and see the
> communication between
> >>> opensips and MSTeams.
> >>>
> >>> Best regards,
> >>>
> >>> Răzvan Crainea
> >>> OpenSIPS Core Developer
> >>> http://www.opensips-solutions.com
> >>>
> >>> On 5/10/21 9:54 AM, Miha via Users wrote:
> >>> > Hello
> >>> >
> >>> > I have used letsenrypt for generating certs for
> Opensips.
> >>> >
> >>> > Regarding configuration i have fallowed your
> configuration steps on
> >>> > OpenSips blog.
> >>> >
> >>> > socket=udp:xxx.xxx.xxx.xxx:5060 # CUSTOMIZE ME
> >>> > socket=tls:xxx.xxx.xxx.xxx:5061
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > ### Proto TLS
> >>> > loadmodule "proto_tls.so"
> >>> > modparam("proto_tls", "tls_handshake_timeout", 300)
> >>> > TLS module
> >>> > loadmodule "tls_mgm.so"
> >>> > #modparam("tls_mgm", "db_url",
> "mysql://root:@localhost/opensips")
> >>> > modparam("tls_mgm", "client_sip_domain_avp",
> "mtsbcs.test.com")
> >>> > modparam("tls_mgm", "server_domain", "mt")
> >>> > #modparam("tls_mgm", "match_ip_address",
> "[mt]xxx.xxx.xxx.xxx:5061")
> >>> > #modparam("tls_mgm", "match_sip_domain",
> "[mt]mtsbcs.test.com")
> >>> > modparam("tls_mgm", "certificate",
> >>> >
> "[mt]/etc/letsencrypt/live/mtsbcs.test.com/cert.pem")
> >>> > modparam("tls_mgm", "private_key",
> >>> >
> "[mt]/etc/letsencrypt/live/mtsbcs.test.com/privkey.pem")
> >>> > modparam("tls_mgm", "ca_list",
> >>> "[mt]/etc/ssl/certs/ca-certificates.crt")
> >>> > modparam("tls_mgm", "ca_dir",
> "[mt]/etc/ssl/certs/")
> >>> > modparam("tls_mgm","verify_cert", "[mt]1")
> >>> > modparam("tls_mgm","require_cert", "[mt]1")
> >>> > modparam("tls_mgm","tls_method", "[mt]TLSv1_2")
> >>> > modparam("proto_tls", "tls_max_msg_chunks", 8)
> >>> > #modparam("tls_mgm", "tls_handshake_timeout", 300)
> >>> >
> >>> > if(is_method("OPTIONS") &&
> is_domain_local("$rd") &&
> >>> > check_source_address(0)) {
> >>> > xlog("L_INFO", "[MS TEAMS] OPTIONS
> In");
> >>> > send_reply(200, "OK");
> >>> > exit;
> >>> > }
> >>> >
> >>> >
> >>> > local_route {
> >>> >$var(dst) = "pstnhub.microsoft.com"
[OpenSIPS-Users] MS team issue
f May 10 08:53:10 mtsbc opensips[1020]: DBG:tm:run_local_route: Change in local route -> rebuilding buffer May 10 08:53:10 mtsbc opensips[1020]: DBG:core:parse_headers: flags=2000 May 10 08:53:10 mtsbc opensips[1020]: DBG:core:parse_headers: flags= May 10 08:53:10 mtsbc opensips[1020]: DBG:core:extract_ftc_hdrs: flags = 15 May 10 08:53:10 mtsbc opensips[1020]: DBG:core:extract_ftc_hdrs: hdr 2 extracted as May 10 08:53:10 mtsbc opensips[1020]: DBG:core:extract_ftc_hdrs: hdr 1 extracted as ;tag=a665d66adab06c7308a33b8567de92d6-f627#015#012> May 10 08:53:10 mtsbc opensips[1020]: DBG:core:extract_ftc_hdrs: hdr 8 extracted as May 10 08:53:10 mtsbc opensips[1020]: DBG:proto_tls:proto_tls_send: no open tcp connection found, opening new one May 10 08:53:10 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: getsockopt: snd is initially 16384 May 10 08:53:10 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: using snd buffer of 416 kb May 10 08:53:10 mtsbc opensips[1020]: DBG:core:init_sock_keepalive: TCP keepalive enabled on socket 5 May 10 08:53:10 mtsbc opensips[1020]: DBG:core:print_ip: tcpconn_new: new tcp connection to: 52.114.75.24 May 10 08:53:10 mtsbc opensips[1020]: DBG:core:tcpconn_new: on port 5061, proto 3 May 10 08:53:10 mtsbc opensips[1020]: DBG:proto_tls:tls_conn_init: Creating a whole new ssl connection May 10 08:53:10 mtsbc opensips[1020]: DBG:core:tcpconn_destroy: destroying connection 0x7f45d7e08078, flags 0018 May 10 08:53:10 mtsbc opensips[1020]: DBG:tm:insert_timer_unsafe: [0]: 0x7f45d7e066b0 (1625) May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:timer_routine: timer routine:0,tl=0x7f45d7e066b0 next=(nil), timeout=1625 May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:final_response_handler: Cancel sent out, sending 408 (0x7f45d7e06460) May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:t_should_relay_response: T_code=0, new_code=408 May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:t_pick_branch: picked branch 0, code 408 (prio=800) May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=408, flags=0 May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:t_should_relay_response: trying DNS-based failover May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:do_dns_failover: new destination available May 10 08:53:15 mtsbc opensips[1020]: DBG:core:parse_headers: flags=2000 May 10 08:53:15 mtsbc opensips[1020]: DBG:core:build_req_buf_from_sip_req: id added: <;i=0>, rcv proto=3 May 10 08:53:15 mtsbc opensips[1020]: DBG:core:parse_headers: flags= May 10 08:53:15 mtsbc opensips[1020]: DBG:proto_tls:proto_tls_send: no open tcp connection found, opening new one May 10 08:53:15 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: getsockopt: snd is initially 16384 May 10 08:53:15 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: using snd buffer of 416 kb May 10 08:53:15 mtsbc opensips[1020]: DBG:core:init_sock_keepalive: TCP keepalive enabled on socket 5 May 10 08:53:15 mtsbc opensips[1020]: DBG:core:print_ip: tcpconn_new: new tcp connection to: 52.114.132.46 May 10 08:53:15 mtsbc opensips[1020]: DBG:core:tcpconn_new: on port 5061, proto 3 May 10 08:53:15 mtsbc opensips[1020]: DBG:proto_tls:tls_conn_init: Creating a whole new ssl connection May 10 08:53:15 mtsbc opensips[1020]: DBG:core:tcpconn_destroy: destroying connection 0x7f45d7e08078, flags 0018 May 10 08:53:15 mtsbc opensips[1020]: DBG:proto_tls:proto_tls_send: no open tcp connection found, opening new one May 10 08:53:15 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: getsockopt: snd is initially 16384 May 10 08:53:15 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: using snd buffer of 416 kb May 10 08:53:15 mtsbc opensips[1020]: DBG:core:init_sock_keepalive: TCP keepalive enabled on socket 5 May 10 08:53:15 mtsbc opensips[1020]: DBG:core:print_ip: tcpconn_new: new tcp connection to: 52.114.14.70 May 10 08:53:15 mtsbc opensips[1020]: DBG:core:tcpconn_new: on port 5061, proto 3 May 10 08:53:15 mtsbc opensips[1020]: DBG:proto_tls:tls_conn_init: Creating a whole new ssl connection May 10 08:53:15 mtsbc opensips[1020]: DBG:core:tcpconn_destroy: destroying connection 0x7f45d7e08078, flags 0018 May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:local_reply: branch=0, save=0, winner=0 May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:local_reply: local transaction completed May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:run_trans_callbacks: trans=0x7f45d7e06460, callback type 256, id 0 entered May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:insert_timer_unsafe: [2]: 0x7f45d7e064e0 (1630) May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:final_response_handler: done Thank you miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] utimer task already scheduled
Hi
I have falow opensips configuration from your blog regarding MsTeams.
Version that I am using is:
(opensips-cli): mi version
{
"Server": "OpenSIPS (3.1.1 (x86_64/linux))"
}
ay 7 13:15:06 mtsbc opensips[1966]: WARNING:core:utimer_ticker: utimer
task already scheduled 100 ms ago (now 155250 ms), delaying
execution
May 7 13:15:06 mtsbc opensips[1966]: WARNING:core:utimer_ticker: utimer
task already scheduled 200 ms ago (now 155350 ms), delaying
execution
May 7 13:15:06 mtsbc opensips[1966]: WARNING:core:utimer_ticker: utimer
task already scheduled 290 ms ago (now 155440 ms), delaying
execution
What could cause this behaviour? On opensips for now nothing is running,
this starts from the beginning when opensips start.
thank you
miha
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Re: [OpenSIPS-Users] opensips-cli issue with DB creation
hello i changed config to like this and now it works: [default] #database_modules: acc clusterer dialog dialplan dispatcher domain rtpproxy usrloc database_modules: ALL log_level: WARNING #prompt_name: opensips-cli #prompt_intro: Welcome to OpenSIPS #prompt_emptyline_repeat_cmd: False #history_file: ~/.opensips-cli.history #history_file_size: 1000 #output_type: pretty-print communication_type: fifo fifo_file: /tmp/opensips_fifo #[mysql] database_path: /tmp/opensips-3.1/scripts database_name: opensips database_admin_url: mysql://root:xxx@localhost database_url: mysql://opensips:opensipsrw@localhost database_schema_path: /tmp/opensips-3.1/scripts Miha via Users je 5/3/2021 ob 12:00 PM napisal: Pasting also log: database create DEBUG: running command 'create' '[]' DEBUG: db_name: 'opensips' Password for admin MySQL user (root): DEBUG: read password: 'xxx' DEBUG: admin DB URL: 'mysql://root:xxx@localhost' DEBUG: connecting to mysql://root:xxx@localhost DEBUG: check database URL 'mysql://root:xxx@localhost/opensips' DEBUG: DB does not exist DEBUG: Create Database 'opensips' for dialect 'mysql' ... DEBUG: success DEBUG: DB URL: 'mysql://opensips:opensipsrw@localhost' DEBUG: connecting to mysql://opensips:opensipsrw@localhost/opensips INFO: creating access user for opensips ... /usr/local/lib/python3.7/dist-packages/SQLAlchemy-1.3.3-py3.7-linux-x86_64.egg/sqlalchemy/engine/default.py:552: Warning: (3163, "Authorization ID 'opensips'@'%' already exists.") cursor.execute(statement, parameters) INFO: created user 'opensips' INFO: set password 'ow' for 'opensips' (MySQL) INFO: granted access to user 'opensips' on DB 'opensips' INFO: flushed privileges DEBUG: connecting to mysql://opensips:opensipsrw@localhost/opensips ERROR: failed to connect to DB as opensips, please provide or fix the 'database_url' Miha via Users je 5/3/2021 ob 10:50 AM napisal: Hello I have config for opensips-cli like this in /etc/opensips-cli.cfg database_modules: ALL log_level: WARNING prompt_name: opensips-cli prompt_intro: Welcome to OpenSIPS at SECUREVOIP prompt_emptyline_repeat_cmd: False history_file: ~/.opensips-cli.history history_file_size: 1000 output_type: pretty-print communication_type: fifo fifo_file: /tmp/opensips_fifo database_path: /tmp/opensips-3.1/scripts database_name: opensips database_admin_url: mysql://root:@localhost:3306/opensips database_url: mysql://opensips:opensipsrw@localhost:3306/opensips When i run in opensips-cli shell "database create" i get: ERROR: failed to connect to DB as opensips, please provide or fix the 'database_url' I can see data DB opensips was created and also opensips user was created. thank you miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips-cli issue with DB creation
Pasting also log: database create DEBUG: running command 'create' '[]' DEBUG: db_name: 'opensips' Password for admin MySQL user (root): DEBUG: read password: 'xxx' DEBUG: admin DB URL: 'mysql://root:xxx@localhost' DEBUG: connecting to mysql://root:xxx@localhost DEBUG: check database URL 'mysql://root:xxx@localhost/opensips' DEBUG: DB does not exist DEBUG: Create Database 'opensips' for dialect 'mysql' ... DEBUG: success DEBUG: DB URL: 'mysql://opensips:opensipsrw@localhost' DEBUG: connecting to mysql://opensips:opensipsrw@localhost/opensips INFO: creating access user for opensips ... /usr/local/lib/python3.7/dist-packages/SQLAlchemy-1.3.3-py3.7-linux-x86_64.egg/sqlalchemy/engine/default.py:552: Warning: (3163, "Authorization ID 'opensips'@'%' already exists.") cursor.execute(statement, parameters) INFO: created user 'opensips' INFO: set password 'ow' for 'opensips' (MySQL) INFO: granted access to user 'opensips' on DB 'opensips' INFO: flushed privileges DEBUG: connecting to mysql://opensips:opensipsrw@localhost/opensips ERROR: failed to connect to DB as opensips, please provide or fix the 'database_url' Miha via Users je 5/3/2021 ob 10:50 AM napisal: Hello I have config for opensips-cli like this in /etc/opensips-cli.cfg database_modules: ALL log_level: WARNING prompt_name: opensips-cli prompt_intro: Welcome to OpenSIPS at SECUREVOIP prompt_emptyline_repeat_cmd: False history_file: ~/.opensips-cli.history history_file_size: 1000 output_type: pretty-print communication_type: fifo fifo_file: /tmp/opensips_fifo database_path: /tmp/opensips-3.1/scripts database_name: opensips database_admin_url: mysql://root:@localhost:3306/opensips database_url: mysql://opensips:opensipsrw@localhost:3306/opensips When i run in opensips-cli shell "database create" i get: ERROR: failed to connect to DB as opensips, please provide or fix the 'database_url' I can see data DB opensips was created and also opensips user was created. thank you miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips-cli issue with DB creation
Hello I have config for opensips-cli like this in /etc/opensips-cli.cfg database_modules: ALL log_level: WARNING prompt_name: opensips-cli prompt_intro: Welcome to OpenSIPS at SECUREVOIP prompt_emptyline_repeat_cmd: False history_file: ~/.opensips-cli.history history_file_size: 1000 output_type: pretty-print communication_type: fifo fifo_file: /tmp/opensips_fifo database_path: /tmp/opensips-3.1/scripts database_name: opensips database_admin_url: mysql://root:@localhost:3306/opensips database_url: mysql://opensips:opensipsrw@localhost:3306/opensips When i run in opensips-cli shell "database create" i get: ERROR: failed to connect to DB as opensips, please provide or fix the 'database_url' I can see data DB opensips was created and also opensips user was created. thank you miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] how can i combine signaling and RTP from rtpproxy
Hello due to debugging i would like to combine cap from opensips and also cap from rtpproxy (they are on different servers) so that I can check if RTP is missing for certain call. Can you help me with solving this issue :) thank you miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Load testing
Hi What is best tool for load testing that can generate also RTP? Tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] migration to 2.4
Hello we are running still on 2.1. Due to some other things I would like first to migrate to version 2.4. I went over documentation for version migration from 2.1 to 2.2 and from 2.2 to 2.3 and from 2.3. to 2.4. What I would like to know is what exactly is wrong in my config in where i should be looking for. the main issue is that I do not see this in logs. Log level is 4 (i tried aslo with 7 and other leves.) This are logs: Is there any other way to find this issue? 6 21:26:24 debian opensips[6423]: NOTICE:core:main: version: opensips 2.4.7 (x86_64/linux) May 6 21:26:24 debian opensips[6423]: INFO:core:main: using 32 Mb of shared memory May 6 21:26:24 debian opensips[6423]: INFO:core:main: using 2 Mb of private process memory May 6 21:26:24 debian opensips[6423]: INFO:core:init_reactor_size: reactor size 1024 (using up to 0.03Mb of memory per process) May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:tm:mod_init: TM - initializing... May 6 21:26:24 debian opensips[6423]: INFO:sl:mod_init: Initializing StateLess engine May 6 21:26:24 debian opensips[6423]: NOTICE:signaling:mod_init: initializing module ... May 6 21:26:24 debian opensips[6423]: INFO:rr:mod_init: rr - initializing May 6 21:26:24 debian opensips[6423]: INFO:maxfwd:mod_init: initializing... May 6 21:26:24 debian opensips[6423]: INFO:sipmsgops:mod_init: initializing... May 6 21:26:24 debian opensips[6423]: INFO:usrloc:ul_init_locks: locks array size 512 May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:registrar:mod_init: initializing... May 6 21:26:24 debian opensips[6423]: INFO:acc:mod_init: initializing... May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: Registered event May 6 21:26:24 debian opensips[6423]: INFO:core:mod_init: initializing UDP-plain protocol May 6 21:26:24 debian opensips[6423]: INFO:core:probe_max_sock_buff: using rcv buffer of 416 kb May 6 21:26:24 debian opensips: INFO:core:daemonize: pre-daemon process exiting with 0 May 7 09:15:33 debian opensips: NOTICE:core:main: Exiting May 7 09:15:57 debian opensips: NOTICE:core:main: Exiting May 7 09:16:56 debian opensips: NOTICE:core:main: Exiting May 7 09:17:23 debian opensips: NOTICE:core:main: Exiting thank you miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips + rtpproxy issue
Hello Maxim
rtpproxy -V
2.2.alpha.aa26b45
it did not. Now rtpproxy is running as sing process, could
it be lunched like opensips (multiple instances)?
Is there any other thing that can be done?
thank you
miha
On Wed, 6 May 2020 07:22:07 -0700
Maxim Sobolev wrote:
> Oh, sorry, looks like the proper way to check that is via
> using -V option.
>
> [ssp-root@macmini2 /home/ssp]$ rtpproxy -V
> 2.2.alpha.3794729c-dirty
>
> Let us know if that upgrade helped. Thanks!
>
> -Max
>
> On Wed, May 6, 2020 at 2:09 AM Miha
> wrote:
>
> > Hello Maxim
> >
> > I removed package via apt and then install it via
> source/git.
> >
> > But from what I see version is still the same:
> > How would I know for sure that the version is something
> like 2.1?
> >
> > ./rtpproxy -version
> > Basic version: 20040107
> > Extension 20040107: Basic RTP proxy functionality
> > Extension 20050322: Support for multiple RTP streams
> and MOH
> > Extension 20060704: Support for extra parameter in the
> V command
> > Extension 20071116: Support for RTP re-packetization
> > Extension 20071218: Support for forking (copying) RTP
> stream
> > Extension 20080403: Support for RTP statistics querying
> > Extension 20081102: Support for setting codecs in the
> update/lookup command
> > Extension 20081224: Support for session timeout
> notifications
> > Extension 20090810: Support for automatic bridging
> > Extension 20140323: Support for tracking/reporting load
> > Extension 20140617: Support for anchoring session
> connect time
> > Extension 20141004: Support for extendable performance
> counters
> > Extension 20150330: Support for allocating a new port
> ("Un"/"Ln" commands)
> > Extension 20150420: Support for SEQ tracking and new
> rtpa_ counters; Q
> > command extended
> > Extension 20150617: Support for the wildcard
> %%CC_SELF%% as a disconnect
> > notify target
> > Extension 20191015: Support for the && sub-command
> specifier
> > Extension 20200226: Support for the N command to stop
> recording
> >
> >
> >
> > Maxim Sobolev je 5/6/2020 ob 2:22 AM napisal:
> >
> > Hi Miha, sorry to hear about your issues. In order to
> troubleshoot it
> > further could you please also provide rtpproxy package
> version as reported
> > by the system package manager (apt, rpm etc) if the
> software has been
> > installed via that channel or branch name if it's been
> built from sources?
> > Unfortunately version reporting of the --version
> command has been bit
> > crippled until recently, already improved in latest
> master and 2.1 I
> > believe.
> >
> > In general performance under virtual environment has
> not been terrific,
> > due to some design choices made early in our work.
> Hovewer I believe it
> > should be much better in 2.0 and 2.1 vs. 1.x series.
> Some of it is
> > inherently due to VM scheduling jitter, some is because
> we are unwilling to
> > put it into unsafe domain (i.e. kernel mode). As a rule
> of thumb, you might
> > expect 3-5x drop in max pps until jitter becomes an
> issue as compared to
> > running on comparable bare metal. Spinning multiple
> instances might help to
> > mitigate some of it though, but it also depends on
> hypervisor version and
> > even particular CPU generation.
> >
> > -Max
> >
> > On Tue., May 5, 2020, 6:10 a.m. Miha via Users,
>
> > wrote:
> >
> >> Hello
> >>
> >> we have virtualized opensips and rtpproxy running on
> the same server
> >> which is virtualized in vmware infrastructure. Servers
> are not old, also
> >> traffic is not so big (cca 50 simultaneous calls).
> when there is a peak cca
> >> 80 simultaneous calls RTP starts to break.
> >>
> >> is there any special setting/flag to be set, so that I
> can optimze this?
> >> load on VM is very low.
> >>
> >> rtpproxy -version
> >> Basic version: 20040107
> >>
> >> Opensips is 2.1
> >>
> >>
> >> thank you for help.
> >> Miha
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >
> >
>
> --
> Maksym Sobolyev
> Sippy Software, Inc.
> Internet Telephony (VoIP) Experts
> Tel (Canada): +1-778-783-0474
> Tel (Toll-Free): +1-855-747-7779
> Fax: +1-866-857-6942
> Web: http://www.sippysoft.com
> MSN: sa...@sippysoft.com
> Skype: SippySoft
___
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Re: [OpenSIPS-Users] Opensips + rtpproxy issue
Hello Maxim
I removed package via apt and then install it via source/git.
But from what I see version is still the same:
How would I know for sure that the version is something like 2.1?
./rtpproxy -version
Basic version: 20040107
Extension 20040107: Basic RTP proxy functionality
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension 20080403: Support for RTP statistics querying
Extension 20081102: Support for setting codecs in the update/lookup command
Extension 20081224: Support for session timeout notifications
Extension 20090810: Support for automatic bridging
Extension 20140323: Support for tracking/reporting load
Extension 20140617: Support for anchoring session connect time
Extension 20141004: Support for extendable performance counters
Extension 20150330: Support for allocating a new port ("Un"/"Ln" commands)
Extension 20150420: Support for SEQ tracking and new rtpa_ counters; Q
command extended
Extension 20150617: Support for the wildcard %%CC_SELF%% as a disconnect
notify target
Extension 20191015: Support for the && sub-command specifier
Extension 20200226: Support for the N command to stop recording
Maxim Sobolev je 5/6/2020 ob 2:22 AM napisal:
Hi Miha, sorry to hear about your issues. In order to troubleshoot it
further could you please also provide rtpproxy package version as
reported by the system package manager (apt, rpm etc) if the software
has been installed via that channel or branch name if it's been built
from sources? Unfortunately version reporting of the --version command
has been bit crippled until recently, already improved in latest
master and 2.1 I believe.
In general performance under virtual environment has not been
terrific, due to some design choices made early in our work. Hovewer I
believe it should be much better in 2.0 and 2.1 vs. 1.x series. Some
of it is inherently due to VM scheduling jitter, some is because we
are unwilling to put it into unsafe domain (i.e. kernel mode). As a
rule of thumb, you might expect 3-5x drop in max pps until jitter
becomes an issue as compared to running on comparable bare metal.
Spinning multiple instances might help to mitigate some of it though,
but it also depends on hypervisor version and even particular CPU
generation.
-Max
On Tue., May 5, 2020, 6:10 a.m. Miha via Users,
mailto:users@lists.opensips.org>> wrote:
Hello
we have virtualized opensips and rtpproxy running on the same
server which is virtualized in vmware infrastructure. Servers are
not old, also traffic is not so big (cca 50 simultaneous calls).
when there is a peak cca 80 simultaneous calls RTP starts to break.
is there any special setting/flag to be set, so that I can optimze
this? load on VM is very low.
rtpproxy -version
Basic version: 20040107
Opensips is 2.1
thank you for help.
Miha
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Re: [OpenSIPS-Users] Opensips + rtpproxy issue
Hello Danies • what do you mean by enough max open do files? I do no linit or set anything • I traced with tshark and i can see issue with A and B leg Thank you for help! Br Miha Miha On 5 May 2020, 16:07 +0200, Daniel Zanutti , wrote: > No special configuration, we just set IP's and ports. > > Since CPU is not your problem, I believe you have some kind of bandwidth > limitation in your network. > > I suggest you confirm: > 1) You have enough max open files in your rtpproxy process -> /proc/PID/limits > 2) Where the bottleneck is: CPU, IO or bandwidth. You can record some packets > in wireshark inside RTPPROXY machine and confirm audio is distorted before > and after rtpproxy. > > Regards > > > > On Tue, May 5, 2020 at 10:35 AM Miha wrote: > > > Hi, > > > > > > no CPU usage is around 1% to 5%, basically nothing. > > > In sound there is big distortion it is impossibly to > > > comunicate with each other. > > > > > > We have two cors deticated to it. Do you have any special > > > thing set on it? > > > > > > tnx > > > miha > > > > > > On Tue, 5 May 2020 10:27:22 -0300 > > > Daniel Zanutti wrote: > > > > Hi Miha > > > > > > > > Could you explaining how does it break? We use it in > > > > virtual machines and > > > > our safe limit is around 500 simultaneous calls, on > > > > dedicated single core > > > > VPS. Does CPU usage reach 100%? > > > > > > > > > > > > > > > > On Tue, May 5, 2020 at 10:11 AM Miha via Users > > > > > > > > wrote: > > > > > > > > > Hello > > > > > > > > > > we have virtualized opensips and rtpproxy running on > > > > the same server which > > > > > is virtualized in vmware infrastructure. Servers are > > > > not old, also traffic > > > > > is not so big (cca 50 simultaneous calls). when there > > > > is a peak cca 80 > > > > > simultaneous calls RTP starts to break. > > > > > > > > > > is there any special setting/flag to be set, so that I > > > > can optimze this? > > > > > load on VM is very low. > > > > > > > > > > rtpproxy -version > > > > > Basic version: 20040107 > > > > > > > > > > Opensips is 2.1 > > > > > > > > > > > > > > > thank you for help. > > > > > Miha > > > > > ___ > > > > > Users mailing list > > > > > Users@lists.opensips.org > > > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips + rtpproxy issue
Hi, no CPU usage is around 1% to 5%, basically nothing. In sound there is big distortion it is impossibly to comunicate with each other. We have two cors deticated to it. Do you have any special thing set on it? tnx miha On Tue, 5 May 2020 10:27:22 -0300 Daniel Zanutti wrote: > Hi Miha > > Could you explaining how does it break? We use it in > virtual machines and > our safe limit is around 500 simultaneous calls, on > dedicated single core > VPS. Does CPU usage reach 100%? > > > > On Tue, May 5, 2020 at 10:11 AM Miha via Users > > wrote: > > > Hello > > > > we have virtualized opensips and rtpproxy running on > the same server which > > is virtualized in vmware infrastructure. Servers are > not old, also traffic > > is not so big (cca 50 simultaneous calls). when there > is a peak cca 80 > > simultaneous calls RTP starts to break. > > > > is there any special setting/flag to be set, so that I > can optimze this? > > load on VM is very low. > > > > rtpproxy -version > > Basic version: 20040107 > > > > Opensips is 2.1 > > > > > > thank you for help. > > Miha > > ___ > > Users mailing list > > Users@lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips + rtpproxy issue
Hello we have virtualized opensips and rtpproxy running on the same server which is virtualized in vmware infrastructure. Servers are not old, also traffic is not so big (cca 50 simultaneous calls). when there is a peak cca 80 simultaneous calls RTP starts to break. is there any special setting/flag to be set, so that I can optimze this? load on VM is very low. rtpproxy -version Basic version: 20040107 Opensips is 2.1 thank you for help. Miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Alias domain / dns srv
Hello Bogdan not mixing, just maybe wrong discribing :) That is what i did. Thank you for your help and explenation! br miha Bogdan-Andrei Iancu je 4/15/2020 ob 7:31 PM napisal: Hi Miha, You are mixing the SIP domains with the SIP server location. The SIP domains have nothing to do with SRV, while for SIP server location you can use it. The idea is to set as SIP user u...@sip.test.com (and 'sip.test.com' is the SIP domain all the time). If the domain does not support SRV, it will do an A lookup on sip.test.com, and you can point it , as IP, to proxy1.test.com. If the domain supports SRVyou know the drill . But in both cases the SIP domain in SIP messages will be 'sip.test.com' Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/14/20 11:57 AM, Miha via Users wrote: Hello we have dns srv record for failover. In dns srv we have two record. So, one version of our devices does not support dns srv records. Is it possible to register device directly to one A record which is wirtten in DNS SRV record and then use ALIAS in opensips to right domain? DNS SRV. sip.test.com (proxy1.test.com, proxy2.test.com) Devices that do not support will register to proxy1.test.com (opensips will have alias which will point to sip.test.com)? thank you miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Alias domain / dns srv
Hello we have dns srv record for failover. In dns srv we have two record. So, one version of our devices does not support dns srv records. Is it possible to register device directly to one A record which is wirtten in DNS SRV record and then use ALIAS in opensips to right domain? DNS SRV. sip.test.com (proxy1.test.com, proxy2.test.com) Devices that do not support will register to proxy1.test.com (opensips will have alias which will point to sip.test.com)? thank you miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] different ip in from as initial invite
Liviu thank you very much for your quick answer! I will try then to stick as it is as it is the right way. If there will be no other choise that maybe i try this. thank you again! miha Liviu Chircu je 1/28/2020 ob 1:52 PM napisal: On 28.01.2020 14:43, Miha via Users wrote: Costumer is saying that he expects from like it was send in 200ok (not in inital invite, tag and CALLERID stays always the same) and we should confirm with ACK that has from same as in 200 ok from them. Hi miha, That is complete nonsense, RFC 3261 is on your side, section § 8.2.6.2: The From field of the response MUST equal the From header field of the request. The Call-ID header field of the response MUST equal the Call-ID header field of the request. The CSeq header field of the response MUST equal the CSeq field of the request. The Via header field values in the response MUST equal the Via header field values in the request and MUST maintain the same ordering. However, if you are really keen to help them out... maybe you could store their 200 OK From header in a $dlg_val, then fix the ACK's From header to use this val. But how will you handle the From header for other sequential requests? And if these requests are initiated by the downstream side, you will have to change the To instead of the From, as the UAC must swap them! We are basically opening Pandora's Box by doing down this route. It's not impossible to get right, but it will take some work. Regards, -- Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 www.opensips.org/events OpenSIPS Bootcamp, Miami, March 2020 www.opensips.org/training ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] different ip in from as initial invite
Hi first call flow. 1. Invite with FROM 12345@1.2.3.4 2. 200 ok with FROM 1.2.3.4@1.2.3.5 3. ACK, FROM is like in initial invite 12345@1.2.3.4 Costumer is saying that he expects from like it was send in 200ok (not in inital invite, tag and CALLERID stays always the same) and we should confirm with ACK that has from same as in 200 ok from them. Problem is that in my case opensips adds FROM from initial invite (ip 1.2.3.4, it should be 1.2.3.5). IN onreply route a can not use uac_change_from. Can this be change and how? thank you miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Remove doubled connection information in SDP
Hello I get two connection infomrmation in SDP (doubled), which are the same. How to remove one? ps.: i am using rtpproxy. thank you. Miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips/rtpporxy and early media
Hi Razvan
just one thing to be clear as I do not see anything in my debugging mode.
So after i get beck 180 ringing, must call:
rtpproxy_answer("rfo",,"2");
and
rtpproxy_stream2uas("/tmp/wav.wav", "-1"); (I did conversion and files
are like wav.wav.3, .0, .8)
Must I do something else? This is done on on_replay route. I guess I
must change to 183 session in progress?
thank you for help!
Miha
On 8/19/2019 8:57 PM, Miha via Users wrote:
Hi, Răzvan!
thank you, so i was thinking right :)
br
miha
On Mon, 19 Aug 2019 17:28:07 +0300
Răzvan Crainea wrote:
Hi, Miha!
You first need to convert the wav file you want to stream
to a RTP payload, one for each codec you support. To do
that, you can use the makeann tool that rtpproxy
provides[1].
Once you have those files (named file.3 for GSM, file.0
for PCMU. file.8 for PCMA), you need to call the
rtpproxy_stream2uac("file"). This will automatically do
the codec selection and choose the right file.
[1] https://github.com/sippy/rtpproxy/tree/master/makeann
Best regards,
Răzvan
On 8/19/19 4:07 PM, Miha via Users wrote:
Hello guys
first time doing this, normally I use freeswitch... Se
in combination with rtpproxy how to enable ringback tone.
I need to call rtpproxy_stream2() i add it as file? Or
there is some other option for this if I would like that
is played by UAS?
thank you for help!
miha
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OpenSIPS Core Developer
http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] opensips/rtpporxy and early media
Hi, Răzvan!
thank you, so i was thinking right :)
br
miha
On Mon, 19 Aug 2019 17:28:07 +0300
Răzvan Crainea wrote:
> Hi, Miha!
>
> You first need to convert the wav file you want to stream
> to a RTP payload, one for each codec you support. To do
> that, you can use the makeann tool that rtpproxy
> provides[1].
> Once you have those files (named file.3 for GSM, file.0
> for PCMU. file.8 for PCMA), you need to call the
> rtpproxy_stream2uac("file"). This will automatically do
> the codec selection and choose the right file.
>
> [1] https://github.com/sippy/rtpproxy/tree/master/makeann
>
> Best regards,
> Răzvan
>
> On 8/19/19 4:07 PM, Miha via Users wrote:
> > Hello guys
> >
> > first time doing this, normally I use freeswitch... Se
> in combination with rtpproxy how to enable ringback tone.
> I need to call rtpproxy_stream2() i add it as file? Or
> there is some other option for this if I would like that
> is played by UAS?
> >
> >
> > thank you for help!
> > miha
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> >
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
>http://www.opensips-solutions.com
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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[OpenSIPS-Users] opensips/rtpporxy and early media
Hello guys first time doing this, normally I use freeswitch... Se in combination with rtpproxy how to enable ringback tone. I need to call rtpproxy_stream2() i add it as file? Or there is some other option for this if I would like that is played by UAS? thank you for help! miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] issue with compiling opensips
hello is there on www where written which libraries are all mandatory (debian)? tnx miha On 17/11/2017 08:42, Ketan Kothari wrote: Hello Miha, Please verify below dependency installed on your server or not. libjson-c-dev libjson-c-doc libjson-c2 libjson-c2-dbg libjson0 libjson0-dev On Fri, Nov 17, 2017 at 12:58 PM, Miha <mailto:m...@softnet.si>> wrote: hey this is what i have: cat /etc/issue Debian GNU/Linux 9 \n \ apt list libjson* Listing... Done libjson-any-perl/stable 1.39-1 all libjson-c-dev/stable,now 0.12.1-1.1 amd64 [installed] libjson-c-doc/stable 0.12.1-1.1 all libjson-c3/stable,now 0.12.1-1.1 amd64 [installed,automatic] libjson-glib-1.0-0/stable,now 1.2.6-1 amd64 [installed,automatic] libjson-glib-1.0-common/stable,now 1.2.6-1 all [installed,automatic] libjson-glib-dev/stable 1.2.6-1 amd64 libjson-glib-doc/stable 1.2.6-1 all libjson-java/stable 2.4-3 all libjson-maybexs-perl/stable 1.003008-1 all libjson-multivalueordered-perl/stable 0.005-1 all libjson-perl/stable 2.90-1 all libjson-pointer-perl/stable 0.07-1 all libjson-pp-perl/stable 2.27400-1 all libjson-rpc-perl/stable 1.06-2 all libjson-simple-doc/stable 1.1.1-4 all libjson-simple-java/stable 1.1.1-4 all libjson-smart-java/stable 2.2-1 all libjson-types-perl/stable 0.05-1 all libjson-validator-perl/stable 0.92+dfsg-1 all libjson-webtoken-perl/stable 0.10-2 all libjson-xs-perl/stable 3.030-1 amd64 libjsoncpp-dev/stable,now 1.7.4-3 amd64 [installed] libjsoncpp-doc/stable 1.7.4-3 all libjsoncpp1/stable,now 1.7.4-3 amd64 [installed,automatic] libjsonm-ocaml/stable 0.9.1-2 amd64 libjsonm-ocaml-dev/stable 0.9.1-2 amd64 libjsonm-ocaml-doc/stable 0.9.1-2 all libjsonp-java/stable 1.0.4-1 all libjsonp-java-doc/stable 1.0.4-1 all libjsonpath-java/stable 2.0.0-3 all libjsonrpccpp-client0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-client0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-common0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-common0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-dev/stable 0.7.0-1+b2 amd64 libjsonrpccpp-server0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-server0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-stub0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-stub0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-tools/stable 0.7.0-1+b2 amd64 On 16/11/2017 16:47, Răzvan Crainea wrote: What version of libjson are you using, and what OS are you running? Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com <http://www.opensips-solutions.com> On 11/16/2017 01:49 PM, Miha wrote: hello Razvan no, i downloaded it from opensips on git, i did this yesterday (2.3) br miha On 16/11/2017 11:23, Răzvan Crainea wrote: Are you using an older version of OpenSIPS? This should have been fixed in all supported versions. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com <http://www.opensips-solutions.com> On 11/15/2017 01:52 PM, Miha wrote: Hello which deb pack should I install regarding this error: https://pastebin.com/c4RHMbcT I installed this package " libjson-c-dev" but it is not ok. tnx miha ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] issue with compiling opensips
hey this is what i have: cat /etc/issue Debian GNU/Linux 9 \n \ apt list libjson* Listing... Done libjson-any-perl/stable 1.39-1 all libjson-c-dev/stable,now 0.12.1-1.1 amd64 [installed] libjson-c-doc/stable 0.12.1-1.1 all libjson-c3/stable,now 0.12.1-1.1 amd64 [installed,automatic] libjson-glib-1.0-0/stable,now 1.2.6-1 amd64 [installed,automatic] libjson-glib-1.0-common/stable,now 1.2.6-1 all [installed,automatic] libjson-glib-dev/stable 1.2.6-1 amd64 libjson-glib-doc/stable 1.2.6-1 all libjson-java/stable 2.4-3 all libjson-maybexs-perl/stable 1.003008-1 all libjson-multivalueordered-perl/stable 0.005-1 all libjson-perl/stable 2.90-1 all libjson-pointer-perl/stable 0.07-1 all libjson-pp-perl/stable 2.27400-1 all libjson-rpc-perl/stable 1.06-2 all libjson-simple-doc/stable 1.1.1-4 all libjson-simple-java/stable 1.1.1-4 all libjson-smart-java/stable 2.2-1 all libjson-types-perl/stable 0.05-1 all libjson-validator-perl/stable 0.92+dfsg-1 all libjson-webtoken-perl/stable 0.10-2 all libjson-xs-perl/stable 3.030-1 amd64 libjsoncpp-dev/stable,now 1.7.4-3 amd64 [installed] libjsoncpp-doc/stable 1.7.4-3 all libjsoncpp1/stable,now 1.7.4-3 amd64 [installed,automatic] libjsonm-ocaml/stable 0.9.1-2 amd64 libjsonm-ocaml-dev/stable 0.9.1-2 amd64 libjsonm-ocaml-doc/stable 0.9.1-2 all libjsonp-java/stable 1.0.4-1 all libjsonp-java-doc/stable 1.0.4-1 all libjsonpath-java/stable 2.0.0-3 all libjsonrpccpp-client0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-client0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-common0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-common0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-dev/stable 0.7.0-1+b2 amd64 libjsonrpccpp-server0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-server0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-stub0/stable 0.7.0-1+b2 amd64 libjsonrpccpp-stub0-dbg/stable 0.7.0-1+b2 amd64 libjsonrpccpp-tools/stable 0.7.0-1+b2 amd64 On 16/11/2017 16:47, Răzvan Crainea wrote: What version of libjson are you using, and what OS are you running? Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/16/2017 01:49 PM, Miha wrote: hello Razvan no, i downloaded it from opensips on git, i did this yesterday (2.3) br miha On 16/11/2017 11:23, Răzvan Crainea wrote: Are you using an older version of OpenSIPS? This should have been fixed in all supported versions. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/15/2017 01:52 PM, Miha wrote: Hello which deb pack should I install regarding this error: https://pastebin.com/c4RHMbcT I installed this package " libjson-c-dev" but it is not ok. tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] issue with compiling opensips
hello Razvan no, i downloaded it from opensips on git, i did this yesterday (2.3) br miha On 16/11/2017 11:23, Răzvan Crainea wrote: Are you using an older version of OpenSIPS? This should have been fixed in all supported versions. Best regards, Răzvan Crainea OpenSIPS Developer www.opensips-solutions.com On 11/15/2017 01:52 PM, Miha wrote: Hello which deb pack should I install regarding this error: https://pastebin.com/c4RHMbcT I installed this package " libjson-c-dev" but it is not ok. tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] issue with compiling opensips
Hello which deb pack should I install regarding this error: https://pastebin.com/c4RHMbcT I installed this package " libjson-c-dev" but it is not ok. tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] radius accounting, Acct-Status-Type = 0
Tnx:) working! br miha On 17/01/2017 12:30, Ionut Ionita wrote: Also if you'll check latest commit in master[0], modified radius code to throw error and not start if any attribute/ value/vendor is not found in order to not get to similar situations in the future. Backported the commit back to 2.1. [0] https://github.com/OpenSIPS/opensips/commit/75573b41c5453b495a3fa9ad1bdf2df3ee0f4c2f Regards, Ionut Ionita OpenSIPS Developer On 01/17/2017 12:06 PM, Ionut Ionita wrote: Hello again, Found a fix for your problem. You should use *dictionary.rfc2866* (freeradius has it). This dictionary has value defined for *Failed*. VALUE› Acct-Status-Type› › Start› › › 1 VALUE› Acct-Status-Type› › Stop› › › 2 VALUE› Acct-Status-Type› › Alive› › › 3 # dup VALUE› Acct-Status-Type› › Interim-Update› › 3 VALUE› Acct-Status-Type› › Accounting-On› › 7 VALUE› Acct-Status-Type› › Accounting-Off› › 8 VALUE› Acct-Status-Type› › Failed› › › 15 Regards, Ionut Ionita OpenSIPS Developer On 01/17/2017 10:33 AM, Miha wrote: ok, then i will use this value and on radius side i will catch it and set to internally to STOP if this is only soluton. The thing is when 486/busy comes, opensips sends this broken radius accouting request and radius server does not replay to it and that is why the opensips waits almost 40s to send back ACK on 486. Or is there any other solution? br miha On 17/01/2017 09:27, Ionut Ionita wrote: No, thank you it's fine now. That value was for testing purposes only, you can remove it now. OpenSIPS it's using *Failed* value for *Acct-Status-Type* which is not defined anywhere (nor in our dictionary, nor in any RFC or somewhere else). Not finding that value results in having *0 *for *Acct-Status-Type*, the value you were seeing before. Will let you know when we'll decide how we should fix this issue. Regards, Ionut Ionita OpenSIPS Developer On 01/17/2017 09:38 AM, Miha wrote: Hi Ionut do I need on bouth sides or opensips side? I can see that now i get: Acct-Status-Type = Modem-Start in radius. br miha On 16/01/2017 16:45, Ionut Ionita wrote: Hi Miha, Can you set in your radius dictionary file where the *Acct-Status-Type* values are defined VALUEAcct-Status-TypeStart› › › 1 VALUEAcct-Status-TypeStop› › › 2 VALUEAcct-Status-TypeAlive› › › 3 VALUEAcct-Status-TypeAccounting-On› › 7 VALUEAcct-Status-TypeAccounting-Off› › 8 the following line, just below the others: VALUEAcct-Status-TypeFailed4 and then check if you'll see *4 *instead of *0* for *Acct-Status-Type*? It seems that opensips it's using a value that's not in the radius dictionary, *Failed* value. Ionut Ionita OpenSIPS Developer On 01/16/2017 04:05 PM, Miha wrote: Hello how can i define that for 486/busy opensips will send Acct-Status-Type = 2 to radius server? Acct-Status-Type = 0 it not like standard thing and it should not be send :) I imported dictinary.opensips and .sip. tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] radius accounting, Acct-Status-Type = 0
ok, then i will use this value and on radius side i will catch it and set to internally to STOP if this is only soluton. The thing is when 486/busy comes, opensips sends this broken radius accouting request and radius server does not replay to it and that is why the opensips waits almost 40s to send back ACK on 486. Or is there any other solution? br miha On 17/01/2017 09:27, Ionut Ionita wrote: No, thank you it's fine now. That value was for testing purposes only, you can remove it now. OpenSIPS it's using *Failed* value for *Acct-Status-Type* which is not defined anywhere (nor in our dictionary, nor in any RFC or somewhere else). Not finding that value results in having *0 *for *Acct-Status-Type*, the value you were seeing before. Will let you know when we'll decide how we should fix this issue. Regards, Ionut Ionita OpenSIPS Developer On 01/17/2017 09:38 AM, Miha wrote: Hi Ionut do I need on bouth sides or opensips side? I can see that now i get: Acct-Status-Type = Modem-Start in radius. br miha On 16/01/2017 16:45, Ionut Ionita wrote: Hi Miha, Can you set in your radius dictionary file where the *Acct-Status-Type* values are defined VALUEAcct-Status-TypeStart› › › 1 VALUEAcct-Status-TypeStop› › › 2 VALUEAcct-Status-TypeAlive› › › 3 VALUEAcct-Status-TypeAccounting-On› › 7 VALUEAcct-Status-TypeAccounting-Off› › 8 the following line, just below the others: VALUEAcct-Status-TypeFailed4 and then check if you'll see *4 *instead of *0* for *Acct-Status-Type*? It seems that opensips it's using a value that's not in the radius dictionary, *Failed* value. Ionut Ionita OpenSIPS Developer On 01/16/2017 04:05 PM, Miha wrote: Hello how can i define that for 486/busy opensips will send Acct-Status-Type = 2 to radius server? Acct-Status-Type = 0 it not like standard thing and it should not be send :) I imported dictinary.opensips and .sip. tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] radius accounting, Acct-Status-Type = 0
Hi Ionut do I need on bouth sides or opensips side? I can see that now i get: Acct-Status-Type = Modem-Start in radius. br miha On 16/01/2017 16:45, Ionut Ionita wrote: Hi Miha, Can you set in your radius dictionary file where the *Acct-Status-Type* values are defined VALUEAcct-Status-TypeStart› › › 1 VALUEAcct-Status-TypeStop› › › 2 VALUEAcct-Status-TypeAlive› › › 3 VALUEAcct-Status-TypeAccounting-On› › 7 VALUEAcct-Status-TypeAccounting-Off› › 8 the following line, just below the others: VALUEAcct-Status-TypeFailed4 and then check if you'll see *4 *instead of *0* for *Acct-Status-Type*? It seems that opensips it's using a value that's not in the radius dictionary, *Failed* value. Ionut Ionita OpenSIPS Developer On 01/16/2017 04:05 PM, Miha wrote: Hello how can i define that for 486/busy opensips will send Acct-Status-Type = 2 to radius server? Acct-Status-Type = 0 it not like standard thing and it should not be send :) I imported dictinary.opensips and .sip. tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] radius accounting, Acct-Status-Type = 0
Hello how can i define that for 486/busy opensips will send Acct-Status-Type = 2 to radius server? Acct-Status-Type = 0 it not like standard thing and it should not be send :) I imported dictinary.opensips and .sip. tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips crash
Hey Denis https://www.opensips.org/Documentation/TroubleShooting-Crash it will create core. miha On 23/12/2016 10:35, Denis wrote: One question. If in the destination directory of the core file will be located another "core" file, what will be? Would "old core" file be replaced by a new one, or Opensips makes another core file with a fresh data? Thank you. -- С уважением, Денис. Best regards, Denis 23.12.2016, 12:23, "Răzvan Crainea" : Please update to the latest 2.2.2. If you still have problems, try to make sure opensips can generate a corefile[1]. [1] http://www.opensips.org/Documentation/TroubleShooting-Crash Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com <http://www.opensips-solutions.com/> On 12/23/2016 11:16 AM, Denis wrote: Hello! Server:: OpenSIPS (2.2.1 (x86_64/linux)) Today i had a crash of Opensips. Everything that i could collect is here https://yadi.sk/i/dyNnXpBr34YJQ3 Unfortunately, i could not find any fresh core file, despite of the fact that Opensips starts with -w /opensipscore option. In opensipscore i found only core file at 29 Nov. Thank you for any help. -- С уважением, Денис. Best regards, Denis ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users , ___ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] nat issue
Hello Bogdan i think it is no need to do that if this client is broken. You already doing so much good with opensips ;) Tnx so much with all explanation and all you work! Miha On 21/11/2016 11:13, Bogdan-Andrei Iancu wrote: Hi Miha, According the SIP grammar, that parameter is perfectly legitimate. The client is broken as it is not able to cope with it (in the worst case, to simply ignore it). There is no out of the box way to disable it, but I may provide you a patch for that - just to see if that fixes your problem. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2016 19:54, Miha wrote: Hello bogdan I guess, but it looks like so. Is it possible to remove it? tnx miha On 18/11/2016 15:39, Bogdan-Andrei Iancu wrote: I guess your UAC freezes when receiving back in the 200 OK REGISTER the "received" hdr param in Contact ?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2016 16:33, Bogdan-Andrei Iancu wrote: HI Miha, Sorry, but I'm not able to follow the case you mentioned with Innovaphone PBX - maybe you can post (to see the differences) the sent and returned contact hdrs in the REGISTER request + reply for the 2 cases (OpenSIPS and Innovaphone PBX). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2016 11:20, Miha wrote: I do not know if this is the case. But from what I can see what I register it on some Innovaphone PBX, innovaphone sends back in contact (200 ok) just ip of IPBX and also when INVITE is send in contact there is URI of PBX and only and it works. i tried this but did not have any luck. br miha On 18/11/2016 09:48, Bogdan-Andrei Iancu wrote: Hi Miha, You mean the UAC does not like the multi-URI Contact header in the 200 OK If so, that UAC is really broken as 1) breaks the SIP syntax (which allows it) and 2) breaks the the SIP Registration as per RFC3261. What about the second contact (the one already existing in usrloc when this registration comes) ? can it be discarded ? If YES, you can try passing the "c1f" flags to save() : http://www.opensips.org/html/docs/modules/2.2.x/registrar.html#id294033 That will make OpenSIPS to accept only 1 contact per AOR/user and any new contact will override the existing one. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2016 10:15, Miha wrote: Hi Bogdan I did few more test. This contact bothers UAC. Is there anything i can do in this case in OpenSIPS so that it will only reply with one URI in contact? Contact:;expires=1518 ;received="sip:84.41.125.21:5060",; expires=180. tnx so much! MIha On 17/11/2016 12:11, Bogdan-Andrei Iancu wrote: Hi Miha, yes, that is parallel forking (you may have more than 2 contacts only). Are you sure your DB was sync'ed? OpenSIPS is periodically flushing the memory cache into the location table (see the "state" of the contact (as per "ul show") if marked as DIRTY). In regards to RFC, I think you quote the wrong section (maybe about callings?) - for REGISTERs, any number of URIs are allowed AFAIK. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 17.11.2016 12:35, Miha wrote: Bodan so this is dual forking...? So if you have one account and you have two phones on it and first will try to register, 200 ok will will have contact of both phones? In location table I can see only one registration for this user but for "opensipsctl ul show" it shows me two contacts, which is strange? (When i do trace only one invite is send) and UAC replay with Busy all the time due to two contacts (this what i have been told). Ok, but if you look at rfc there is only one URI allowed in contact if I understand this right? The Contact header field MUST be present and contain exactly one SIP or SIPS URI in any request that can result in the establishment of a dialog Please correct me if I am wrong. tnx so much! Miha On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote: Hi Miha, OpenSIPS returns in the 200 OK for a REGISTER all the valid registrations for that user (for all the devices the user may have). I guess your user has 2 registrations, so the 200 OK will report back both of them. You can check via "opensipsctl ul show" Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 17.11.2016 12:13, Miha wrote: Hello Bogdan i changed this and it works in all cases, only in one I noticed today this (Opensips reply only in this case with two URI on contact): UAC:5060 ->OpenSIPS:5060 REGISTER sip:opsp.test.net:5060 SIP/2.0. Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6. Max-Forwards: 70. From: 042335040 ;tag=1f62205074. To: 042335040 . Cal
Re: [OpenSIPS-Users] nat issue
Hello bogdan I guess, but it looks like so. Is it possible to remove it? tnx miha On 18/11/2016 15:39, Bogdan-Andrei Iancu wrote: I guess your UAC freezes when receiving back in the 200 OK REGISTER the "received" hdr param in Contact ?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2016 16:33, Bogdan-Andrei Iancu wrote: HI Miha, Sorry, but I'm not able to follow the case you mentioned with Innovaphone PBX - maybe you can post (to see the differences) the sent and returned contact hdrs in the REGISTER request + reply for the 2 cases (OpenSIPS and Innovaphone PBX). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2016 11:20, Miha wrote: I do not know if this is the case. But from what I can see what I register it on some Innovaphone PBX, innovaphone sends back in contact (200 ok) just ip of IPBX and also when INVITE is send in contact there is URI of PBX and only and it works. i tried this but did not have any luck. br miha On 18/11/2016 09:48, Bogdan-Andrei Iancu wrote: Hi Miha, You mean the UAC does not like the multi-URI Contact header in the 200 OK If so, that UAC is really broken as 1) breaks the SIP syntax (which allows it) and 2) breaks the the SIP Registration as per RFC3261. What about the second contact (the one already existing in usrloc when this registration comes) ? can it be discarded ? If YES, you can try passing the "c1f" flags to save() : http://www.opensips.org/html/docs/modules/2.2.x/registrar.html#id294033 That will make OpenSIPS to accept only 1 contact per AOR/user and any new contact will override the existing one. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 18.11.2016 10:15, Miha wrote: Hi Bogdan I did few more test. This contact bothers UAC. Is there anything i can do in this case in OpenSIPS so that it will only reply with one URI in contact? Contact:;expires=1518 ;received="sip:84.41.125.21:5060",; expires=180. tnx so much! MIha On 17/11/2016 12:11, Bogdan-Andrei Iancu wrote: Hi Miha, yes, that is parallel forking (you may have more than 2 contacts only). Are you sure your DB was sync'ed? OpenSIPS is periodically flushing the memory cache into the location table (see the "state" of the contact (as per "ul show") if marked as DIRTY). In regards to RFC, I think you quote the wrong section (maybe about callings?) - for REGISTERs, any number of URIs are allowed AFAIK. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 17.11.2016 12:35, Miha wrote: Bodan so this is dual forking...? So if you have one account and you have two phones on it and first will try to register, 200 ok will will have contact of both phones? In location table I can see only one registration for this user but for "opensipsctl ul show" it shows me two contacts, which is strange? (When i do trace only one invite is send) and UAC replay with Busy all the time due to two contacts (this what i have been told). Ok, but if you look at rfc there is only one URI allowed in contact if I understand this right? The Contact header field MUST be present and contain exactly one SIP or SIPS URI in any request that can result in the establishment of a dialog Please correct me if I am wrong. tnx so much! Miha On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote: Hi Miha, OpenSIPS returns in the 200 OK for a REGISTER all the valid registrations for that user (for all the devices the user may have). I guess your user has 2 registrations, so the 200 OK will report back both of them. You can check via "opensipsctl ul show" Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 17.11.2016 12:13, Miha wrote: Hello Bogdan i changed this and it works in all cases, only in one I noticed today this (Opensips reply only in this case with two URI on contact): UAC:5060 ->OpenSIPS:5060 REGISTER sip:opsp.test.net:5060 SIP/2.0. Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6. Max-Forwards: 70. From: 042335040 ;tag=1f62205074. To: 042335040 . Call-ID: 61c67f739bef5a2e. CSeq: 1804289391 REGISTER. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, INFO. Authorization: Digest username="99942335040",realm="opsp.test.net",nonce="582d810c58b d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res ponse="bc0c757c17f9b0976af35ec633dd83ca". Contact: 042335040 ;ex pires=3600. Privacy: none. Supported: path. User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2. Content-Length: 0. UOpenSIPS:5060 -> UAC:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022 5bd7495297c6. F
Re: [OpenSIPS-Users] nat issue
I do not know if this is the case. But from what I can see what I
register it on some Innovaphone PBX, innovaphone sends back in contact
(200 ok) just ip of IPBX and also when INVITE is send in contact there
is URI of PBX and only and it works.
i tried this but did not have any luck.
br
miha
On 18/11/2016 09:48, Bogdan-Andrei Iancu wrote:
Hi Miha,
You mean the UAC does not like the multi-URI Contact header in the 200
OK If so, that UAC is really broken as 1) breaks the SIP syntax
(which allows it) and 2) breaks the the SIP Registration as per RFC3261.
What about the second contact (the one already existing in usrloc when
this registration comes) ? can it be discarded ? If YES, you can try
passing the "c1f" flags to save() :
http://www.opensips.org/html/docs/modules/2.2.x/registrar.html#id294033
That will make OpenSIPS to accept only 1 contact per AOR/user and any
new contact will override the existing one.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 18.11.2016 10:15, Miha wrote:
Hi Bogdan
I did few more test. This contact bothers UAC. Is there anything i
can do in this case in OpenSIPS so that it will only reply with one
URI in contact?
Contact:;expires=1518
;received="sip:84.41.125.21:5060",;
expires=180.
tnx so much!
MIha
On 17/11/2016 12:11, Bogdan-Andrei Iancu wrote:
Hi Miha,
yes, that is parallel forking (you may have more than 2 contacts only).
Are you sure your DB was sync'ed? OpenSIPS is periodically flushing
the memory cache into the location table (see the "state" of the
contact (as per "ul show") if marked as DIRTY).
In regards to RFC, I think you quote the wrong section (maybe about
callings?) - for REGISTERs, any number of URIs are allowed AFAIK.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:35, Miha wrote:
Bodan
so this is dual forking...?
So if you have one account and you have two phones on it and first
will try to register, 200 ok will will have contact of both phones?
In location table I can see only one registration for this user but
for "opensipsctl ul show" it shows me two contacts, which is
strange? (When i do trace only one invite is send) and UAC replay
with Busy all the time due to two contacts (this what i have been
told).
Ok, but if you look at rfc there is only one URI allowed in contact
if I understand this right?
The Contact header field MUST be present and contain exactly one SIP
or SIPS URI in any request that can result in the establishment of a
dialog
Please correct me if I am wrong. tnx so much! Miha
On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote:
Hi Miha,
OpenSIPS returns in the 200 OK for a REGISTER all the valid
registrations for that user (for all the devices the user may have).
I guess your user has 2 registrations, so the 200 OK will report
back both of them. You can check via "opensipsctl ul show"
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:13, Miha wrote:
Hello Bogdan
i changed this and it works in all cases, only in one I noticed
today this (Opensips reply only in this case with two URI on
contact):
UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d810c58b
d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res
ponse="bc0c757c17f9b0976af35ec633dd83ca".
Contact: 042335040 ;ex
pires=3600.
Privacy: none.
Supported: path.
User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2.
Content-Length: 0.
UOpenSIPS:5060 -> UAC:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022
5bd7495297c6.
From: 042335040 ;tag=1f62205074.
To: 042335040 ;tag=0c7ff67d927afc274
b272138ce65100a.ac4d.
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
WWW-Authenticate: Digest realm="opsp.test.net",
nonce="582d81135a88b92d0287a7460acce0a84e5d2a200b33", stale=true.
Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)).
Content-Length: 0.
U UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKb5f2bbbf80e346f48.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289392 REGISTER.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d8
Re: [OpenSIPS-Users] nat issue
Hi Bogdan I did few more test. This contact bothers UAC. Is there anything i can do in this case in OpenSIPS so that it will only reply with one URI in contact? Contact:;expires=1518 ;received="sip:84.41.125.21:5060",; expires=180. tnx so much! MIha On 17/11/2016 12:11, Bogdan-Andrei Iancu wrote: Hi Miha, yes, that is parallel forking (you may have more than 2 contacts only). Are you sure your DB was sync'ed? OpenSIPS is periodically flushing the memory cache into the location table (see the "state" of the contact (as per "ul show") if marked as DIRTY). In regards to RFC, I think you quote the wrong section (maybe about callings?) - for REGISTERs, any number of URIs are allowed AFAIK. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 17.11.2016 12:35, Miha wrote: Bodan so this is dual forking...? So if you have one account and you have two phones on it and first will try to register, 200 ok will will have contact of both phones? In location table I can see only one registration for this user but for "opensipsctl ul show" it shows me two contacts, which is strange? (When i do trace only one invite is send) and UAC replay with Busy all the time due to two contacts (this what i have been told). Ok, but if you look at rfc there is only one URI allowed in contact if I understand this right? The Contact header field MUST be present and contain exactly one SIP or SIPS URI in any request that can result in the establishment of a dialog Please correct me if I am wrong. tnx so much! Miha On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote: Hi Miha, OpenSIPS returns in the 200 OK for a REGISTER all the valid registrations for that user (for all the devices the user may have). I guess your user has 2 registrations, so the 200 OK will report back both of them. You can check via "opensipsctl ul show" Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 17.11.2016 12:13, Miha wrote: Hello Bogdan i changed this and it works in all cases, only in one I noticed today this (Opensips reply only in this case with two URI on contact): UAC:5060 ->OpenSIPS:5060 REGISTER sip:opsp.test.net:5060 SIP/2.0. Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6. Max-Forwards: 70. From: 042335040 ;tag=1f62205074. To: 042335040 . Call-ID: 61c67f739bef5a2e. CSeq: 1804289391 REGISTER. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, INFO. Authorization: Digest username="99942335040",realm="opsp.test.net",nonce="582d810c58b d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res ponse="bc0c757c17f9b0976af35ec633dd83ca". Contact: 042335040 ;ex pires=3600. Privacy: none. Supported: path. User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2. Content-Length: 0. UOpenSIPS:5060 -> UAC:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022 5bd7495297c6. From: 042335040 ;tag=1f62205074. To: 042335040 ;tag=0c7ff67d927afc274 b272138ce65100a.ac4d. Call-ID: 61c67f739bef5a2e. CSeq: 1804289391 REGISTER. WWW-Authenticate: Digest realm="opsp.test.net", nonce="582d81135a88b92d0287a7460acce0a84e5d2a200b33", stale=true. Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)). Content-Length: 0. U UAC:5060 ->OpenSIPS:5060 REGISTER sip:opsp.test.net:5060 SIP/2.0. Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKb5f2bbbf80e346f48. Max-Forwards: 70. From: 042335040 ;tag=1f62205074. To: 042335040 . Call-ID: 61c67f739bef5a2e. CSeq: 1804289392 REGISTER. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, INFO. Authorization: Digest username="99942335040",realm="opsp.test.net",nonce="582d81135a8 8b92d0287a7460acce0a84e5d2a200b33",uri="sip:opsp.test.net:5060",res ponse="9ce3622addeedf74622a23697e6f3728". Contact: 042335040 ;ex pires=3600. Privacy: none. Supported: path. User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2. Content-Length: 0. . UOpenSIPS:5060 -> UAC:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKb5f2b bbf80e346f48. From: 042335040 ;tag=1f62205074. To: 042335040 ;tag=766e4f757c55b3450 c9992a50fb64799-9163. Call-ID: 61c67f739bef5a2e. CSeq: 1804289392 REGISTER. Contact: ;expires=3600 ;received="sip:UAC:5060", ;expires=119. Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)). Content-Length: 0. Do you see where could be an issue? tnx miha On 16/11/2016 08:11, Miha wrote: Hello Bogdan yes this was the case... thank you! br miha On 15/11/2016 18:35, Bogdan-Andrei Iancu wrote: Hi Miha, When you handle REGISTER requests (from behind NAT) most probably you use fix_nated_contact() instead of fix_nated_register(). Regards, Bogdan-Andrei Iancu
Re: [OpenSIPS-Users] nat issue
Hello Bogdan how would I know that is marked as DIRTY? how this will look like? tnx miha On 17/11/2016 12:11, Bogdan-Andrei Iancu wrote: Hi Miha, yes, that is parallel forking (you may have more than 2 contacts only). Are you sure your DB was sync'ed? OpenSIPS is periodically flushing the memory cache into the location table (see the "state" of the contact (as per "ul show") if marked as DIRTY). In regards to RFC, I think you quote the wrong section (maybe about callings?) - for REGISTERs, any number of URIs are allowed AFAIK. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 17.11.2016 12:35, Miha wrote: Bodan so this is dual forking...? So if you have one account and you have two phones on it and first will try to register, 200 ok will will have contact of both phones? In location table I can see only one registration for this user but for "opensipsctl ul show" it shows me two contacts, which is strange? (When i do trace only one invite is send) and UAC replay with Busy all the time due to two contacts (this what i have been told). Ok, but if you look at rfc there is only one URI allowed in contact if I understand this right? The Contact header field MUST be present and contain exactly one SIP or SIPS URI in any request that can result in the establishment of a dialog Please correct me if I am wrong. tnx so much! Miha On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote: Hi Miha, OpenSIPS returns in the 200 OK for a REGISTER all the valid registrations for that user (for all the devices the user may have). I guess your user has 2 registrations, so the 200 OK will report back both of them. You can check via "opensipsctl ul show" Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 17.11.2016 12:13, Miha wrote: Hello Bogdan i changed this and it works in all cases, only in one I noticed today this (Opensips reply only in this case with two URI on contact): UAC:5060 ->OpenSIPS:5060 REGISTER sip:opsp.test.net:5060 SIP/2.0. Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6. Max-Forwards: 70. From: 042335040 ;tag=1f62205074. To: 042335040 . Call-ID: 61c67f739bef5a2e. CSeq: 1804289391 REGISTER. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, INFO. Authorization: Digest username="99942335040",realm="opsp.test.net",nonce="582d810c58b d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res ponse="bc0c757c17f9b0976af35ec633dd83ca". Contact: 042335040 ;ex pires=3600. Privacy: none. Supported: path. User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2. Content-Length: 0. UOpenSIPS:5060 -> UAC:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022 5bd7495297c6. From: 042335040 ;tag=1f62205074. To: 042335040 ;tag=0c7ff67d927afc274 b272138ce65100a.ac4d. Call-ID: 61c67f739bef5a2e. CSeq: 1804289391 REGISTER. WWW-Authenticate: Digest realm="opsp.test.net", nonce="582d81135a88b92d0287a7460acce0a84e5d2a200b33", stale=true. Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)). Content-Length: 0. U UAC:5060 ->OpenSIPS:5060 REGISTER sip:opsp.test.net:5060 SIP/2.0. Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKb5f2bbbf80e346f48. Max-Forwards: 70. From: 042335040 ;tag=1f62205074. To: 042335040 . Call-ID: 61c67f739bef5a2e. CSeq: 1804289392 REGISTER. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, INFO. Authorization: Digest username="99942335040",realm="opsp.test.net",nonce="582d81135a8 8b92d0287a7460acce0a84e5d2a200b33",uri="sip:opsp.test.net:5060",res ponse="9ce3622addeedf74622a23697e6f3728". Contact: 042335040 ;ex pires=3600. Privacy: none. Supported: path. User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2. Content-Length: 0. . UOpenSIPS:5060 -> UAC:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKb5f2b bbf80e346f48. From: 042335040 ;tag=1f62205074. To: 042335040 ;tag=766e4f757c55b3450 c9992a50fb64799-9163. Call-ID: 61c67f739bef5a2e. CSeq: 1804289392 REGISTER. Contact: ;expires=3600 ;received="sip:UAC:5060", ;expires=119. Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)). Content-Length: 0. Do you see where could be an issue? tnx miha On 16/11/2016 08:11, Miha wrote: Hello Bogdan yes this was the case... thank you! br miha On 15/11/2016 18:35, Bogdan-Andrei Iancu wrote: Hi Miha, When you handle REGISTER requests (from behind NAT) most probably you use fix_nated_contact() instead of fix_nated_register(). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 15.11.2016 09:11, Miha wrote: Hello i need one info. I have one phone behind NAT and it is registe
Re: [OpenSIPS-Users] nat issue
Bodan so this is dual forking...? So if you have one account and you have two phones on it and first will try to register, 200 ok will will have contact of both phones? In location table I can see only one registration for this user but for "opensipsctl ul show" it shows me two contacts, which is strange? (When i do trace only one invite is send) and UAC replay with Busy all the time due to two contacts (this what i have been told). Ok, but if you look at rfc there is only one URI allowed in contact if I understand this right? The Contact header field MUST be present and contain exactly one SIP or SIPS URI in any request that can result in the establishment of a dialog Please correct me if I am wrong. tnx so much! Miha On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote: Hi Miha, OpenSIPS returns in the 200 OK for a REGISTER all the valid registrations for that user (for all the devices the user may have). I guess your user has 2 registrations, so the 200 OK will report back both of them. You can check via "opensipsctl ul show" Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 17.11.2016 12:13, Miha wrote: Hello Bogdan i changed this and it works in all cases, only in one I noticed today this (Opensips reply only in this case with two URI on contact): UAC:5060 ->OpenSIPS:5060 REGISTER sip:opsp.test.net:5060 SIP/2.0. Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6. Max-Forwards: 70. From: 042335040 ;tag=1f62205074. To: 042335040 . Call-ID: 61c67f739bef5a2e. CSeq: 1804289391 REGISTER. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, INFO. Authorization: Digest username="99942335040",realm="opsp.test.net",nonce="582d810c58b d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res ponse="bc0c757c17f9b0976af35ec633dd83ca". Contact: 042335040 ;ex pires=3600. Privacy: none. Supported: path. User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2. Content-Length: 0. UOpenSIPS:5060 -> UAC:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022 5bd7495297c6. From: 042335040 ;tag=1f62205074. To: 042335040 ;tag=0c7ff67d927afc274 b272138ce65100a.ac4d. Call-ID: 61c67f739bef5a2e. CSeq: 1804289391 REGISTER. WWW-Authenticate: Digest realm="opsp.test.net", nonce="582d81135a88b92d0287a7460acce0a84e5d2a200b33", stale=true. Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)). Content-Length: 0. U UAC:5060 ->OpenSIPS:5060 REGISTER sip:opsp.test.net:5060 SIP/2.0. Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKb5f2bbbf80e346f48. Max-Forwards: 70. From: 042335040 ;tag=1f62205074. To: 042335040 . Call-ID: 61c67f739bef5a2e. CSeq: 1804289392 REGISTER. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, INFO. Authorization: Digest username="99942335040",realm="opsp.test.net",nonce="582d81135a8 8b92d0287a7460acce0a84e5d2a200b33",uri="sip:opsp.test.net:5060",res ponse="9ce3622addeedf74622a23697e6f3728". Contact: 042335040 ;ex pires=3600. Privacy: none. Supported: path. User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2. Content-Length: 0. . UOpenSIPS:5060 -> UAC:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKb5f2b bbf80e346f48. From: 042335040 ;tag=1f62205074. To: 042335040 ;tag=766e4f757c55b3450 c9992a50fb64799-9163. Call-ID: 61c67f739bef5a2e. CSeq: 1804289392 REGISTER. Contact: ;expires=3600 ;received="sip:UAC:5060", ;expires=119. Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)). Content-Length: 0. Do you see where could be an issue? tnx miha On 16/11/2016 08:11, Miha wrote: Hello Bogdan yes this was the case... thank you! br miha On 15/11/2016 18:35, Bogdan-Andrei Iancu wrote: Hi Miha, When you handle REGISTER requests (from behind NAT) most probably you use fix_nated_contact() instead of fix_nated_register(). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 15.11.2016 09:11, Miha wrote: Hello i need one info. I have one phone behind NAT and it is registered on OpenSIPS. IN register packet, which is send to OpenSIPS I can see contact: "sip:11181600519@192.168.0.101:5060;transport=UDP" and let says that the public ip for this device is xxx.xxx.xxx.xxx. When opensips sends INVITE it send to right public ip and right port (source ip and source port generated by router). The issue is this: Invite is like: "sip:11181600...@xxx.xxx.xxx.xxx:5060;transport=UDP" and this request is then fw to this UAC behind router. The UAC replays to this INVITE with 404 Not found as it is waiting to receive the same URI which was written in contact (the userpart is ok, put the ip is public, not private and this is the issue).From what I can s
Re: [OpenSIPS-Users] nat issue
Hello Bogdan i changed this and it works in all cases, only in one I noticed today this (Opensips reply only in this case with two URI on contact): UAC:5060 ->OpenSIPS:5060 REGISTER sip:opsp.test.net:5060 SIP/2.0. Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6. Max-Forwards: 70. From: 042335040 ;tag=1f62205074. To: 042335040 . Call-ID: 61c67f739bef5a2e. CSeq: 1804289391 REGISTER. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, INFO. Authorization: Digest username="99942335040",realm="opsp.test.net",nonce="582d810c58b d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res ponse="bc0c757c17f9b0976af35ec633dd83ca". Contact: 042335040 ;ex pires=3600. Privacy: none. Supported: path. User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2. Content-Length: 0. UOpenSIPS:5060 -> UAC:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022 5bd7495297c6. From: 042335040 ;tag=1f62205074. To: 042335040 ;tag=0c7ff67d927afc274 b272138ce65100a.ac4d. Call-ID: 61c67f739bef5a2e. CSeq: 1804289391 REGISTER. WWW-Authenticate: Digest realm="opsp.test.net", nonce="582d81135a88b92d0287a7460acce0a84e5d2a200b33", stale=true. Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)). Content-Length: 0. U UAC:5060 ->OpenSIPS:5060 REGISTER sip:opsp.test.net:5060 SIP/2.0. Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKb5f2bbbf80e346f48. Max-Forwards: 70. From: 042335040 ;tag=1f62205074. To: 042335040 . Call-ID: 61c67f739bef5a2e. CSeq: 1804289392 REGISTER. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, INFO. Authorization: Digest username="99942335040",realm="opsp.test.net",nonce="582d81135a8 8b92d0287a7460acce0a84e5d2a200b33",uri="sip:opsp.test.net:5060",res ponse="9ce3622addeedf74622a23697e6f3728". Contact: 042335040 ;ex pires=3600. Privacy: none. Supported: path. User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2. Content-Length: 0. . UOpenSIPS:5060 -> UAC:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKb5f2b bbf80e346f48. From: 042335040 ;tag=1f62205074. To: 042335040 ;tag=766e4f757c55b3450 c9992a50fb64799-9163. Call-ID: 61c67f739bef5a2e. CSeq: 1804289392 REGISTER. Contact: ;expires=3600 ;received="sip:UAC:5060", ;expires=119. Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)). Content-Length: 0. Do you see where could be an issue? tnx miha On 16/11/2016 08:11, Miha wrote: Hello Bogdan yes this was the case... thank you! br miha On 15/11/2016 18:35, Bogdan-Andrei Iancu wrote: Hi Miha, When you handle REGISTER requests (from behind NAT) most probably you use fix_nated_contact() instead of fix_nated_register(). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 15.11.2016 09:11, Miha wrote: Hello i need one info. I have one phone behind NAT and it is registered on OpenSIPS. IN register packet, which is send to OpenSIPS I can see contact: "sip:11181600519@192.168.0.101:5060;transport=UDP" and let says that the public ip for this device is xxx.xxx.xxx.xxx. When opensips sends INVITE it send to right public ip and right port (source ip and source port generated by router). The issue is this: Invite is like: "sip:11181600...@xxx.xxx.xxx.xxx:5060;transport=UDP" and this request is then fw to this UAC behind router. The UAC replays to this INVITE with 404 Not found as it is waiting to receive the same URI which was written in contact (the userpart is ok, put the ip is public, not private and this is the issue).From what I can see in RFC this is the case. Till now Idid not have any issues with this, but now I found first phone which replays with 404 and from RFC point of view there should be private ip request :) . So is there anything I can do :)? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] nat issue
Hello Bogdan yes this was the case... thank you! br miha On 15/11/2016 18:35, Bogdan-Andrei Iancu wrote: Hi Miha, When you handle REGISTER requests (from behind NAT) most probably you use fix_nated_contact() instead of fix_nated_register(). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 15.11.2016 09:11, Miha wrote: Hello i need one info. I have one phone behind NAT and it is registered on OpenSIPS. IN register packet, which is send to OpenSIPS I can see contact: "sip:11181600519@192.168.0.101:5060;transport=UDP" and let says that the public ip for this device is xxx.xxx.xxx.xxx. When opensips sends INVITE it send to right public ip and right port (source ip and source port generated by router). The issue is this: Invite is like: "sip:11181600...@xxx.xxx.xxx.xxx:5060;transport=UDP" and this request is then fw to this UAC behind router. The UAC replays to this INVITE with 404 Not found as it is waiting to receive the same URI which was written in contact (the userpart is ok, put the ip is public, not private and this is the issue).From what I can see in RFC this is the case. Till now Idid not have any issues with this, but now I found first phone which replays with 404 and from RFC point of view there should be private ip request :) . So is there anything I can do :)? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] nat issue
Hello i need one info. I have one phone behind NAT and it is registered on OpenSIPS. IN register packet, which is send to OpenSIPS I can see contact: "sip:11181600519@192.168.0.101:5060;transport=UDP" and let says that the public ip for this device is xxx.xxx.xxx.xxx. When opensips sends INVITE it send to right public ip and right port (source ip and source port generated by router). The issue is this: Invite is like: "sip:11181600...@xxx.xxx.xxx.xxx:5060;transport=UDP" and this request is then fw to this UAC behind router. The UAC replays to this INVITE with 404 Not found as it is waiting to receive the same URI which was written in contact (the userpart is ok, put the ip is public, not private and this is the issue).From what I can see in RFC this is the case. Till now Idid not have any issues with this, but now I found first phone which replays with 404 and from RFC point of view there should be private ip request :) . So is there anything I can do :)? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Help with big amount of data for routing
ok tnx so much Bogdan!
br
miha
On 12/10/2016 14:39, Bogdan-Andrei Iancu wrote:
Give it a try. Just be sure you properly adjust the shared memory and
note that the loading will take a bit of a time.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 15:37, Miha wrote:
this is great! tnx :) I was thinking that but as it was so much data
i did not even want to try it :)
I will try and let you know!
br
miha
On 12/10/2016 14:33, Bogdan-Andrei Iancu wrote:
Why don't you use DR for that translation. Make a routing group
where you put all DIDs (as prefixes) in dr_rules and have the NET_ID
as attribute for the rule. And when looking it up:
do_routing("group","LC");
See :
http://www.opensips.org/html/docs/modules/2.2.x/drouting.html#id295067
DR is more memory efficient and much much faster in the lookup.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 15:30, Miha wrote:
yes. I have in redis like: KEY (DID) and VALUE (NETID), than I am
doing lookup in opensips script.
I was looking what is the most a appropriate way to do this. As
redis is quite good in this cases I choose it but the issue is
memory :(
tnx
miha
On 12/10/2016 14:25, Bogdan-Andrei Iancu wrote:
So basically you need to determine the NET_ID based on the DID
number ? this is what you do in REDIS now ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 15:09, Miha wrote:
Hi Bogdan
i missed your email, sorry...
Operator need to have number like NET_ID + DID. So routing is
based on NET_ID. I manage to get all number in redis (quite a big
amount of memory is used :) ).So first I do lookup in redis, get
NET_ID and then I am using d_routing based od NET_IDs.
br
miha
On 05/10/2016 15:17, Bogdan-Andrei Iancu wrote:
Hi Miha,
so you have 120M records (NET_ID + DID) - how do you use them
from OpenSIPS ? As I fail to understand what are the operations
you want to perform over this data.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05.10.2016 09:44, Miha wrote:
HI
the is not really opensips issue:) I need somehow to store big
amount of data for routing.
To every telephone operator I must send RURI like
Net_ID+Telephone_number (value indicates to who number belongs
to). In this country they have around 120 millions of numbers.
After i have all NET_IDs with numbers I will use drouting for
routing numbers to right operator based on NET_ID.
Here is the issue:
- I tried this with redis (lookup must be quick) but this takes
so much memory that basically redis brakes everytime in between
50 millions and 70 millions entries
- I tried with hash (hset) in redis but did not do any good
Do you have any suggestion how to deal with this, what would be
the best thing to use?
tnx
miha
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Re: [OpenSIPS-Users] Help with big amount of data for routing
this is great! tnx :) I was thinking that but as it was so much data i
did not even want to try it :)
I will try and let you know!
br
miha
On 12/10/2016 14:33, Bogdan-Andrei Iancu wrote:
Why don't you use DR for that translation. Make a routing group where
you put all DIDs (as prefixes) in dr_rules and have the NET_ID as
attribute for the rule. And when looking it up:
do_routing("group","LC");
See :
http://www.opensips.org/html/docs/modules/2.2.x/drouting.html#id295067
DR is more memory efficient and much much faster in the lookup.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 15:30, Miha wrote:
yes. I have in redis like: KEY (DID) and VALUE (NETID), than I am
doing lookup in opensips script.
I was looking what is the most a appropriate way to do this. As
redis is quite good in this cases I choose it but the issue is memory :(
tnx
miha
On 12/10/2016 14:25, Bogdan-Andrei Iancu wrote:
So basically you need to determine the NET_ID based on the DID
number ? this is what you do in REDIS now ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 15:09, Miha wrote:
Hi Bogdan
i missed your email, sorry...
Operator need to have number like NET_ID + DID. So routing is based
on NET_ID. I manage to get all number in redis (quite a big amount
of memory is used :) ).So first I do lookup in redis, get NET_ID
and then I am using d_routing based od NET_IDs.
br
miha
On 05/10/2016 15:17, Bogdan-Andrei Iancu wrote:
Hi Miha,
so you have 120M records (NET_ID + DID) - how do you use them from
OpenSIPS ? As I fail to understand what are the operations you
want to perform over this data.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05.10.2016 09:44, Miha wrote:
HI
the is not really opensips issue:) I need somehow to store big
amount of data for routing.
To every telephone operator I must send RURI like
Net_ID+Telephone_number (value indicates to who number belongs
to). In this country they have around 120 millions of numbers.
After i have all NET_IDs with numbers I will use drouting for
routing numbers to right operator based on NET_ID.
Here is the issue:
- I tried this with redis (lookup must be quick) but this takes
so much memory that basically redis brakes everytime in between
50 millions and 70 millions entries
- I tried with hash (hset) in redis but did not do any good
Do you have any suggestion how to deal with this, what would be
the best thing to use?
tnx
miha
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Re: [OpenSIPS-Users] Help with big amount of data for routing
yes. I have in redis like: KEY (DID) and VALUE (NETID), than I am doing lookup in opensips script. I was looking what is the most a appropriate way to do this. As redis is quite good in this cases I choose it but the issue is memory :( tnx miha On 12/10/2016 14:25, Bogdan-Andrei Iancu wrote: So basically you need to determine the NET_ID based on the DID number ? this is what you do in REDIS now ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.10.2016 15:09, Miha wrote: Hi Bogdan i missed your email, sorry... Operator need to have number like NET_ID + DID. So routing is based on NET_ID. I manage to get all number in redis (quite a big amount of memory is used :) ).So first I do lookup in redis, get NET_ID and then I am using d_routing based od NET_IDs. br miha On 05/10/2016 15:17, Bogdan-Andrei Iancu wrote: Hi Miha, so you have 120M records (NET_ID + DID) - how do you use them from OpenSIPS ? As I fail to understand what are the operations you want to perform over this data. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05.10.2016 09:44, Miha wrote: HI the is not really opensips issue:) I need somehow to store big amount of data for routing. To every telephone operator I must send RURI like Net_ID+Telephone_number (value indicates to who number belongs to). In this country they have around 120 millions of numbers. After i have all NET_IDs with numbers I will use drouting for routing numbers to right operator based on NET_ID. Here is the issue: - I tried this with redis (lookup must be quick) but this takes so much memory that basically redis brakes everytime in between 50 millions and 70 millions entries - I tried with hash (hset) in redis but did not do any good Do you have any suggestion how to deal with this, what would be the best thing to use? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Help with big amount of data for routing
Hi Bogdan i missed your email, sorry... Operator need to have number like NET_ID + DID. So routing is based on NET_ID. I manage to get all number in redis (quite a big amount of memory is used :) ).So first I do lookup in redis, get NET_ID and then I am using d_routing based od NET_IDs. br miha On 05/10/2016 15:17, Bogdan-Andrei Iancu wrote: Hi Miha, so you have 120M records (NET_ID + DID) - how do you use them from OpenSIPS ? As I fail to understand what are the operations you want to perform over this data. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05.10.2016 09:44, Miha wrote: HI the is not really opensips issue:) I need somehow to store big amount of data for routing. To every telephone operator I must send RURI like Net_ID+Telephone_number (value indicates to who number belongs to). In this country they have around 120 millions of numbers. After i have all NET_IDs with numbers I will use drouting for routing numbers to right operator based on NET_ID. Here is the issue: - I tried this with redis (lookup must be quick) but this takes so much memory that basically redis brakes everytime in between 50 millions and 70 millions entries - I tried with hash (hset) in redis but did not do any good Do you have any suggestion how to deal with this, what would be the best thing to use? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Help with big amount of data for routing
Tnx Daniel for this!
I will try.
br
miha
On 05/10/2016 15:09, Daniel Zanutti wrote:
Hi Miha
I have a similar situation, but around 20 M routes.
The native routing mecanims wasn't performing well, so I developed a
custom mecanism using Opensips scripting. Everything is stored on
MySQL database.
The best approach was use avp_db_query to get the route, the primary
key (and index) of the table is the route prefix and stored as BIG
INT, so you have up to 19 digits of routes, which is OK to me. I could
achieve more than 100 cps with this method.
You have to find the longest route "by hand", so I developed this
procedure:
DELIMITER $$
CREATE DEFINER=`root`@`localhost` PROCEDURE `getLongestRoute`(IN route
VARCHAR(50), OUT bestroute BIGINT, OUT regionid INT)
BEGIN
DECLARE rotatemp VARCHAR(50);
DECLARE tempprefix BIGINT;
CREATE TEMPORARY TABLE IF NOT EXISTS temptabrotas ( prefix BIGINT
UNSIGNED) ENGINE=HEAP;
SET rotatemp = SUBSTRING(route, 1, LENGTH(route));
INSERT INTO temptabrotas (prefix) VALUES (rotatemp);
WHILE (LENGTH(rotatemp) > 1) DO
SET rotatemp = SUBSTRING(route, 1, LENGTH(rotatemp)-1);
INSERT INTO temptabrotas (prefix) VALUES (rotatemp);
END WHILE;
SELECT routes.prefix, routes.regionid FROM routes
INNER JOIN temptabrotas
ON routes.prefix = temptabrotas.prefix
ORDER BY routes.prefix DESC
LIMIT 1
INTO bestroute, regionid;
DROP TABLE temptabrotas;
END$$
DELIMITER ;
Hope it helps.
Regards
On Wed, Oct 5, 2016 at 4:16 AM, Miha <mailto:m...@softnet.si>> wrote:
Hi Alex
i tried, but mysql takes so long time for every select. What do u
have in mind?
tnx
miha
On 05/10/2016 08:46, Alex Balashov wrote:
Why do you believe that using a traditional RDBM necessarily
means slow lookups?
On 10/05/2016 02:44 AM, Miha wrote:
HI
the is not really opensips issue:) I need somehow to store
big amount of
data for routing.
To every telephone operator I must send RURI like
Net_ID+Telephone_number (value indicates to who number
belongs to). In
this country they have around 120 millions of numbers.
After i have all NET_IDs with numbers I will use drouting
for routing
numbers to right operator based on NET_ID.
Here is the issue:
- I tried this with redis (lookup must be quick) but this
takes so much
memory that basically redis brakes everytime in between 50
millions and
70 millions entries
- I tried with hash (hset) in redis but did not do any good
Do you have any suggestion how to deal with this, what
would be the best
thing to use?
tnx
miha
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Re: [OpenSIPS-Users] Help with big amount of data for routing
Hi Alex i tried, but mysql takes so long time for every select. What do u have in mind? tnx miha On 05/10/2016 08:46, Alex Balashov wrote: Why do you believe that using a traditional RDBM necessarily means slow lookups? On 10/05/2016 02:44 AM, Miha wrote: HI the is not really opensips issue:) I need somehow to store big amount of data for routing. To every telephone operator I must send RURI like Net_ID+Telephone_number (value indicates to who number belongs to). In this country they have around 120 millions of numbers. After i have all NET_IDs with numbers I will use drouting for routing numbers to right operator based on NET_ID. Here is the issue: - I tried this with redis (lookup must be quick) but this takes so much memory that basically redis brakes everytime in between 50 millions and 70 millions entries - I tried with hash (hset) in redis but did not do any good Do you have any suggestion how to deal with this, what would be the best thing to use? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Help with big amount of data for routing
HI the is not really opensips issue:) I need somehow to store big amount of data for routing. To every telephone operator I must send RURI like Net_ID+Telephone_number (value indicates to who number belongs to). In this country they have around 120 millions of numbers. After i have all NET_IDs with numbers I will use drouting for routing numbers to right operator based on NET_ID. Here is the issue: - I tried this with redis (lookup must be quick) but this takes so much memory that basically redis brakes everytime in between 50 millions and 70 millions entries - I tried with hash (hset) in redis but did not do any good Do you have any suggestion how to deal with this, what would be the best thing to use? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] PRACK, 404 not here
Hi Razvan
i was on trip... so sorry for my late response.
I tried with match_dialog() but then I got loop.
59.878504 SBC_2 -> SBC_1 SIP 524 Request: PRACK
sip:38111422@SBC_1:5060;transport=udp |
59.878742 SBC_1 -> SBC_1 SIP 530 Request: PRACK
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
59.878959 SBC_1 -> SBC_1 SIP 596 Request: PRACK
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
59.879128 SBC_1 -> SBC_1 SIP 662 Request: PRACK
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
59.879299 SBC_1 -> SBC_1 SIP 728 Request: PRACK
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
59.879892 SBC_1 -> SBC_1 SIP 794 Request: PRACK
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
59.880214 SBC_1 -> SBC_1 SIP 860 Request: PRACK
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
59.880417 SBC_1 -> SBC_1 SIP 926 Request: PRACK
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
59.880945 SBC_1 -> SBC_1 SIP 992 Request: PRACK
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
Yes in my script I have: listen=udp:SBC_1:5060
Do you have any other recommendation what I should try?
tnx
miha
On 07/09/2016 10:12, Răzvan Crainea wrote:
Hi, Miha!
It looks like loose_route() fails - did you try to look into the logs
and see if it indicates something? Is the SBC_1 IP advertised in the
Route header a listener of OpenSIPS?
Also, if loose_route() fails, you should still try to match the PRACK
against the dialog. So your scripting logic should look like this:
if (has_totag()) {
if (loose_route() || match_dialog()) {
...
}
}
Let us know how that goes.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 09/07/2016 10:38 AM, Miha wrote:
Hi
i have one issue and do not know how to solve it...
Initial invite:
U SBC_2:5060 -> SBC_1:5060 INVITE
sip:7422@SBC_1:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP
SBC_2:5060;branch=z9hG4bK57fa.67ccbb16.0. From:
;tag=*1875283502*. To:
. Call-ID:
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193456 INVITE.
Contact: . Alert-Info:
. Allow:
REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,P
SeqU PBX:5060 -> SBC_2:5060 PRACK sip:SBC_2;did=8d9.43418513 SIP/2.0.
Via: SIP/2.0/UDP PBX:5060;branch=z9hG4bK-002AF6E3;rport. From:
;tag=*1875283502*. To:
;tag=*FamBBcayZeKgF*.
Call-ID:*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK.
Content-Length: 0. Max-Forwards: 70. RAck: 1601153264 1193456 INVITE.
. Seq U SBC_2:5060 -> SBC_1:5060 PRACK
sip:7422@SBC_1:5060;transport=udp SIP/2.0. Route:
. Via: SIP/2.0/UDP
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From:
;tag=*1875283502*. To:
;tag=*FamBBcayZeKgF*. Call-ID:
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK.
Content-Length: 0. Max-Forwards: 69. RAck: 1601153264 1193456 INVITE.
Seq
U SBC_1:5060 -> SBC_2:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From:
;tag=1875283502. To:
;tag=FamBBcayZeKgF. Call-ID:
fb9e258ae909d311a85a0090332e03ed@PBX. CSeq: 1193457 PRACK. Server:
OpenSIPS (2.1.1 (x86_64/linux)). Content-Length: 0.
Why I am getting 404 from Opensips. is should be routed like seq request, right?
if (has_totag()) { # sequential requests within a dialog should #
take the path determined by record-routing if (loose_route()) {
xlog("loose_route"); #if ($DLG_status!=NULL) xlog("dlg_status"); if
(!validate_dialog()){ fix_route_dialog(); xlog("fix_route_dialog"); }
if (is_method("BYE")) { setflag(1); # do accounting ...
#setflag(ACC_FAILED); # ... even if the transaction fails } else if
(is_method("INVITE")) { # even if in most of the cases is useless, do
RR for # re-INVITEs alos, as some buggy clients do change route set #
during the dialog. record_route(); xlog("check_fraud"); } # route it
out to whatever destination was set by loose_route() # in $du
(destination URI). route(relay); } else { if ( is_method("ACK") ) {
if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be
an ACK after # a 487 or e.g. 404 from upstream server t_relay();
exit; } else { # ACK without matching transaction -> # ignore and
discard exit; } } sl_send_reply("404","Not here"); } exit; tnx miha
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Re: [OpenSIPS-Users] PRACK, 404 not here
Hi Razvan
will take a look and let you know.
I tried with match_dialog() but then got some loop.
tnx
miha
On 07/09/2016 10:12, Răzvan Crainea wrote:
Hi, Miha!
It looks like loose_route() fails - did you try to look into the logs
and see if it indicates something? Is the SBC_1 IP advertised in the
Route header a listener of OpenSIPS?
Also, if loose_route() fails, you should still try to match the PRACK
against the dialog. So your scripting logic should look like this:
if (has_totag()) {
if (loose_route() || match_dialog()) {
...
}
}
Let us know how that goes.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 09/07/2016 10:38 AM, Miha wrote:
Hi
i have one issue and do not know how to solve it...
Initial invite:
U SBC_2:5060 -> SBC_1:5060 INVITE
sip:7422@SBC_1:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP
SBC_2:5060;branch=z9hG4bK57fa.67ccbb16.0. From:
;tag=*1875283502*. To:
. Call-ID:
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193456 INVITE.
Contact: . Alert-Info:
. Allow:
REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,P
SeqU PBX:5060 -> SBC_2:5060 PRACK sip:SBC_2;did=8d9.43418513 SIP/2.0.
Via: SIP/2.0/UDP PBX:5060;branch=z9hG4bK-002AF6E3;rport. From:
;tag=*1875283502*. To:
;tag=*FamBBcayZeKgF*.
Call-ID:*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK.
Content-Length: 0. Max-Forwards: 70. RAck: 1601153264 1193456 INVITE.
. Seq U SBC_2:5060 -> SBC_1:5060 PRACK
sip:7422@SBC_1:5060;transport=udp SIP/2.0. Route:
. Via: SIP/2.0/UDP
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From:
;tag=*1875283502*. To:
;tag=*FamBBcayZeKgF*. Call-ID:
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK.
Content-Length: 0. Max-Forwards: 69. RAck: 1601153264 1193456 INVITE.
Seq
U SBC_1:5060 -> SBC_2:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From:
;tag=1875283502. To:
;tag=FamBBcayZeKgF. Call-ID:
fb9e258ae909d311a85a0090332e03ed@PBX. CSeq: 1193457 PRACK. Server:
OpenSIPS (2.1.1 (x86_64/linux)). Content-Length: 0.
Why I am getting 404 from Opensips. is should be routed like seq request, right?
if (has_totag()) { # sequential requests within a dialog should #
take the path determined by record-routing if (loose_route()) {
xlog("loose_route"); #if ($DLG_status!=NULL) xlog("dlg_status"); if
(!validate_dialog()){ fix_route_dialog(); xlog("fix_route_dialog"); }
if (is_method("BYE")) { setflag(1); # do accounting ...
#setflag(ACC_FAILED); # ... even if the transaction fails } else if
(is_method("INVITE")) { # even if in most of the cases is useless, do
RR for # re-INVITEs alos, as some buggy clients do change route set #
during the dialog. record_route(); xlog("check_fraud"); } # route it
out to whatever destination was set by loose_route() # in $du
(destination URI). route(relay); } else { if ( is_method("ACK") ) {
if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be
an ACK after # a 487 or e.g. 404 from upstream server t_relay();
exit; } else { # ACK without matching transaction -> # ignore and
discard exit; } } sl_send_reply("404","Not here"); } exit; tnx miha
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[OpenSIPS-Users] PRACK, 404 not here
Hi
i have one issue and do not know how to solve it...
Initial invite:
U SBC_2:5060 -> SBC_1:5060 INVITE sip:7422@SBC_1:5060;user=phone
SIP/2.0. Via: SIP/2.0/UDP SBC_2:5060;branch=z9hG4bK57fa.67ccbb16.0.
From: ;tag=*1875283502*. To:
. Call-ID:
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193456 INVITE. Contact:
. Alert-Info: .
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,P
SeqU PBX:5060 -> SBC_2:5060 PRACK sip:SBC_2;did=8d9.43418513 SIP/2.0. Via:
SIP/2.0/UDP PBX:5060;branch=z9hG4bK-002AF6E3;rport. From:
;tag=*1875283502*. To:
;tag=*FamBBcayZeKgF*.
Call-ID:*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK.
Content-Length: 0. Max-Forwards: 70. RAck: 1601153264 1193456 INVITE. .
Seq U SBC_2:5060 -> SBC_1:5060 PRACK
sip:7422@SBC_1:5060;transport=udp SIP/2.0. Route:
. Via: SIP/2.0/UDP
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From:
;tag=*1875283502*. To:
;tag=*FamBBcayZeKgF*. Call-ID:
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK.
Content-Length: 0. Max-Forwards: 69. RAck: 1601153264 1193456 INVITE.
Seq
U SBC_1:5060 -> SBC_2:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From:
;tag=1875283502. To:
;tag=FamBBcayZeKgF. Call-ID:
fb9e258ae909d311a85a0090332e03ed@PBX. CSeq: 1193457 PRACK. Server:
OpenSIPS (2.1.1 (x86_64/linux)). Content-Length: 0.
Why I am getting 404 from Opensips. is should be routed like seq request, right?
if (has_totag()) { # sequential requests within a dialog should # take
the path determined by record-routing if (loose_route()) {
xlog("loose_route"); #if ($DLG_status!=NULL) xlog("dlg_status"); if
(!validate_dialog()){ fix_route_dialog(); xlog("fix_route_dialog"); } if
(is_method("BYE")) { setflag(1); # do accounting ...
#setflag(ACC_FAILED); # ... even if the transaction fails } else if
(is_method("INVITE")) { # even if in most of the cases is useless, do RR
for # re-INVITEs alos, as some buggy clients do change route set #
during the dialog. record_route(); xlog("check_fraud"); } # route it out
to whatever destination was set by loose_route() # in $du (destination
URI). route(relay); } else { if ( is_method("ACK") ) { if (
t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK
after # a 487 or e.g. 404 from upstream server t_relay(); exit; } else {
# ACK without matching transaction -> # ignore and discard exit; } }
sl_send_reply("404","Not here"); } exit; tnx miha
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Re: [OpenSIPS-Users] Issue with ACK and rtpproxy, setID
Ben tnx. make sense :) br miha On 30/06/2016 15:17, Newlin, Ben wrote: AVPs are tied to a transaction, so the transaction must be matched before they will be available. You should use t_check_trans() to do this. However, I think this will not work for you because ACKs are their own transactions and I don’t believe they will have access to the AVPs from the INVITE transaction. If you need to store state information across multiple transactions, you will need to use the dialog module and the $dlg_val variables. These persist across the entire SIP call. Ben Newlin On 6/30/16, 8:06 AM, "users-boun...@lists.opensips.org on behalf of Miha" wrote: HI I have two RTPproxies and doing spiral so that I can put them in chain. Before t_relay() I am setting avps (setID) so that I can do rtpproxy_answer latter if there is SDP in ACK. The issue is that avp is all the time null for ACK (for Initial invite avp was set). In TM module i set onreply_avp_mode to 1. Is there anything else I must do or do you suggest some other approche? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Issue with ACK and rtpproxy, setID
HI I have two RTPproxies and doing spiral so that I can put them in chain. Before t_relay() I am setting avps (setID) so that I can do rtpproxy_answer latter if there is SDP in ACK. The issue is that avp is all the time null for ACK (for Initial invite avp was set). In TM module i set onreply_avp_mode to 1. Is there anything else I must do or do you suggest some other approche? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] engage_rtp_proxy()
HI
engage_rtp_proxy() work ok. I was having some other issue with dialog.
Tnx to @Bogdan I figure it out.
br
miha
On 31/05/2016 15:54, Sasmita Panda wrote:
In my case its working great . So I haven't done such experiments to
know what is happening with dialog module . We are using this form
years .
If you got to know then let me know also . That may help me in
future .
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 6:56 PM, Miha <mailto:m...@softnet.si>> wrote:
@Sasmita I had writen cfg script like this and it works. I tried
than with engage_rtp_proxy() but did not work automatically, that
is why i asked :)
So, can be some issue with dialog module? Not configured properly?
tnx
miha
On 31/05/2016 15:21, Sasmita Panda wrote:
Yes . This should happen . But I don't know the exact problem .
What I explain is the way we are using rtpproxy .
This is clearly mention in the document also .. You can go
through opensips.org <http://opensips.org>
This is what we are doing . Rest I am not an expertise in
opensips .
route {
...
if (is_method("INVITE")) {
if (has_body("application/sdp")) {
if (rtpproxy_offer())
t_on_reply("1");
} else {
t_on_reply("2");
}
}
if (is_method("ACK") && has_body("application/sdp"))
rtpproxy_answer();
...
}
onreply_route[1]
{
...
if (has_body("application/sdp"))
rtpproxy_answer();
...
}
onreply_route[2]
{
...
if (has_body("application/sdp"))
rtpproxy_offer();
...
}
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 6:32 PM, Max Mühlbronner mailto:m...@42com.com>> wrote:
Hi,
@Miha: Are you sure that it does not automatically set the
rtpproxies for 200OK & ACK?
@Sasmita: According to the documentation it is not necessary
to invoke engage_rtp_proxy() for replies as this is handled
by the dialog module.
"Function must only be called for the initial INVITE and
internally takes care of rewriting the body of 200 OKs and
ACKs. "
Best Regards
Max M.
On 31.05.2016 14:42, Miha wrote:
@Sasmita, totally clear :)
I asked wrong question :)
What is the difference between using engage_rtp_proxy() or
using rtpproxy_offer(), rtpproxy_answer()?
tnx
miha
On 31/05/2016 14:39, Sasmita Panda wrote:
If you are using in INVITE , then it should be
offer . Because firstly we are offering media to someone .
If its 200 Ok then it will be answer because the 2nd party
is answering the call .
If there is no sdp in INVITE but in ACK , then it
will get reversed . In 200 OK you should offer and in ACK
you have to answer .
This can be done in loop .
I hope I make you understand .
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 6:02 PM, Miha mailto:m...@softnet.si> <mailto:m...@softnet.si>
<mailto:m...@softnet.si>> wrote:
ok tnx. I understand documentation on wrong way.
But then, what is the difference with using rtpproxy
offer, answer ?
br
mia
On 31/05/2016 14:17, Sasmita Panda wrote:
If there is sdp in ACK and u wanted to engage rtp
proxy , the
you have to write it inside ACK also ... By writing
for INVITE
cant help you to update ACK also . For 200 OK , you
must write it
in reply route .
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
mailto:jo...@democon.be>
<mailto:jo...@democon.be> <mailto:jo...@democon.be>> wrote:
put it also in reply route.
2016-05-31 13:42 GMT+02:00 Miha mailto:m...@softnet.si>
<mailto:m...@softnet.si> <mailto:m...@softnet.si>>:
HI
if I use engage_rtp_proxy(), I can use it only
on initial
INVITE and opensips should automatically
rewritten also
Re: [OpenSIPS-Users] engage_rtp_proxy()
@Sasmita I had writen cfg script like this and it works. I tried than
with engage_rtp_proxy() but did not work automatically, that is why i
asked :)
So, can be some issue with dialog module? Not configured properly?
tnx
miha
On 31/05/2016 15:21, Sasmita Panda wrote:
Yes . This should happen . But I don't know the exact problem . What I
explain is the way we are using rtpproxy .
This is clearly mention in the document also .. You can go through
opensips.org <http://opensips.org>
This is what we are doing . Rest I am not an expertise in opensips .
route {
...
if (is_method("INVITE")) {
if (has_body("application/sdp")) {
if (rtpproxy_offer())
t_on_reply("1");
} else {
t_on_reply("2");
}
}
if (is_method("ACK") && has_body("application/sdp"))
rtpproxy_answer();
...
}
onreply_route[1]
{
...
if (has_body("application/sdp"))
rtpproxy_answer();
...
}
onreply_route[2]
{
...
if (has_body("application/sdp"))
rtpproxy_offer();
...
}
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 6:32 PM, Max Mühlbronner <mailto:m...@42com.com>> wrote:
Hi,
@Miha: Are you sure that it does not automatically set the
rtpproxies for 200OK & ACK?
@Sasmita: According to the documentation it is not necessary to
invoke engage_rtp_proxy() for replies as this is handled by the
dialog module.
"Function must only be called for the initial INVITE and
internally takes care of rewriting the body of 200 OKs and ACKs. "
Best Regards
Max M.
On 31.05.2016 14:42, Miha wrote:
@Sasmita, totally clear :)
I asked wrong question :)
What is the difference between using engage_rtp_proxy() or using
rtpproxy_offer(), rtpproxy_answer()?
tnx
miha
On 31/05/2016 14:39, Sasmita Panda wrote:
If you are using in INVITE , then it should be offer .
Because firstly we are offering media to someone . If its 200 Ok
then it will be answer because the 2nd party is answering the
call .
If there is no sdp in INVITE but in ACK , then it will get
reversed . In 200 OK you should offer and in ACK you have to
answer .
This can be done in loop .
I hope I make you understand .
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 6:02 PM, Miha mailto:m...@softnet.si> <mailto:m...@softnet.si>
<mailto:m...@softnet.si>> wrote:
ok tnx. I understand documentation on wrong way.
But then, what is the difference with using rtpproxy offer,
answer ?
br
mia
On 31/05/2016 14:17, Sasmita Panda wrote:
If there is sdp in ACK and u wanted to engage rtp proxy , the
you have to write it inside ACK also ... By writing for
INVITE
cant help you to update ACK also . For 200 OK , you must
write it
in reply route .
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
mailto:jo...@democon.be>
<mailto:jo...@democon.be> <mailto:jo...@democon.be>> wrote:
put it also in reply route.
2016-05-31 13:42 GMT+02:00 Miha mailto:m...@softnet.si>
<mailto:m...@softnet.si> <mailto:m...@softnet.si>>:
HI
if I use engage_rtp_proxy(), I can use it only on
initial
INVITE and opensips should automatically rewritten
also
200 OK and ACK with SDP, right?
But when I am using this function, I can see from
trace
that only SDP for initial invite is rewritten, 200 ok
with sdp is not changed. Must I do something else?
Rtpproxy is not running in bridge mode.
tnx
miha
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Re: [OpenSIPS-Users] engage_rtp_proxy()
@Max, yes I am sure. Did some traces...
That is way I asked what I could be doing wrong. I read that there is an
issue doing it with rtpproxy in bridge, but in my case it is not in bridge.
tnx
miha
On 31/05/2016 15:02, Max Mühlbronner wrote:
Hi,
@Miha: Are you sure that it does not automatically set the rtpproxies
for 200OK & ACK?
@Sasmita: According to the documentation it is not necessary to invoke
engage_rtp_proxy() for replies as this is handled by the dialog module.
"Function must only be called for the initial INVITE and internally
takes care of rewriting the body of 200 OKs and ACKs. "
Best Regards
Max M.
On 31.05.2016 14:42, Miha wrote:
@Sasmita, totally clear :)
I asked wrong question :)
What is the difference between using engage_rtp_proxy() or using
rtpproxy_offer(), rtpproxy_answer()?
tnx
miha
On 31/05/2016 14:39, Sasmita Panda wrote:
If you are using in INVITE , then it should be offer .
Because firstly we are offering media to someone . If its 200 Ok
then it will be answer because the 2nd party is answering the call .
If there is no sdp in INVITE but in ACK , then it will get
reversed . In 200 OK you should offer and in ACK you have to answer .
This can be done in loop .
I hope I make you understand .
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 6:02 PM, Miha <mailto:m...@softnet.si>> wrote:
ok tnx. I understand documentation on wrong way.
But then, what is the difference with using rtpproxy offer,
answer ?
br
mia
On 31/05/2016 14:17, Sasmita Panda wrote:
If there is sdp in ACK and u wanted to engage rtp proxy , the
you have to write it inside ACK also ... By writing for INVITE
cant help you to update ACK also . For 200 OK , you must write it
in reply route .
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
mailto:jo...@democon.be>> wrote:
put it also in reply route.
2016-05-31 13:42 GMT+02:00 Miha mailto:m...@softnet.si>>:
HI
if I use engage_rtp_proxy(), I can use it only on initial
INVITE and opensips should automatically rewritten also
200 OK and ACK with SDP, right?
But when I am using this function, I can see from trace
that only SDP for initial invite is rewritten, 200 ok
with sdp is not changed. Must I do something else?
Rtpproxy is not running in bridge mode.
tnx
miha
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Re: [OpenSIPS-Users] engage_rtp_proxy()
@Sasmita, totally clear :)
I asked wrong question :)
What is the difference between using engage_rtp_proxy() or using
rtpproxy_offer(), rtpproxy_answer()?
tnx
miha
On 31/05/2016 14:39, Sasmita Panda wrote:
If you are using in INVITE , then it should be offer . Because
firstly we are offering media to someone . If its 200 Ok then it will
be answer because the 2nd party is answering the call .
If there is no sdp in INVITE but in ACK , then it will get
reversed . In 200 OK you should offer and in ACK you have to answer .
This can be done in loop .
I hope I make you understand .
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 6:02 PM, Miha <mailto:m...@softnet.si>> wrote:
ok tnx. I understand documentation on wrong way.
But then, what is the difference with using rtpproxy offer, answer ?
br
mia
On 31/05/2016 14:17, Sasmita Panda wrote:
If there is sdp in ACK and u wanted to engage rtp proxy , the
you have to write it inside ACK also ... By writing for INVITE
cant help you to update ACK also . For 200 OK , you must write it
in reply route .
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
mailto:jo...@democon.be>> wrote:
put it also in reply route.
2016-05-31 13:42 GMT+02:00 Miha mailto:m...@softnet.si>>:
HI
if I use engage_rtp_proxy(), I can use it only on initial
INVITE and opensips should automatically rewritten also
200 OK and ACK with SDP, right?
But when I am using this function, I can see from trace
that only SDP for initial invite is rewritten, 200 ok
with sdp is not changed. Must I do something else?
Rtpproxy is not running in bridge mode.
tnx
miha
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Re: [OpenSIPS-Users] engage_rtp_proxy()
ok tnx. I understand documentation on wrong way.
But then, what is the difference with using rtpproxy offer, answer ?
br
mia
On 31/05/2016 14:17, Sasmita Panda wrote:
If there is sdp in ACK and u wanted to engage rtp proxy , the you
have to write it inside ACK also ... By writing for INVITE cant help
you to update ACK also . For 200 OK , you must write it in reply route .
*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/
On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq <mailto:jo...@democon.be>> wrote:
put it also in reply route.
2016-05-31 13:42 GMT+02:00 Miha mailto:m...@softnet.si>>:
HI
if I use engage_rtp_proxy(), I can use it only on initial
INVITE and opensips should automatically rewritten also 200 OK
and ACK with SDP, right?
But when I am using this function, I can see from trace that
only SDP for initial invite is rewritten, 200 ok with sdp is
not changed. Must I do something else?
Rtpproxy is not running in bridge mode.
tnx
miha
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[OpenSIPS-Users] engage_rtp_proxy()
HI if I use engage_rtp_proxy(), I can use it only on initial INVITE and opensips should automatically rewritten also 200 OK and ACK with SDP, right? But when I am using this function, I can see from trace that only SDP for initial invite is rewritten, 200 ok with sdp is not changed. Must I do something else? Rtpproxy is not running in bridge mode. tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Compiling issue, recipe for target 'sipmemcache.o' failed
Hi, need a little help. I have installed memcache-dev, what else could i be missing? make[2]: Entering directory '/usr/src/opensips_2_1/modules/lua' Compiling sipmemcache.c sipmemcache.c:25:22: fatal error: memcache.h: No such file or directory #include ^ compilation terminated. ../../Makefile.rules:25: recipe for target 'sipmemcache.o' failed make[2]: *** [sipmemcache.o] Error 1 make[2]: Leaving directory '/usr/src/opensips_2_1/modules/lua' Makefile:194: recipe for target 'modules' failed make[1]: *** [modules] Error 2 make[1]: Leaving directory '/usr/src/opensips_2_1' tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Listening ips and sending call from them
Tnx Sammy, let me try :) br miha On Tue, 1 Mar 2016 16:49:29 -0500 SamyGo wrote: > Hey Miha, > > See if this thread helps you: > http://lists.opensips.org/pipermail/users/2010-October/015150.html > > Regards, > Sammy > > On Tue, Mar 1, 2016 at 9:35 AM, Miha > wrote: > > > HI. > > > > If you have two ips on your server, let say X in Y. > > When you route calls to providers is it possible to use > ip > > X for one provider and ip Y for other provider? > > > > > > tnx > > miah > > > > > > ___ > > Users mailing list > > Users@lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Listening ips and sending call from them
HI. If you have two ips on your server, let say X in Y. When you route calls to providers is it possible to use ip X for one provider and ip Y for other provider? tnx miah ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] websocket t_relay()
tnx Răzvan :) br miha On 23/02/2016 15:24, Răzvan Crainea wrote: Hi, Miha! Make sure you are calling fix_nated_register() function for all REGISTERS coming from a Websocket client. Best regards, Răzvan On 02/23/2016 04:12 PM, Miha wrote: Hi, first time looking at module for websocket (opensips 2.1). Reciving calls from SipJs is working, but how to send call beck to websocket? If i just do lookup location and do t_relay() I am getting "Unresolvable destination"? In location table I have: "sip:a4tn9tda@iibhaol8elto.invalid;transport=ws" tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] websocket t_relay()
Hi, first time looking at module for websocket (opensips 2.1). Reciving calls from SipJs is working, but how to send call beck to websocket? If i just do lookup location and do t_relay() I am getting "Unresolvable destination"? In location table I have: "sip:a4tn9tda@iibhaol8elto.invalid;transport=ws" tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] enum asynchronous
Hi, is there any news when enum could be done asynchronously? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] 2.1.1 segfault
Hi, could some look at this crash, what could be wrong: Nov 2 09:20:17 sbc-adria kernel: opensips[3791]: segfault at 2 ip 0002 sp 7fff0ca4d638 error 14 in opensips[40+1e2000] Nov 2 09:20:17 sbc-adria abrtd: Directory 'ccpp-2015-11-02-09:20:17-3791' creation detected Nov 2 09:20:17 sbc-adria abrt[3811]: Saved core dump of pid 3791 (/usr/local/src/opensips_2_1/opensips) to /var/spool/abrt/ccpp-2015-11-02-09:20:17-3791 (38289408 bytes) Nov 2 09:20:17 sbc-adria ./opensips[3776]: INFO:core:handle_sigs: child process 3791 exited by a signal 11 Nov 2 09:20:17 sbc-adria ./opensips[3776]: INFO:core:handle_sigs: core was generated Nov 2 09:20:17 sbc-adria ./opensips[3776]: INFO:core:handle_sigs: terminating due to SIGCHLD Nov 2 09:20:18 sbc-adria ./opensips[3803]: INFO:core:sig_usr: signal 15 received Nov 2 09:20:18 sbc-adria ./opensips[3802]: INFO:core:sig_usr: signal 15 received Nov 2 09:20:18 sbc-adria ./opensips[3801]: INFO:core:sig_usr: signal 15 received Nov 2 09:20:18 sbc-adria ./opensips[3800]: INFO:core:sig_usr: signal 15 received Nov 2 09:20:18 sbc-adria ./opensips[3799]: INFO:core:sig_usr: signal 15 received Nov 2 09:20:18 sbc-adria ./opensips[3793]: INFO:core:sig_usr: signal 15 received Nov 2 09:20:18 sbc-adria ./opensips[3792]: INFO:core:sig_usr: signal 15 received Nov 2 09:20:18 sbc-adria ./opensips[3790]: INFO:core:sig_usr: signal 15 received Nov 2 09:20:18 sbc-adria ./opensips[3789]: INFO:core:sig_usr: signal 15 received Nov 2 09:20:18 sbc-adria ./opensips[3788]: INFO:core:sig_usr: signal 15 received Nov 2 09:20:18 sbc-adria ./opensips[3787]: INFO:core:sig_usr: signal 15 received Nov 2 09:20:18 sbc-adria ./opensips[3776]: INFO:core:cleanup: cleanup Nov 2 09:20:18 sbc-adria abrtd: Executable '/usr/local/src/opensips_2_1/opensips' doesn't belong to any package and ProcessUnpackaged is set to 'no' Nov 2 09:20:18 sbc-adria abrtd: 'post-create' on '/var/spool/abrt/ccpp-2015-11-02-09:20:17-3791' exited with 1 Nov 2 09:20:18 sbc-adria abrtd: Deleting problem directory '/var/spool/abrt/ccpp-2015-11-02-09:20:17-3791' http://pastebin.com/p5827h1e tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] from 2.1.0 to 2.1.1 version upgrade issue
Hi, I had installed opensips 2.1.0 from git. Now I have try to upgrade (to 2.1.1) it like this: git pull --rebase make all make install After "make all" I got: make[1]: Entering directory `/usr/local/src/opensips_2_1/modules/pdt' make[1]: *** No targets specified and no makefile found. Stop. make[1]: Leaving directory `/usr/local/src/opensips_2_1/modules/pdt' make: *** [modules] Error 2 And after I tried with running it I am getting in logs: Nov 2 08:42:03 sip-adria opensips: ERROR:core:version_control: module version mismatch for regex; core: opensips 2.2-dev (x86_64/linux); module: opensips 2.1.0 (x86_64/linux) The same steps I have done on other server and it worked. Could you please help me with this upgrade from 2.1.0 to 2.1.1. tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] NAT issue
Hi, I have one big issue and I do not know how to fix it. Invite is recived, which has: - User Datagram Protocol, Src Port: 1334 (1334), Dst Port: 5060 (5060) - Via: SIP/2.0/UDP 192.168.131.120:5072;branch=z9hG4bK-a7b4d998 Than 407 is replied back on wrong port (5072) and UAC send again same INVITE with not credentials in it. - User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5072 (5072) This does not work as reply should be send to port 1334. Please let me know if I can fix this and how :) tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parse_uri: bad uri
Hi Razvan, you are right, it was wrong URI. Regarding drouting, i can no to routing with drouting module if in Ruri is "E", right? So the best case would be to transform "E" with dialplan in some number and then do drouting or do you have any better preposition? br miha On 15/09/2015 12:06, Răzvan Crainea wrote: Hi, Miha! That does not look like an URI at all: it contains only the username, not the scheme, host, port, etc. Do you have the original message to track this down? Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 09/15/2015 11:41 AM, Miha wrote: Hi, one of our costumer is sending as uri with "E" like "and opensips return: Sep 15 10:36:30 sip-adria ./opensips[18583]: ERROR:core:parse_uri: bad uri, state 0 parsed: (4) / (14) Sep 15 10:36:30 sip-adria ./opensips[18583]: ERROR:core:parse_sip_msg_uri: bad uri how to deal with this as we need also to send him with this kind of prefix? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] parse_uri: bad uri
Hi, one of our costumer is sending as uri with "E" like "and opensips return: Sep 15 10:36:30 sip-adria ./opensips[18583]: ERROR:core:parse_uri: bad uri, state 0 parsed: (4) / (14) Sep 15 10:36:30 sip-adria ./opensips[18583]: ERROR:core:parse_sip_msg_uri: bad uri how to deal with this as we need also to send him with this kind of prefix? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Group module problem / db_get_user_group
Hi, if I use db_is_user_in() this function returns me ok if user is in this group. if I use db_get_user_group() and do group lookup for exactly the same user (URI) which was used for db_is_user_in() I do not get anything. Could some help we figure it out what I am doing wrong? I am using $fu for lookup. tnx miha | | | | ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] closeddial, migration from 1.10 to 2.1
Hi, Razvan helped me on irc. Group and dialplan module. br miha On 27/05/2015 09:06, Miha wrote: Hi, i noticed that module closeddial was removed. What can be used instead? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] closeddial, migration from 1.10 to 2.1
Hi, i noticed that module closeddial was removed. What can be used instead? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Enum modul
Hi Bogdan, Tnx for info. In future this future would be good to implement;) Ne Miha On Thu, 15 Jan 2015 14:30:59 +0200 Bogdan-Andrei Iancu wrote: > Hi Miha, > > In OpenSIPS, for any DNS lookup, the res_search() > frunction from glib is used, function that uses the > /etc/resolv.conf file for fetching the default DNS > server. > I guess this function does not have the ability to figure > out what's the direct DNS server and go directly there. > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 14.01.2015 08:17, Miha wrote: > > Hi, > > > > I am using enum module for ported numbers. Everything > works fine just one thing is that is bothering me. > > > > If I use dig to see who is authoritative dns server for > one enum zone I can see that this not not dns server > which is set in resolve.conf, so opensips should be doing > this dns request for enum to authoritative dns server. > > > > I have exactly the same config on nextone sbc and it is > sending request to authoritative dns server for this > zone. > > > > How to fix this as now opensips sends request to non > authoritative server for this zone, dns server send it to > enum server and then in oposite direction for resonses > which is not ok. > > > > Thx > > Miha > > > > trace: > > > > 7.476490 opensips -> DNS SERVER DNS 102 Standard query > 0x0687 NAPTR 0.5.1.3.0.3.8.1.6.8.3.enumzone.domain.com > > 7.478184 DNS SERVER -> opensips DNS 143 Standard > query response 0x0687 No such name > > > > [root@sip2 ~]# dig enumzone.domain.com NS > > > > ; <<>> DiG 9.8.2rc1-RedHat-9.8.2-0.17.rc1.el6_4.6 <<>> > enumzone.domain.com NS > > ;; global options: +cmd > > ;; Got answer: > > ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: > 42310 > > ;; flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 0, > ADDITIONAL: 1 > > > > ;; QUESTION SECTION: > > ;enumzone.domain.com.IN NS > > > > ;; ANSWER SECTION: > > enumzone.domain.com. 239 IN NS > ns1.enum.domain.com. > > > > ;; ADDITIONAL SECTION: > > ns1.enum.domain.com.251 IN A > ENUM_SERVER > > > > ;; Query time: 0 msec > > ;; SERVER: DNS SERVER#53(DNS SERVER) > > ;; WHEN: Wed Jan 14 07:05:28 2015 > > ;; MSG SIZE rcvd: 75 > > > > > > > > ___ > > Users mailing list > > Users@lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Enum modul
Hi, I am using enum module for ported numbers. Everything works fine just one thing is that is bothering me. If I use dig to see who is authoritative dns server for one enum zone I can see that this not not dns server which is set in resolve.conf, so opensips should be doing this dns request for enum to authoritative dns server. I have exactly the same config on nextone sbc and it is sending request to authoritative dns server for this zone. How to fix this as now opensips sends request to non authoritative server for this zone, dns server send it to enum server and then in oposite direction for resonses which is not ok. Thx Miha trace: 7.476490 opensips -> DNS SERVER DNS 102 Standard query 0x0687 NAPTR 0.5.1.3.0.3.8.1.6.8.3.enumzone.domain.com 7.478184 DNS SERVER -> opensips DNS 143 Standard query response 0x0687 No such name [root@sip2 ~]# dig enumzone.domain.com NS ; <<>> DiG 9.8.2rc1-RedHat-9.8.2-0.17.rc1.el6_4.6 <<>> enumzone.domain.com NS ;; global options: +cmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 42310 ;; flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 0, ADDITIONAL: 1 ;; QUESTION SECTION: ;enumzone.domain.com.IN NS ;; ANSWER SECTION: enumzone.domain.com. 239 IN NS ns1.enum.domain.com. ;; ADDITIONAL SECTION: ns1.enum.domain.com.251 IN A ENUM_SERVER ;; Query time: 0 msec ;; SERVER: DNS SERVER#53(DNS SERVER) ;; WHEN: Wed Jan 14 07:05:28 2015 ;; MSG SIZE rcvd: 75 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] table_version: invalid version
Tnx Liviu,
I have imported mysql shama of 1.10 and now it is ok.
tnx
miha
On 20/10/2014 12:48, Liviu Chircu wrote:
It looks like you're using a pre-1.11 OpenSIPS and you are pointing it
to a future (1.11+) dr_gateways table structure. There are several
ways to fix this specific problem:
- simply change version table entry from 6 -> 5 only if no other
OpenSIPS instance uses it
- create a new database for your current version with opensipsdbctl
Liviu
On 10/20/2014 12:55 PM, Miha wrote:
Liviu,
this is the same thing: ERROR:core:db_check_table_version: invalid
version 6 for table dr_gateways found, expected 5
as i do not find any syntax error.
tnx
miha
On 20/10/2014 11:17, Miha wrote:
Hi Liviu,
tnx for quick reponse. Yes, you were right, I was looking at logs
and did not know that this could mean also invalid syntax.
tnx
miha
On 20/10/2014 11:05, Liviu Chircu wrote:
Hi miha,
It looks like you have a typo in your script. Probably something
like "www_authorize("...", "*suscriber*")" !!
Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 10/20/2014 11:51 AM, Miha wrote:
Hi,
i have installed opensips from repository. I am using centos os.
From what i can see in logs my opensips does not start do to wrong
db version
Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]:
ERROR:core:db_check_table_version: invalid version 0 for table
suscriber found, expected 7
Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]:
ERROR:auth_db:auth_fixup: error during table version check.
I tried to import db also from source file on web but still the same.
tnx for help
miha
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Re: [OpenSIPS-Users] table_version: invalid version
Liviu,
this is the same thing: ERROR:core:db_check_table_version: invalid
version 6 for table dr_gateways found, expected 5
as i do not find any syntax error.
tnx
miha
On 20/10/2014 11:17, Miha wrote:
Hi Liviu,
tnx for quick reponse. Yes, you were right, I was looking at logs and
did not know that this could mean also invalid syntax.
tnx
miha
On 20/10/2014 11:05, Liviu Chircu wrote:
Hi miha,
It looks like you have a typo in your script. Probably something like
"www_authorize("...", "*suscriber*")" !!
Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 10/20/2014 11:51 AM, Miha wrote:
Hi,
i have installed opensips from repository. I am using centos os.
From what i can see in logs my opensips does not start do to wrong
db version
Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]:
ERROR:core:db_check_table_version: invalid version 0 for table
suscriber found, expected 7
Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]:
ERROR:auth_db:auth_fixup: error during table version check.
I tried to import db also from source file on web but still the same.
tnx for help
miha
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Re: [OpenSIPS-Users] table_version: invalid version
Hi Liviu,
tnx for quick reponse. Yes, you were right, I was looking at logs and
did not know that this could mean also invalid syntax.
tnx
miha
On 20/10/2014 11:05, Liviu Chircu wrote:
Hi miha,
It looks like you have a typo in your script. Probably something like
"www_authorize("...", "*suscriber*")" !!
Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 10/20/2014 11:51 AM, Miha wrote:
Hi,
i have installed opensips from repository. I am using centos os. From
what i can see in logs my opensips does not start do to wrong db version
Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]:
ERROR:core:db_check_table_version: invalid version 0 for table
suscriber found, expected 7
Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]:
ERROR:auth_db:auth_fixup: error during table version check.
I tried to import db also from source file on web but still the same.
tnx for help
miha
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[OpenSIPS-Users] table_version: invalid version
Hi, i have installed opensips from repository. I am using centos os. From what i can see in logs my opensips does not start do to wrong db version Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]: ERROR:core:db_check_table_version: invalid version 0 for table suscriber found, expected 7 Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]: ERROR:auth_db:auth_fixup: error during table version check. I tried to import db also from source file on web but still the same. tnx for help miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] NAT, UPDATE problem
hi,
I have solved this by adding onreply route and it it I put
fix_nated_contact();
Now contact in 200ok is fixed and media is ok.
tnx
miha
On 16/10/2014 15:53, Miha wrote:
Hi,
in my cfg file i have this:
if (nat_uac_test("18")) {
xlog("fixing nat");
if (method=="REGISTER") {
fix_nated_register();
fix_nated_contact();
} else {
fix_nated_contact();
}
force_rport();
}
But when 200ok with sdp is send this part of script does not execute
as in contact is still private ip. I have 18 in nat_uac_test so if src
port port in via are different this should be triggered but it is not
that is why proxy sends media to private ip.
Here is my sip trace:
http://pastebin.com/KvNdttj9
Tnx
miha
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[OpenSIPS-Users] NAT, UPDATE problem
Hi,
in my cfg file i have this:
if (nat_uac_test("18")) {
xlog("fixing nat");
if (method=="REGISTER") {
fix_nated_register();
fix_nated_contact();
} else {
fix_nated_contact();
}
force_rport();
}
But when 200ok with sdp is send this part of script does not execute as
in contact is still private ip. I have 18 in nat_uac_test so if src port
port in via are different this should be triggered but it is not that is
why proxy sends media to private ip.
Here is my sip trace:
http://pastebin.com/KvNdttj9
Tnx
miha
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Re: [OpenSIPS-Users] About "Not enough free memory, no more shm memory and out of mem" Error
how do you start opensips, how much private mem and shared mam do you asigne? brM On 15/10/2014 14:25, wilddra...@sina.com wrote: Hi, My friends, When I do a stress testing for opensips(version: 1.11.2TLS) , I received the following errors. How to tuning it to solve these problem. anyone can help me, thanks. opensips[1362]: ERROR:dialog:dialog_update_db: could not add another dialog to db opensips[1358]: WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation opensips[1358]: ERROR:usrloc:new_ucontact: no more shm memory opensips[1358]: ERROR:usrloc:mem_insert_ucontact: failed to create new contact opensips[1358]: ERROR:usrloc:insert_ucontact: failed to insert contact opensips[1358]: ERROR:registrar:insert_contacts: failed to insert contact opensips[1359]: WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation opensips[1359]: ERROR:tm:new_t: out of mem opensips[1359]: ERROR:tm:t_newtran: new_t failed opensips[1360]: WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation opensips[1360]: ERROR:tm:new_t: out of mem opensips[1360]: ERROR:tm:t_newtran: new_t failed opensips[1359]: WARNING:core:fm_malloc: Not enough free memory, will attempt defragmentation opensips[1359]: ERROR:usrloc:new_ucontact: no more shm memory opensips[1359]: ERROR:usrloc:mem_insert_ucontact: failed to create new contact opensips[1359]: ERROR:usrloc:insert_ucontact: failed to insert contact opensips[1359]: ERROR:registrar:insert_contacts: failed to insert contact opensips[1362]: CRITICAL:db_mysql:wrapper_single_mysql_stmt_execute: driver error (1048): Column 'callee_contact' cannot be null wilddra...@sina.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] enum waiting
Hi Bogdan, tnx for this info. Will try with this dns_retr_time and dns_retr_no. And yes, this is very painful as whole voip system is not working not just in this exp portable numbers ;) Br Miha On 14/10/2014 18:28, Bogdan-Andrei Iancu wrote: Hi Miha, Indeed, these blocking I/O's are painful and the 2.x release will start addressing this problem. The 2.1 code is already making first steps in solving it. For radius, OpenSIPS cannot control (via the libradiusclient lib) the timeout on how long to wait for the responses. For enum/dns, see dns_retr_time and dns_retr_no : http://www.opensips.org/Documentation/Script-CoreParameters-1-11#toc44 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 14.10.2014 09:20, Miha wrote: Hi, last time radius server was not sending back requests, so opensips was waiting and waiting and voip was not working. Today our enum server broke and opensips was agin waiting and waiting for responses and our voip did not work for the time that enum server was down. Is it possible to define some time for how long opensips should wait so that all other call will work and not that the whole system drops? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] enum waiting
Hi, last time radius server was not sending back requests, so opensips was waiting and waiting and voip was not working. Today our enum server broke and opensips was agin waiting and waiting for responses and our voip did not work for the time that enum server was down. Is it possible to define some time for how long opensips should wait so that all other call will work and not that the whole system drops? tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] strange behaviour, UAC needs to be rebooted
Hi Bogdan, it is not related with NAT as modems are on public ip's. From what I have understand is that modem memory become fully used as some memory leak thing. I also thruly do not know when exacly happens as I am waiting for some explenation feedback from vendor (about memory they told me throught the phone). I will try to get some info from vendor as quick as possible to describe things more clear. br miha Dne 6/5/2014 1:49 PM, piše Bogdan-Andrei Iancu: Hi Miha, I do not understand the explanation from the modem vendor (about the buffer) - what data is kept in that buffer ? is it something SIP related ? Do you do NAT pinging to those devices ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05.06.2014 14:15, Miha wrote: Hi, we are experiancing strange thing on UAC's that are registered on opensips (not all). After I while they just stop responding, call reaches UAC, you can hear rinback tone but the UAC does not send this call to heandset, also outside calls do not work. When you restart UAC everything is back to normal, for 1 week, maybe 2 but than again UAC must be restarted. This we are experiancing on cable modems and also on gigaset ip phones. Company that we are buying modems from, send as fw fix (they said that the modem buffer is full and that is why this happens on modem). Ok on first look this is not related with opensips but it is UAC thing, yes i know that but interesting thing is that when we have this modems on three other softswithes this did not happen for about 6 years (this is how long we are having this modems). If someone has experianced the same thing i have a clue what could be wrong or could halp in this case please help me:) p.s.: company send as non production fw, which has hallped on this issue but still they are refusing to do official releas as some of this modem are out of support (not supported any more). tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] strange behaviour, UAC needs to be rebooted
Hi, we are experiancing strange thing on UAC's that are registered on opensips (not all). After I while they just stop responding, call reaches UAC, you can hear rinback tone but the UAC does not send this call to heandset, also outside calls do not work. When you restart UAC everything is back to normal, for 1 week, maybe 2 but than again UAC must be restarted. This we are experiancing on cable modems and also on gigaset ip phones. Company that we are buying modems from, send as fw fix (they said that the modem buffer is full and that is why this happens on modem). Ok on first look this is not related with opensips but it is UAC thing, yes i know that but interesting thing is that when we have this modems on three other softswithes this did not happen for about 6 years (this is how long we are having this modems). If someone has experianced the same thing i have a clue what could be wrong or could halp in this case please help me:) p.s.: company send as non production fw, which has hallped on this issue but still they are refusing to do official releas as some of this modem are out of support (not supported any more). tnx miha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
