Re: [OpenSIPS-Users] CDR in flat file
Hi, Gary!
The configuration file looks ok. Can you check if the /var/log/acc
directory exists? Do you see any errors in the logs?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/05/2014 11:40 PM, Gary Nyquist wrote:
Hi,
I am trying to configure the OpenSIPS v.1.10 to make it write the CDRs
to flat files.
But no luck yet.
The "opensips.cfg" looks like this:
...
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
loadmodule "acc.so"
modparam("acc", "detect_direction", 1)
modparam("acc", "failed_transaction_flag", "ACC_FAILED")
modparam("acc", "db_url", "flatstore:/var/log/acc")
modparam("acc", "log_flag", "LOG_FLAG")
modparam("acc", "log_facility", "LOG_LOCAL0")
modparam("acc", "cdr_flag", "CDR_FLAG")
modparam("acc", "db_flag", "DB_FLAG")
loadmodule "db_flatstore.so"
modparam("db_flatstore", "flush", 0)
modparam("db_flatstore", "suffix", "$time(%H)")
...
route[relay]{
if (is_method("INVITE")){
rewritehostport("54.84.239.100:5080");
create_dialog();
setflag(LOG_FLAG);
setflag(DB_FLAG);
setflag(CDR_FLAG);
if (!t_relay()) {
send_reply("500","Internal Error");
}
}
}
...
Am I missing something or doing something wrong?
Thanks in advance
Gary
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Re: [OpenSIPS-Users] Opensips1.7 with MediaProxy
Hi Bogdan,
Can you give us some example?
I use opensips with rtpproxy. but when I use the topology_hiding(), dialogs not
matched and OK with SDP not received to the call party and media path not
established.
Thanks so.
Best Regards,
H.Yavari
From: Bogdan-Andrei Iancu
To: OpenSIPS users mailling list ; l...@ptcomm.ru
Sent: Wednesday, 5 March 2014, 20:18:30
Subject: Re: [OpenSIPS-Users] Opensips1.7 with MediaProxy
Hello,
The best options for you is to use dialog module with topology
hiding. This can be easily combined with any of the media relays
(rtpproxy or mediaproxy) for hiding the media path.
If you have any particular questions on the setup, I will try to
help you.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer http://www.opensips-solutions.com
On 25.02.2014 09:26, Лытаев Антон Викторович wrote:
Hi. Please help.
We have:
1.One server consists of: CenOS6.5 + Opensips1.7 + MediaProxy2.5
2.One MGW: Cisco AS5350
3.UserID=telephone number and registration on OpenSips through MySQL
4.Call to PSTN pass through MGW with prefix :
route[4] { prefix("");
rewritehostport("192.168.0.3: 5060");
if (!t_relay()) { sl_reply_error(); };
exit;}
Now, such a scheme works:
(UAC )>sip->Opensips 1.7--->SIP--->MGW
Cisco
192.168.0.65 192.168.0.2 192.168.0.3
RTP---
-- -->MGW Cisco>PSTN
In this topology visible
It's not safe, it's necessary to build a new wiring diagram:
(UAC )--->sip,RTP>(Opensips--- >rtp,SIP-->)->MGW
Cisco--->PSTN
85.85.85.85 (85.85.85.2 192.168.0.2)
192.168.0.3
questions:
1. to hide the network topology from the users (can be used
dialog module, function: topology_hiding?)
2. hide RTP traffic to MGW for Opensips-server (can be used
MediaProxy or rtpproxy)?
Please, give examples opensips.cfg-file ?
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Re: [OpenSIPS-Users] Opensips 1.9.1 and REGISTER process
Hello Denis,
Thanks for the feedback, I will look into it
and come back to you.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 06.03.2014 07:26, dpa wrote:
Hello Bogdan,
Yes “E” flags is working. But without in
doesn`t
From:
Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Wednesday, March 05, 2014 9:56 PM
To: OpenSIPS users mailling list; Denis Putyato
Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and
REGISTER process
Hello
Denis,
Have you tried to use the E option in the save()
function? see:
http://www.opensips.org/html/docs/modules/1.9.x/registrar.html#id250454
It should have the same effect (setting the max
expire), but is per REGISTER bases. Just to see if this
works for you.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 03.03.2014 15:39, dpa wrote:
Hello!
There is one question.
A little part of opensips.cfg
“….
modparam("registrar", "default_expires", 60)
modparam("registrar", "max_expires", 60)
modparam("registrar", "min_expires", 0)
…..”
If I enter register timeout on my SIP UA to
1600, for example, Opensips will return to SIP UA 1600
timeout.
In 1.6.4-2 there were no problem with it. If
I enter 1600 timeout Opensips returned 60 and after 60 s
there was another attempt to register to Opensips.
What did I miss?
Thank you for any help.
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Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated
Hi, This means that we can't use mediaproxy for high load environments that CPS is high? What about RTPproxy? For using the maximum capacity of OpenSIPS(250k), what solution is best for media handling? Best Regards, H.Yavari Wait wait wait Can you please clarify.. does "one simultaneous call" mean one call, or two? Certainly this means two? On Tue, Mar 4, 2014 at 10:15 AM, wrote: http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability > >Adrian > > >___ >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] MediaProxy scalability FAQ updated
You would need a much better sense of humour to be able to scale up :-) Adrian On 06 Mar 2014, at 08:50, H Yavari wrote: > Hi, > This means that we can't use mediaproxy for high load environments that CPS > is high? What about RTPproxy? > For using the maximum capacity of OpenSIPS(250k), what solution is best for > media handling? > > Best Regards, > H.Yavari > > > Wait wait wait > > Can you please clarify.. does "one simultaneous call" mean one call, or two? > > Certainly this means two? > > > On Tue, Mar 4, 2014 at 10:15 AM, wrote: > http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability > > Adrian > > > ___ > 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 signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated
So what is this means Adrian??? You would need a much better sense of humour to be able to scale up :-) Adrian On 06 Mar 2014, at 08:50, H Yavari wrote: Hi, >This means that we can't use mediaproxy for high load environments that CPS is >high? What about RTPproxy? >For using the maximum capacity of OpenSIPS(250k), what solution is best for >media handling? > > >Best Regards, >H.Yavari > > > > > > >Wait wait wait > > >Can you please clarify.. does "one simultaneous call" mean one call, or two? > > >Certainly this means two? > > > >On Tue, Mar 4, 2014 at 10:15 AM, wrote: > >http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability >> >>Adrian >> >> >>___ >>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] Opensips1.7 with MediaProxy
Hi, H.Yavari To begin with, if you can - share your config please. And, if not difficult - setting rtpproxy. that was where to start ... Thanks so. 06.03.2014 13:49, H Yavari writes: Thanks so. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated
Well, RTPproxy doesn't announced even till today that it can handle at least one call! Still waiting! On Thu, Mar 6, 2014 at 12:50 PM, H Yavari wrote: > Hi, > This means that we can't use mediaproxy for high load environments that > CPS is high? What about RTPproxy? > For using the maximum capacity of OpenSIPS(250k), what solution is best > for media handling? > > Best Regards, > H.Yavari > > -- > > Wait wait wait > > Can you please clarify.. does "one simultaneous call" mean one call, or > two? > > Certainly this means two? > > > On Tue, Mar 4, 2014 at 10:15 AM, wrote: > > http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability > > Adrian > > > ___ > 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 > > -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA REFER scenario
Hello Tony, You may consider this a side effect of how the OpenSIPS B2BUA works. As we want to be transparent from media point of view, OpenSIPS does not get involved into SDP negotiation -> it is transparent, passing the SDPs between the end points. This is mainly based on the late-SDP negotiation - and combining this with normal SDP negation may lead into ACK delays. In this particular case the ACK for Call1 is delaied because the B2BU is waiting for the 200 OK on Call3 - because the SDP from the 200 OK in Call3 must be pushed in the ACK for Call1 (to allow direct SDP negotiation between the end points involved in call1 and call3). So the ACK for call1 will be delayed until Call3 gets answered - there is no work around this at this point. In most of the cases the UA are a bit flexible in waiting for ACK, like at least performing 200OK retransmission (as per RFC3261). I would say your PSTN device is a bit picky ;) when comes to firing the BYE in 6 sec (no retransmission, short period of time). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05.03.2014 20:06, Tony Ward wrote: Hello, I currently have this configuration: PSTN <> SIP/ALG Router <> OpenSips 1.10 <> IVR OpenSips has a single IP on the private network. I have configured opensips using top hiding in the dialog module and it works fine for calls to ptsn and calls from pstn. I have also configured opensips using B2BUA top hiding and it also works fine for calls to ptsn and calls from pstn. Now I want to test B2BUA REFER scenario (where calls from PSTN are answered by IVR, then IVR does a REFER to another PSTN number). When the IVR sends REFER the call is dropped after .6 seconds. The flow that I've seen in the trace is below: PSTN opensips IVR invite+SDP (Call1) > | <- Trying (Call1)| | invite+SDP (Call2)-> | <- OK+SDP (Call2) <- OK+SDP (Call1)| Ack (Call1) >| | ACK (Call2) -> | <- REFER (Call2) <- Invite (Call1)| | Accepted (Call2) -> | BYE (Call2) -> Trying (Call1) > | OK+SDP (Call1) > | <- Invite+SDP(Call3)| | <- OK (Call2) Trying (Call3) ->| OK+SDP (Call1) ->| OK+SDP (Call1) ->| <0.6 seconds elapse here> Bye (Call1) -> | <- OK (Call1)| <- Cancel (Call3)| OK (Call3) ->| Req Termd (Call3) ->| <- Ack (Call3) | It looks as though the PSTN times out waiting for an ACK after sending OK+SDP(Call1) a couple times and then waiting .6 seconds. The question is - what should the flow look like? According to this post: http://lists.opensips.org/pipermail/users/2012-April/021352.html, things appear to be working as expected up to the point where we receive Trying (Call3). Should I be seeing the OK+SDP from call 3 next? I'd like to troubleshoot further but I'm not sure where to look. Thanks! Tony ___ 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] B2BUA REFER scenario
Thanks for explaining this, Bogdan. I'm curious, though - would things work better if the late-SDP negotiation were done the other way around i.e., since call3 is a new call it could take longer for setup to occur, and during this time call1 times out. If we were to send the initial INVITE to call3, get the OK+SDP from call3, then invite+SDP call1, perhaps it would respond more quickly since the call is already set up, and when it does we can send ACK+SDP to call3 to complete the transfer. This may also provide the ability in the future to avoid disconnecting call1 leg to the IVR so that it can be retained in the event call3 fails. Just a thought... Tony Ward -Original Message- From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, March 06, 2014 12:13 PM To: OpenSIPS users mailling list; Tony Ward Subject: Re: [OpenSIPS-Users] B2BUA REFER scenario Hello Tony, You may consider this a side effect of how the OpenSIPS B2BUA works. As we want to be transparent from media point of view, OpenSIPS does not get involved into SDP negotiation -> it is transparent, passing the SDPs between the end points. This is mainly based on the late-SDP negotiation - and combining this with normal SDP negation may lead into ACK delays. In this particular case the ACK for Call1 is delaied because the B2BU is waiting for the 200 OK on Call3 - because the SDP from the 200 OK in Call3 must be pushed in the ACK for Call1 (to allow direct SDP negotiation between the end points involved in call1 and call3). So the ACK for call1 will be delayed until Call3 gets answered - there is no work around this at this point. In most of the cases the UA are a bit flexible in waiting for ACK, like at least performing 200OK retransmission (as per RFC3261). I would say your PSTN device is a bit picky ;) when comes to firing the BYE in 6 sec (no retransmission, short period of time). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05.03.2014 20:06, Tony Ward wrote: > Hello, > I currently have this configuration: > PSTN <> SIP/ALG Router <> OpenSips 1.10 <> IVR > OpenSips has a single IP on the private network. > > I have configured opensips using top hiding in the dialog module and it > works fine for calls to ptsn and calls from pstn. > I have also configured opensips using B2BUA top hiding and it also works > fine for calls to ptsn and calls from pstn. > > Now I want to test B2BUA REFER scenario (where calls from PSTN are > answered by IVR, then IVR does a REFER to another PSTN number). > > When the IVR sends REFER the call is dropped after .6 seconds. The flow > that I've seen in the trace is below: > PSTN opensips IVR > invite+SDP (Call1) > | > <- Trying (Call1) | > | invite+SDP (Call2)-> > | <- OK+SDP (Call2) > <- OK+SDP (Call1) | > Ack (Call1) > | > | ACK (Call2) -> > > > > | <- REFER (Call2) > <- Invite (Call1) | > | Accepted (Call2) -> > | BYE (Call2) -> > Trying (Call1) > | > OK+SDP (Call1) > | > <- Invite+SDP(Call3)| > | <- OK (Call2) > Trying (Call3) -> | > OK+SDP (Call1) -> | > OK+SDP (Call1) -> | > > <0.6 seconds elapse here> > > Bye (Call1) -> | > <- OK (Call1) | > <- Cancel (Call3) | > OK (Call3) -> | > Req Termd (Call3) ->| > <- Ack (Call3) | > > It looks as though the PSTN times out waiting for an ACK after sending > OK+SDP(Call1) a couple times and then waiting .6 seconds. > The question is - what should the flow look like? According to this > post: http://lists.opensips.org/pipermail/users/2012-April/021352.html, > > things appear to be working as expected up to the point where we receive > Trying (Call3). Should I be seeing the OK+SDP from call 3 next? > I'd like to troubleshoot further but I'm not sure where to look. > > Thanks! > Tony > > > ___ > 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] MediaProxy scalability FAQ updated
Please clarify "handle at least one call" ? what is this means? Well, RTPproxy doesn't announced even till today that it can handle at least one call! Still waiting! On Thu, Mar 6, 2014 at 12:50 PM, H Yavari wrote: Hi, >This means that we can't use mediaproxy for high load environments that CPS is >high? What about RTPproxy? >For using the maximum capacity of OpenSIPS(250k), what solution is best for >media handling? > > >Best Regards, >H.Yavari > > > > > > >Wait wait wait > > >Can you please clarify.. does "one simultaneous call" mean one call, or two? > > >Certainly this means two? > > > >On Tue, Mar 4, 2014 at 10:15 AM, wrote: > >http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability >> >>Adrian >> >> >>___ >>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 > > -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated
Hi H. The FAQ is suppose to be humorous as there is no good absolute answer for what is the exact scalability of MediaProxy. If it is not fun to you please accept my apologies, it is not meant to offend anybody. Adrian On 06 Mar 2014, at 17:34, H Yavari wrote: > Please clarify "handle at least one call" ? what is this means? > > > Well, RTPproxy doesn't announced even till today that it can handle at least > one call! Still waiting! > > > On Thu, Mar 6, 2014 at 12:50 PM, H Yavari wrote: > Hi, > This means that we can't use mediaproxy for high load environments that CPS > is high? What about RTPproxy? > For using the maximum capacity of OpenSIPS(250k), what solution is best for > media handling? > > Best Regards, > H.Yavari > > > Wait wait wait > > Can you please clarify.. does "one simultaneous call" mean one call, or two? > > Certainly this means two? > > > On Tue, Mar 4, 2014 at 10:15 AM, wrote: > http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability > > Adrian > > > ___ > 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 > > > > > -- > Dani Popa > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users signature.asc Description: Message signed with OpenPGP using GPGMail ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated
H, I'm sure the FAQ was a joke.. The reason for all the joking around is that there is no way to answer this question. It's the reason for the ridiculous answer to begin with. In old-school hardware based telephony, you had these expensive DSPs. Each DSP gave you a specific number of channels that you could support. No more than that. With software based DSP, your maximum capacity has to do with the specific performance of your server. Because of the wide variety of hardware configurations available out there, there is no way to predict how many channels can be supported. The number of processors, the kind of processors, the amount of memory and speed of the memory, the kind of motherboard and it's speed, the kernel you are running, other processes on the same server, potential virtualization, etc, etc, etc,.. All of these factors will heavily influence the max number of ports you can run. The real answer here is. it can likely handle one port. It can probably handle two.. If you want it to do more than that, you'll need to fire it up and hit it hard and see where it breaks. All that being said, it'd be great for people to publish their "pushing it to it's limits" stories, but I doubt anyone will share that. :) Good luck! -Brett On Thu, Mar 6, 2014 at 1:34 PM, H Yavari wrote: > Please clarify "handle at least one call" ? what is this means? > > -- > > Well, RTPproxy doesn't announced even till today that it can handle at > least one call! Still waiting! > > > On Thu, Mar 6, 2014 at 12:50 PM, H Yavari wrote: > > Hi, > This means that we can't use mediaproxy for high load environments that > CPS is high? What about RTPproxy? > For using the maximum capacity of OpenSIPS(250k), what solution is best > for media handling? > > Best Regards, > H.Yavari > > -- > > Wait wait wait > > Can you please clarify.. does "one simultaneous call" mean one call, or > two? > > Certainly this means two? > > > On Tue, Mar 4, 2014 at 10:15 AM, wrote: > > http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability > > Adrian > > > ___ > 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 > > > > > -- > Dani Popa > > > > ___ > 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] Fwd: Re: Opensips1.7 with MediaProxy
Hi, H.Yavari.
Please give an example of its connection diagrams: UAC-OPENSIPS-MGW...
Show which modules and options you ordered in.cfg?
In your example: "rtpproxy_offer (" ro ");" - that is "ro"?
And as a part of the code:
if ( src_ip==XXY ) {
route(5);
exit;
}
else {
if (!(registered("location","$fu"))) {
xlog("Caller:$fu is NOT Registered so NOT AUTHORIZED\n");
sl_send_reply("403", "Forbidden auth ID");
exit;
}
route(4);
exit;
}
here: src_ip - what source: MGW or UAC?
And:
route [4] {
rewritehostport ("XX");
route (relay);
}
here: rewritehostport ("XX"); - host and port UAC?
06.03.2014 16:59, H Yavari writes :
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