Re: [Freeswitch-users] "show channels" command with duration - patch included
Now that I come to think of it...
It would be useful if we had the timestamp (and epoch)
of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract
the PDD (Post Dial Delay) which is a very useful statistic.
Adding the user's (for the incoming) and the gateway's (for the
outbound) id would also be useful. In case these fields are empty the
show channels command could ommit the (empty string).
Are you planning to implement them yourselves or should I begin looking
at the code?
Anthony Minessale wrote:
> I'm ok with the idea as long as it's thoroughly tested.
> If there is any more info you want to save from those events you should
> consider it now while we are modifying it.
>
>
> On Thu, Jul 16, 2009 at 9:02 AM, <mailto:freeswitch-users@lists.freeswitch.org>> wrote:
>
> Hi,
>
> I usually find it very useful when I can retrieve a list of the
> currents calls along with durations. I noticed that the 'show
> channels' format does not include the duration (or the answered
> timestamp - so that one can extract it from there). So, I made a
> patch that includes the answered timestamp, the answered timestamp
> in epoch, and the duration in seconds. Of course these fields remain
> empty when the call hasn't been
> answered yet.
>
> I don't know if anyone else finds this functionality useful, so I am
> posting this patch here first (instead of JIRA) in order to get
> feedback from the users. If many of you (or the maintainers) find it
> interesting I can then proceed in posting it to JIRA.
>
> --
> ---
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: r...@kinetix.gr <mailto:r...@kinetix.gr>
> ---
>
> Index: src/mod/applications/mod_commands/mod_commands.c
> ===
> --- src/mod/applications/mod_commands/mod_commands.c(revision 14256)
> +++ src/mod/applications/mod_commands/mod_commands.c(working copy)
> @@ -2827,10 +2827,10 @@
>}
>}
>if (strchr(argv[2], '%')) {
> - sprintf(sql, "select * from channels
> where uuid like '%s' or name like '%s' or cid_name like '%s' or
> cid_num like '%s' order by created_epoch",
> + sprintf(sql, "select
> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from
> channels where uuid like '%s' or name like '%s' or cid_name like
> '%s' or cid_num like '%s' order by created_epoch",
>argv[2], argv[2],
> argv[2], argv[2]);
>} else {
> - sprintf(sql, "select * from channels
> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like
> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch",
> + sprintf(sql, "select
> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from
> channels where uuid like '%%%s%%' or name like '%%%s%%' or cid_name
> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch",
>argv[2], argv[2],
> argv[2], argv[2]);
>
>}
> @@ -2839,10 +2839,10 @@
>as = argv[4];
>}
>} else {
> - sprintf(sql, "select * from channels order
> by created_epoch");
> + sprintf(sql, "select
> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from
> channels order by created_epoch");
>}
>} else if (!strcasecmp(command, "channels")) {
> - sprintf(sql, "select * from channels order by
> created_epoch");
> + sprintf(sql, "select
> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from
> channels order by created_epoch");
>if (argv[1] && !strcasecmp(argv[1],"count")) {
>holder.justcount = 1;
>if (a
Re: [Freeswitch-users] "show channels" command with duration - patch included
OK I 'll start implementing the progress timestamp field.
The only reason I mentioned the user/gateway id field is that FS admins
gain a lot by looking at a row of "show channels" result and be able to
see who is the caller and who is the callee.
Anthony Minessale wrote:
> it doesn't really break anything to add more fields besides the ability
> to read it but it's already fairly wide as it is.
> Thats why we have "show channels as xml"
>
>
> On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins <mailto:m...@freeswitch.org>> wrote:
>
> I wonder if it would make sense to create a separate sub-command like
> "show channels stats" or something. That way we could put all sorts of
> nifty info there without breaking the existing command.
>
> Thoughts?
> -MC
>
> Sent from my iPhone
>
> On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos
> mailto:r...@kinetix.gr>>
> wrote:
>
> > Now that I come to think of it...
> >
> > It would be useful if we had the timestamp (and epoch)
> > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract
> > the PDD (Post Dial Delay) which is a very useful statistic.
> >
> > Adding the user's (for the incoming) and the gateway's (for the
> > outbound) id would also be useful. In case these fields are empty the
> > show channels command could ommit the (empty string).
> >
> > Are you planning to implement them yourselves or should I begin
> > looking
> > at the code?
> >
> >
> > Anthony Minessale wrote:
> >> I'm ok with the idea as long as it's thoroughly tested.
> >> If there is any more info you want to save from those events you
> >> should
> >> consider it now while we are modifying it.
> >>
> >>
> >> On Thu, Jul 16, 2009 at 9:02 AM,
> <mailto:freeswitch-users@lists.freeswitch.org>
> >> <mailto:freeswitch-users@lists.freeswitch.org
> <mailto:freeswitch-users@lists.freeswitch.org>>> wrote:
> >>
> >>Hi,
> >>
> >>I usually find it very useful when I can retrieve a list of the
> >>currents calls along with durations. I noticed that the 'show
> >>channels' format does not include the duration (or the answered
> >>timestamp - so that one can extract it from there). So, I made a
> >>patch that includes the answered timestamp, the answered
> timestamp
> >>in epoch, and the duration in seconds. Of course these fields
> >> remain
> >>empty when the call hasn't been
> >> answered yet.
> >>
> >>I don't know if anyone else finds this functionality useful, so
> >> I am
> >>posting this patch here first (instead of JIRA) in order to get
> >>feedback from the users. If many of you (or the maintainers)
> >> find it
> >>interesting I can then proceed in posting it to JIRA.
> >>
> >>--
> >>---
> >>Apostolos Pantsiopoulos
> >>Kinetix Tele.com R & D
> >>email: r...@kinetix.gr <mailto:r...@kinetix.gr>
> <mailto:r...@kinetix.gr <mailto:r...@kinetix.gr>>
> >>---
> >>
> >>Index: src/mod/applications/mod_commands/mod_commands.c
> >>
> >> ===
> >>--- src/mod/applications/mod_commands/mod_commands.c
> >> (revision 14256)
> >>+++ src/mod/applications/mod_commands/mod_commands.c(working
> >> copy)
> >>@@ -2827,10 +2827,10 @@
> >> }
> >> }
> >> if (strchr(argv[2], '%')) {
> >>- sprintf(sql, "select * from
> >> channels
> >>where uuid like '%s' or name like '%s' or cid_name like '%s' or
> >>cid_num like '%s' order by created_epoch",
> >>+ sprintf(sql, "select
> >>
Re: [Freeswitch-users] "show channels" command with duration - patch included
On a second thought... I am starting to agree with Michael. Nobody really collects stats about PDD from the 'show channels' command. This is usually done by using the cdrs. So, forget about the second request. The patch I sent earlier covers my needs (and I hope everybody else's too.) Apostolos Pantsiopoulos wrote: > OK I 'll start implementing the progress timestamp field. > > The only reason I mentioned the user/gateway id field is that FS admins > gain a lot by looking at a row of "show channels" result and be able to > see who is the caller and who is the callee. > > Anthony Minessale wrote: >> it doesn't really break anything to add more fields besides the ability >> to read it but it's already fairly wide as it is. >> Thats why we have "show channels as xml" >> >> >> On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins > <mailto:m...@freeswitch.org>> wrote: >> >> I wonder if it would make sense to create a separate sub-command like >> "show channels stats" or something. That way we could put all sorts of >> nifty info there without breaking the existing command. >> >> Thoughts? >> -MC >> >> Sent from my iPhone >> >> On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos >> mailto:r...@kinetix.gr>> >> wrote: >> >> > Now that I come to think of it... >> > >> > It would be useful if we had the timestamp (and epoch) >> > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract >> > the PDD (Post Dial Delay) which is a very useful statistic. >> > >> > Adding the user's (for the incoming) and the gateway's (for the >> > outbound) id would also be useful. In case these fields are empty the >> > show channels command could ommit the (empty string). >> > >> > Are you planning to implement them yourselves or should I begin >> > looking >> > at the code? >> > >> > >> > Anthony Minessale wrote: >> >> I'm ok with the idea as long as it's thoroughly tested. >> >> If there is any more info you want to save from those events you >> >> should >> >> consider it now while we are modifying it. >> >> >> >> >> >> On Thu, Jul 16, 2009 at 9:02 AM, >> > <mailto:freeswitch-users@lists.freeswitch.org> >> >> <mailto:freeswitch-users@lists.freeswitch.org >> <mailto:freeswitch-users@lists.freeswitch.org>>> wrote: >> >> >> >>Hi, >> >> >> >>I usually find it very useful when I can retrieve a list of the >> >>currents calls along with durations. I noticed that the 'show >> >>channels' format does not include the duration (or the answered >> >>timestamp - so that one can extract it from there). So, I made a >> >>patch that includes the answered timestamp, the answered >> timestamp >> >>in epoch, and the duration in seconds. Of course these fields >> >> remain >> >>empty when the call hasn't been >> >>answered yet. >> >> >> >>I don't know if anyone else finds this functionality useful, so >> >> I am >> >>posting this patch here first (instead of JIRA) in order to get >> >>feedback from the users. If many of you (or the maintainers) >> >> find it >> >>interesting I can then proceed in posting it to JIRA. >> >> >> >>-- >> >>--- >> >>Apostolos Pantsiopoulos >> >>Kinetix Tele.com R & D >> >>email: r...@kinetix.gr <mailto:r...@kinetix.gr> >> <mailto:r...@kinetix.gr <mailto:r...@kinetix.gr>> >> >>--- >> >> >> >>Index: src/mod/applications/mod_commands/mod_commands.c >> >> >> >> === >> >>--- src/mod/applications/mod_commands/mod_commands.c >> >> (revision 14256) >> >>+++ src/mod/applications/mod_commands/mod_commands
Re: [Freeswitch-users] Possible memory leak - need a second opinion
Since nobody replied I am posting it to JIRA. Apostolos Pantsiopoulos wrote: > Hi I noticed that after a day of relatively moderate traffic (about 400 > simultaneous channels average) the memory used by FS reached 1.3 GB of > RAM. I tried to trace the leak (if any) with valgrind and got that : > > ==18894== 572 bytes in 1 blocks are definitely lost in loss record 148 > of 193 > ==18894==at 0x40053C0: malloc (vg_replace_malloc.c:149) > ==18894==by 0x7AFDE93: sofia_glue_do_invite (sofia_glue.c:1599) > ==18894==by 0x7ACB191: sofia_on_init (mod_sofia.c:102) > ==18894==by 0x405B124: switch_core_session_run > (switch_core_state_machine.c:480) > ==18894==by 0x4058204: switch_core_session_thread > (switch_core_session.c:1064) > ==18894==by 0x40CD6C5: dummy_worker (thread.c:138) > ==18894==by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) > ==18894==by 0xAAADBD: clone (in /lib/libc-2.5.so) > > > So, I assummed that this happens for every call. I tried testing it > again by placing two calls before shutting down FS, but it only came up > once. I wanted to get a second opinion before posting this to JIRA as an > issue. > > I used revision 14269 of the SVN. I am attaching the valgrind output as > well. > > I also noticed that only one of my CPU cores gets really busy when > dealing with moderate traffic. From what I read in the mailing list > users are encouraged to use 64bit multi core servers for FS because it > scales up better. But this is not what I am seeing in practice. Could > the single threaded architecture of libsofia be the cause of that behavior? > > > > > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] multiple commands in execute_on_answer
Hi, Is there a way to execute more than 1 commands in the execute_on_answer variable? I want to execute both a python script AND the sched_hangup application. -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] multiple commands in execute_on_answer
Yes, I could do that but the logic of the python script is not dialplan oriented and I hate mixing different things. I just found a workaround though. I am executing the python script on the b leg and the sched_hangup on the a leg (with the help of nolocal:) Thanx for the help anyway. Matthew Fong wrote: > can't you just have your python script execute the sched_hangup command, > and then finish the remainder of the python script? > > On Wed, Jul 29, 2009 at 12:15 AM, Apostolos Pantsiopoulos > mailto:r...@kinetix.gr>> wrote: > > Hi, > > Is there a way to execute more than 1 commands in the execute_on_answer > variable? I want to execute both a python script AND the sched_hangup > application. > > > -- > ------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: r...@kinetix.gr <mailto:r...@kinetix.gr> > --- > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Rewriting SDP with switch_r_sdp
I am trying to use switch_r_sdp to rewrite the SDP. The problem I am facing has to do with the way of doing it. Let's say I have: v=0 o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx s=Opal SIP Session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 5144 RTP/AVP 18 3 101 120 c=IN IP4 xxx.xxx.xxx.xxx a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:3 gsm/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:120 NSE/8000 a=fmtp:120 192-193 who to I set the switch_r_sdp variable in xml? Obviously this doesn't work : Do I have to escape any special characters and how? I tried using escaped quotes, escaped spaces, escaped tabs etc. Nothing worked. Any suggestions? -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Rewriting SDP with switch_r_sdp
Thanx! That worked fine. Anthony Minessale wrote: > I should have said > > > > > > > On Tue, Nov 17, 2009 at 2:15 PM, Anthony Minessale > mailto:anthony.miness...@gmail.com>> wrote: > > you can do > > > > > > > On Tue, Nov 17, 2009 at 1:44 PM, r...@kinetix.gr > <mailto:r...@kinetix.gr> mailto:r...@kinetix.gr>> > wrote: > > I am trying to achieve something similar to that : > > http://wiki.freeswitch.org/wiki/Codec_negotiation#Modifying_the_codec_when_using_proxy_media_mode > > but I am using xml_curl to create the dialplan (i.e. the web > server that > serves the dialplan makes the decision about the SDP). So I need > a way > to write > the new SDP in the XML dialplan response. However, in the above > example > due to the regex manipulation the user is not facing the > problem that I am > with setting the switch_r_sdp to a complex value that contains =, > spaces, new lines etc. > > Brian West wrote: > > Why are you needing to rewrite it? > > > > /b > > > > On Nov 17, 2009, at 7:42 AM, Apostolos Pantsiopoulos wrote: > > > > > >> I am trying to use switch_r_sdp to rewrite the SDP. > >> The problem I am facing has to do with the way of doing it. > >> > >> Let's say I have: > >> > >> v=0 > >> o=- 1258463684 1258463684 IN IP4 xxx.xxx.xxx.xxx > >> s=Opal SIP Session > >> c=IN IP4 xxx.xxx.xxx.xxx > >> t=0 0 > >> m=audio 5144 RTP/AVP 18 3 101 120 > >> c=IN IP4 xxx.xxx.xxx.xxx > >> a=rtpmap:18 G729/8000/1 > >> a=fmtp:18 annexb=no > >> a=rtpmap:3 gsm/8000/1 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-16,32,36 > >> a=rtpmap:120 NSE/8000 > >> a=fmtp:120 192-193 > >> > >> who to I set the switch_r_sdp variable in xml? > >> > >> Obviously this doesn't work : > >> > >> > >> > >> Do I have to escape any special characters and how? > >> I tried using escaped quotes, escaped spaces, escaped tabs etc. > >> Nothing worked. > >> > >> Any suggestions? > >> > >> > >> > >> > >> -- > >> --- > >> Apostolos Pantsiopoulos > >> Kinetix Tele.com R & D > >> email: r...@kinetix.gr <mailto:r...@kinetix.gr> > >> --- > >> > >> ___ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > > > > > > ___ > > FreeSWITCH-users mailing list > > FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter:
[Freeswitch-users] mod_opal compile error
Hi,
I have installed ptlib 2.4.3 and opal 3.4.3 and still cannot get
freeswitch to
compile mod_opal. The error I am getting is this :
mod_opal.h:58: error: expected class-name before β{β token
I am using the 11094 revision of FS on CentOS 5.2.
Has anyone faced anything similar?
--
---
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: r...@kinetix.gr
---
___
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[Freeswitch-users] mod_opal first unsuccessful test
Hi, I successfully compiled mod_opal using the latest svn for both opal and ptlib as Brian suggested. When I try to establish a call using h323 from my openphone client I get no audio although I can see RTP packets in both directions when I am doing a capture. Some details : I am using the 11094 revision of the FS trunk. I am using the PCMU codec. I tried dialing the default IVR (5000) and other testing extensions (freeswitch conference, echo test etc.) I tried using fast start on and off , h245 tunneling on and off, h245 in SETUP on and off. In my captures I have also noticed a strange behavior : FS sends to my client 2 "alerting" packets for no apparent reason. Could this be a cause of the problem? Had anyone any success with mod_opal lately? If yes, could you please reply quoting your config options (both on FS and on your client)? -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_opal first unsuccessful test
Hi, Yes, openPhone is working with my soundcard. I am using it every day for testing purposes. I use the 1.8.1 version. Is there a newer version that uses OPAL? I didn't know that. Where can I get it from? Robert Jongbloed wrote: Hi guys, I was using the OpenPhone that you build with OPAL for my testing. So that is identical (I think) to you. I have not (yet) do any third party client testing. Two ALERTING messages are fine, perfectly legal and OPAL can handle it. You say you can see the RTP packets flowing so that implies that the mod_opal is actually working, so let's look somewhere else. Have you confirmed that OpenPhone is using the sound card correctly? Made a call between two machines JUST using OpenPhone for example? Robert Jongbloed OPAL/OpenH323/PTLib Architect and Co-founder. *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com] *Sent:* Tuesday, 13 January 2009 6:20 AM *To:* Robert Jongbloed *Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test Heh, what client are you using in your tests that are working? -- Forwarded message -- From: *Apostolos Pantsiopoulos* mailto:r...@kinetix.gr>> Date: Mon, Jan 12, 2009 at 9:51 AM Subject: [Freeswitch-users] mod_opal first unsuccessful test To: freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> Hi, I successfully compiled mod_opal using the latest svn for both opal and ptlib as Brian suggested. When I try to establish a call using h323 from my openphone client I get no audio although I can see RTP packets in both directions when I am doing a capture. Some details : I am using the 11094 revision of the FS trunk. I am using the PCMU codec. I tried dialing the default IVR (5000) and other testing extensions (freeswitch conference, echo test etc.) I tried using fast start on and off , h245 tunneling on and off, h245 in SETUP on and off. In my captures I have also noticed a strange behavior : FS sends to my client 2 "alerting" packets for no apparent reason. Could this be a cause of the problem? Had anyone any success with mod_opal lately? If yes, could you please reply quoting your config options (both on FS and on your client)? -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr <mailto:r...@kinetix.gr> --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com <mailto:msn%3aanthony_miness...@hotmail.com> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com <mailto:paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <mailto:sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 <http://iax:gu...@conference.freeswitch.org/888> googletalk:conf+...@conference.freeswitch.org <mailto:googletalk%3aconf%2b...@conference.freeswitch.org> pstn:213-799-1400 -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_opal first unsuccessful test
I also tried using Ekiga - which is OPAL based - and got the same behavior. No audio - although I can see RTP packets. Apostolos Pantsiopoulos wrote: Hi, Yes, openPhone is working with my soundcard. I am using it every day for testing purposes. I use the 1.8.1 version. Is there a newer version that uses OPAL? I didn't know that. Where can I get it from? Robert Jongbloed wrote: Hi guys, I was using the OpenPhone that you build with OPAL for my testing. So that is identical (I think) to you. I have not (yet) do any third party client testing. Two ALERTING messages are fine, perfectly legal and OPAL can handle it. You say you can see the RTP packets flowing so that implies that the mod_opal is actually working, so let's look somewhere else. Have you confirmed that OpenPhone is using the sound card correctly? Made a call between two machines JUST using OpenPhone for example? Robert Jongbloed OPAL/OpenH323/PTLib Architect and Co-founder. *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com] *Sent:* Tuesday, 13 January 2009 6:20 AM *To:* Robert Jongbloed *Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test Heh, what client are you using in your tests that are working? -- Forwarded message -- From: *Apostolos Pantsiopoulos* <mailto:r...@kinetix.gr>> Date: Mon, Jan 12, 2009 at 9:51 AM Subject: [Freeswitch-users] mod_opal first unsuccessful test To: freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> Hi, I successfully compiled mod_opal using the latest svn for both opal and ptlib as Brian suggested. When I try to establish a call using h323 from my openphone client I get no audio although I can see RTP packets in both directions when I am doing a capture. Some details : I am using the 11094 revision of the FS trunk. I am using the PCMU codec. I tried dialing the default IVR (5000) and other testing extensions (freeswitch conference, echo test etc.) I tried using fast start on and off , h245 tunneling on and off, h245 in SETUP on and off. In my captures I have also noticed a strange behavior : FS sends to my client 2 "alerting" packets for no apparent reason. Could this be a cause of the problem? Had anyone any success with mod_opal lately? If yes, could you please reply quoting your config options (both on FS and on your client)? -- ------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr <mailto:r...@kinetix.gr> --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com <mailto:msn%3aanthony_miness...@hotmail.com> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com <mailto:paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <mailto:sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 <http://iax:gu...@conference.freeswitch.org/888> googletalk:conf+...@conference.freeswitch.org <mailto:googletalk%3aconf%2b...@conference.freeswitch.org> pstn:213-799-1400 -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Gateways vs Directory Users
I think one of the most difficult ideas to grasp when someone migrates to FS from other soft-switches is the definition of endpoints (in the generic meaning of the word) i.e. other gateways that your FS gateway will exchange traffic with. From what I understood from the wiki : a user (directory) is an entitiy that defines our subscribers (with registration or not) a gateway is another VoIP gateway that we use to send traffic to (provider) and sometimes receive traffic from (correct me If I am wrong this far) Now, I have the following scenario : I have an FS box that will exhchange traffic with certain gateways. Sometimes a gateway will act as a client (customer) or a provider, depending on the traffic direction. 1st approach : The first plan I tried, was to use the directory to "define" these other gateways since they are clients. I turned off registrations in my SIP profile (because I don't need any), I wrote an xml file for each gateway in the directory with the appropriate cidr and that was it. I then realized that I could not use my definitions in the users directory so that I can send a call to that client. In order to do so, I had to define that particular IP as a gateway in my SIP profile section. 2nd approach : I forgot all about the directory thing (after all I needed not any mailboxes or registrations) and tried to implement the same scenario by just using gateways. I declared the gateways in my SIP profile and was able to dial out to them just by using : which was fine. But when I got an inbound call from that gateway I realized (1) that I had to define this gateway in an acl and (2) the dialplan used (context) was always the inbound one since there is no way of telling a gateway which context to use when accepting a call. I found a way in the wiki that suggests using DIDs to accomplish that, but I find it this to be not a very elegant solution because I don't want to use prefixes or DIDs to accomplish my goal. I want to be able to define a gateway using a gateway name (label) of my choice and an IP address, so when I come to the point of having 100+ gateways i won't have to search wich DID to use for a new one. 3rd approach : Use a combination of the above. Declare a gateway that receives and sends traffic twice. Once as a user in the directory and once in the gateways section of my profile. Does all these sound correct to you? Am I missing something? Which approach is the best? Please send me your feedback. Inline comments would be ideal. -- ------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Gateways vs Directory Users
In the http://wiki.freeswitch.org/wiki/Clarification:gateways we can read : "What I gather from this is that if you only want certain extensions to be registered with your voip provider when a specific user registers with freeswitch you should define gateways in the directory section rather than in the sofia configuration. Conversely, if you always want an extension registered with a provider you would define the gateway as part of the sip profile." So If I understand the above statement correctly the gateways that are defined within a user tag are not necessarily the gateways of the specific user (i.e. he/she physically owns them) , but gateways that our FS may use to terminate calls of this specific user. What I also understand from the above statement is that these gateways get registered (or are generally accessible) only when that particular user registers to our FS. Is this correct? Anthony Minessale wrote: What you are doing is correct. In FS gateways only represent a set of credentials that the profile uses to authenticate to another sip UA The actual UA is the sofia profile and the gateways are assumed identities to deal with challenges. Also to manage everything from the same place, you can put the tag inside the tag in the directory then tell sofia to parse the whole domain for gateways in the profile in section of your sofia profile add http://domain.com>" parse="true"/> this will tell FS to open that domain in the user directory and check every user for and parse them all. On Tue, Jan 13, 2009 at 10:45 AM, Apostolos Pantsiopoulos mailto:r...@kinetix.gr>> wrote: I think one of the most difficult ideas to grasp when someone migrates to FS from other soft-switches is the definition of endpoints (in the generic meaning of the word) i.e. other gateways that your FS gateway will exchange traffic with. From what I understood from the wiki : a user (directory) is an entitiy that defines our subscribers (with registration or not) a gateway is another VoIP gateway that we use to send traffic to (provider) and sometimes receive traffic from (correct me If I am wrong this far) Now, I have the following scenario : I have an FS box that will exhchange traffic with certain gateways. Sometimes a gateway will act as a client (customer) or a provider, depending on the traffic direction. 1st approach : The first plan I tried, was to use the directory to "define" these other gateways since they are clients. I turned off registrations in my SIP profile (because I don't need any), I wrote an xml file for each gateway in the directory with the appropriate cidr and that was it. I then realized that I could not use my definitions in the users directory so that I can send a call to that client. In order to do so, I had to define that particular IP as a gateway in my SIP profile section. 2nd approach : I forgot all about the directory thing (after all I needed not any mailboxes or registrations) and tried to implement the same scenario by just using gateways. I declared the gateways in my SIP profile and was able to dial out to them just by using : which was fine. But when I got an inbound call from that gateway I realized (1) that I had to define this gateway in an acl and (2) the dialplan used (context) was always the inbound one since there is no way of telling a gateway which context to use when accepting a call. I found a way in the wiki that suggests using DIDs to accomplish that, but I find it this to be not a very elegant solution because I don't want to use prefixes or DIDs to accomplish my goal. I want to be able to define a gateway using a gateway name (label) of my choice and an IP address, so when I come to the point of having 100+ gateways i won't have to search wich DID to use for a new one. 3rd approach : Use a combination of the above. Declare a gateway that receives and sends traffic twice. Once as a user in the directory and once in the gateways section of my profile. Does all these sound correct to you? Am I missing something? Which approach is the best? Please send me your feedback. Inline comments would be ideal. -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr <mailto:r...@kinetix.gr> --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] ipauth - directory
I noticed an "ip=" setting in the brian.xml sample file. The comments state that this is used for ipauth (IP based authentication?) What exactly is this setting. I cannot find anything in the wiki about it. Does it replace the use of the + ACL mechanism for IP authentication? -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ipauth - directory
Yes I know that. But what does the "ip=" setting do? Brian West wrote: cidr= and the domains acl in acl.conf.xml then apply that ACL to the sofia profile. /b On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: I noticed an "ip=" setting in the brian.xml sample file. The comments state that this is used for ipauth (IP based authentication?) What exactly is this setting. I cannot find anything in the wiki about it. Does it replace the use of the + ACL mechanism for IP authentication? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_opal first unsuccessful test
Any news regarding this issue? Apostolos Pantsiopoulos wrote: I am attaching the wireshark capture. Openphone is on xxx.xxx.xxx.202 and FS is on xxx.xxx.xxx.212 Robert Jongbloed wrote: Can you send me a WireShark capture? Robert Jongbloed OPAL/OpenH323/PTLib Architect and Co-founder. *From:* Apostolos Pantsiopoulos [mailto:r...@kinetix.gr] *Sent:* Tuesday, 13 January 2009 11:05 PM *To:* freeswitch-users@lists.freeswitch.org *Cc:* Robert Jongbloed *Subject:* Re: [Freeswitch-users] mod_opal first unsuccessful test I also tried using Ekiga - which is OPAL based - and got the same behavior. No audio - although I can see RTP packets. Apostolos Pantsiopoulos wrote: Hi, Yes, openPhone is working with my soundcard. I am using it every day for testing purposes. I use the 1.8.1 version. Is there a newer version that uses OPAL? I didn't know that. Where can I get it from? Robert Jongbloed wrote: Hi guys, I was using the OpenPhone that you build with OPAL for my testing. So that is identical (I think) to you. I have not (yet) do any third party client testing. Two ALERTING messages are fine, perfectly legal and OPAL can handle it. You say you can see the RTP packets flowing so that implies that the mod_opal is actually working, so let's look somewhere else. Have you confirmed that OpenPhone is using the sound card correctly? Made a call between two machines JUST using OpenPhone for example? Robert Jongbloed OPAL/OpenH323/PTLib Architect and Co-founder. *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com] *Sent:* Tuesday, 13 January 2009 6:20 AM *To:* Robert Jongbloed *Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test Heh, what client are you using in your tests that are working? -- Forwarded message -- From: *Apostolos Pantsiopoulos* <mailto:r...@kinetix.gr>> Date: Mon, Jan 12, 2009 at 9:51 AM Subject: [Freeswitch-users] mod_opal first unsuccessful test To: freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> Hi, I successfully compiled mod_opal using the latest svn for both opal and ptlib as Brian suggested. When I try to establish a call using h323 from my openphone client I get no audio although I can see RTP packets in both directions when I am doing a capture. Some details : I am using the 11094 revision of the FS trunk. I am using the PCMU codec. I tried dialing the default IVR (5000) and other testing extensions (freeswitch conference, echo test etc.) I tried using fast start on and off , h245 tunneling on and off, h245 in SETUP on and off. In my captures I have also noticed a strange behavior : FS sends to my client 2 "alerting" packets for no apparent reason. Could this be a cause of the problem? Had anyone any success with mod_opal lately? If yes, could you please reply quoting your config options (both on FS and on your client)? -- ------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr <mailto:r...@kinetix.gr> --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com <mailto:msn%3aanthony_miness...@hotmail.com> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com <mailto:paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <mailto:sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 <http://iax:gu...@conference.freeswitch.org/888> googletalk:conf+...@conference.freeswitch.org <mailto:googletalk%3aconf%2b...@conference.freeswitch.org> pstn:213-799-1400 -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr <mailto:r...@kinetix.gr> --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --
[Freeswitch-users] Outbound call - choose profile
When I am using the following method to place a call from the dialplan : sofia/gateway// how do I tell FS which profile to use (as in the sofia// method?) I am asking that because all my calls to my declared use the 5080 port, and I want them to use the 5060 port. Is there a way to configure a to use a specific profile when making outbound calls? -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Outbound call - choose profile
Raymond Chandler wrote: > Apostolos Pantsiopoulos wrote: >> When I am using the following method to place a call from the dialplan : >> >> sofia/gateway// >> >> how do I tell FS which profile to use (as in the >> sofia// method?) >> >> I am asking that because all my calls to my declared use >> the 5080 port, >> and I want them to use the 5060 port. Is there a way to configure a >> to use >> a specific profile when making outbound calls? >> >> > the gateway should always use the profile from which it was included. > so, for instance, if you include gw1 from internal.xml, then gw1 > should always use the internal profile > > -Ray > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > I am using the directory to define my gateways. Specifically, because this certain gateway is related to a specific user I am including this gateway in the user's xml file (in the dieractory). Then I use : in my internal profile to let FS "parse" the gateways that I have declared in the directory. So one would expect that the gateways declared in the user's file would belong to the internal profile. Yet, when I am using the sofia/gateway// notation to send a call through this gateway the SIP packets get send from the 5080 port (which is my external profile's port) -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Outbound call - choose profile
Yes, I am aware of that fact : insteaad of the pre-configured internal/external profiles I coulf have myprofile1/myprofile2. But since I am using the pre-configured profiles (without changing the ports) and since the internal profile initiated the parsing I would expect that all the gateways defined in my directory user's xml files would belong to the internal profile (which is preconfigured to use port 5060). But when I am capturing the packets sent to that gateway I can see that the port used from FS is 5080 (which is the preconfigured port of the internal profile). Anthony Minessale wrote: all gateways are properties of a profile so even in the parse domain way of handling gateways whatever profile parsed the domain will be the owner of all the gateways discovered from the domain. FYI, The internal profile has nothing to do with users or anything else. You are confusing the default examples for absolute fact. On Fri, Jan 16, 2009 at 8:53 AM, Apostolos Pantsiopoulos mailto:r...@kinetix.gr>> wrote: Raymond Chandler wrote: > Apostolos Pantsiopoulos wrote: >> When I am using the following method to place a call from the dialplan : >> >> sofia/gateway// >> >> how do I tell FS which profile to use (as in the >> sofia// method?) >> >> I am asking that because all my calls to my declared use >> the 5080 port, >> and I want them to use the 5060 port. Is there a way to configure a >> to use >> a specific profile when making outbound calls? >> >> > the gateway should always use the profile from which it was included. > so, for instance, if you include gw1 from internal.xml, then gw1 > should always use the internal profile > > -Ray > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > I am using the directory to define my gateways. Specifically, because this certain gateway is related to a specific user I am including this gateway in the user's xml file (in the dieractory). Then I use : in my internal profile to let FS "parse" the gateways that I have declared in the directory. So one would expect that the gateways declared in the user's file would belong to the internal profile. Yet, when I am using the sofia/gateway// notation to send a call through this gateway the SIP packets get send from the 5080 port (which is my external profile's port) -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr <mailto:r...@kinetix.gr> --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com <mailto:msn%3aanthony_miness...@hotmail.com> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com <mailto:paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <mailto:sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 <http://iax:gu...@conference.freeswitch.org/888> googletalk:conf+...@conference.freeswitch.org <mailto:googletalk%3aconf%2b...@conference.freeswitch.org> pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Outbound call - choose profile
Thanks for the clarification Brian West wrote: The external profile is setup to parse the gateways... internal is setup to parse the domains and apply an alias to the internal profile for all domains in the directory. On internal you have: Then on external you have: Notice this tells internal to add an alias for every domain to internal, while external is told to parse for gateways. /b On Jan 17, 2009, at 9:29 AM, Apostolos Pantsiopoulos wrote: Yes, I am aware of that fact : insteaad of the pre-configured internal/external profiles I coulf have myprofile1/myprofile2. But since I am using the pre-configured profiles (without changing the ports) and since the internal profile initiated the parsing I would expect that all the gateways defined in my directory user's xml files would belong to the internal profile (which is preconfigured to use port 5060). But when I am capturing the packets sent to that gateway I can see that the port used from FS is 5080 (which is the preconfigured port of the internal profile). ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] enforcing codec in gateway (or call) level
I am using xml_curl to create my dialplans on the fly. I want to be able to use deferent codec preferences on each call (depending on various conditions i.e. gateway,destination etc.) I understand that I have to set inbound-late-negotiation=true in my profile in order for the dialplan to be processed before the codec negotiation. The only thing that I cannot figure out is how to "tell" FS to use a specific codec (or series of codecs) before bridging a call. Which app do I use or which variable do I set (without of course changing any global variable or profile's "codec-prefs" variable)? -- ------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] enforcing codec in gateway (or call) level (SOLVED)
Sorry I just found that you can achieve that by codec_string and absolute_codec_string by grepping the code :) Apostolos Pantsiopoulos wrote: > I am using xml_curl to create my dialplans on the fly. > I want to be able to use deferent codec preferences on > each call (depending on various conditions i.e. gateway,destination etc.) > I understand that I have to set inbound-late-negotiation=true in my profile > in order for the dialplan to be processed before the codec negotiation. > > The only thing that I cannot figure out is how to "tell" FS to use > a specific codec (or series of codecs) before bridging a call. Which > app do I use or which variable do I set (without of course > changing any global variable or profile's "codec-prefs" variable)? > > -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] mod_radius_cdr questions and thoughts
I am trying to implement a radius based solution using FS. I have seen that the mod_radius_cdr module is actively maintained. so I have a few questions/remarks : 1) When I place a call and my radius server is down, the call blocks forever instead of just radius_timeout * radius_retries seconds (I have declared only one server). I would expect that FS would stop trying to send an Acc Start packet after some time and get on with the call. 2) I have also noticed that FS sends only 1 packet (I waited for a minute) instead of 3 (default in the config) since the first (and second) attempt failed. If my server was up (the port was responding) but it returned a req. failed answer would the above time-out be valid? 3) When I tried to load the dictionary.freeswitch to my freeradius server, it complained : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[33]: dict_addattr: Duplicate attribute name NAS-Port-Id" How can I overcome this? When I "instructed" my freeradius to only load the freesitch dictionary I got this : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[257]: unknown option "Freeswitch"" How can I use the dictionary with the freeradius? (or other radius servers?) 4) The radius attributes included in the current requests are a) hard-coded, b) limited in number. I think many of us would like to use more attributes. Or even better define what to include (and what to put in them) using a config file (the same maybe?) 5) Does the module send accounting packets only for the a-leg of a call or for both legs? (Maybe that could be configurable too). If anyone is interested in the above questions/remarks please post a reply. I would really like to know how many of the mailing list users are also interested in FS radius support and your opinions on the matter. Regards, -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_radius_cdr questions and thoughts
Chris Parker wrote: On Thu, Jan 22, 2009 at 10:45 AM, Apostolos Pantsiopoulos mailto:r...@kinetix.gr>> wrote: I am trying to implement a radius based solution using FS. I have seen that the mod_radius_cdr module is actively maintained. so I have a few questions/remarks : 1) When I place a call and my radius server is down, the call blocks forever instead of just radius_timeout * radius_retries seconds (I have declared only one server). I would expect that FS would stop trying to send an Acc Start packet after some time and get on with the call. I have not seen this behavior. If you can duplicate this, and propose a patch, it would be gladly welcomed. I rebuilt and retried and the behavior persists. The call progress freezes and I get the following in the log : 2009-01-22 20:48:32 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/internal/9...@xxx.xxx.xxx.xxx) State ROUTING 2009-01-22 20:48:32 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/9...@xx.xxx.xxx.xxx SOFIA ROUTING 2009-01-22 20:48:32 [DEBUG] mod_radius_cdr.c:152 my_on_routing() [mod_radius_cdr] Entering my_on_routing After I hangup the client and issue a shutdown in FS I get the following : 2009-01-22 20:50:50 [CRIT] sofia.c:794 sofia_profile_thread_run() Waiting for 1 session(s) repeatedly and FS never exits. 2) I have also noticed that FS sends only 1 packet (I waited for a minute) instead of 3 (default in the config) since the first (and second) attempt failed. If my server was up (the port was responding) but it returned a req. failed answer would the above time-out be valid? I have not seen this behavior. The same here after the rebuild. 3) When I tried to load the dictionary.freeswitch to my freeradius server, it complained : Don't do that. The dictionary is for use with the radiusclient library. FreeRADIUS already includes a dictionary for FreeSWITCH VSAs ( you may need to uncomment it to have it loaded into FreeRADIUS ). I cannot find any reference to Freeswitch in the freeradius integrated dictionaries (in the share folder). Can you pinpoint the directory that a dictionary.freeswitch (or other FS related dictionary) resides? 4) The radius attributes included in the current requests are a) hard-coded, b) limited in number. I think many of us would like to use more attributes. Or even better define what to include (and what to put in them) using a config file (the same maybe?) This has been proposed. There isn't yet a mechanism, though the intent is to use a general purpose FS VSA for this. The code needs to be added to the mod_radius_cdr module to allow that to be a run_time configuration option. A general purpose VSA that holds only one value or many? Or a mix (array like)? 5) Does the module send accounting packets only for the a-leg of a call or for both legs? (Maybe that could be configurable too). If anyone is interested in the above questions/remarks please post a reply. I would really like to know how many of the mailing list users are also interested in FS radius support and your opinions on the matter. Again, patches are welcome. :) -Chris ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] mod_radius_cdr questions and thoughts
I am trying to implement a radius based solution using FS. I have seen that the mod_radius_cdr module is actively maintained. so I have a few questions/remarks : 1) When I place a call and my radius server is down, the call blocks forever instead of just radius_timeout * radius_retries seconds (I have declared only one server). I would expect that FS would stop trying to send an Acc Start packet after some time and get on with the call. 2) I have also noticed that FS sends only 1 packet (I waited for a minute) instead of 3 (default in the config) since the first (and second) attempt failed. If my server was up (the port was responding) but it returned a req. failed answer would the above time-out be valid? 3) When I tried to load the dictionary.freeswitch to my freeradius server, it complained : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[33]: dict_addattr: Duplicate attribute name NAS-Port-Id" How can I overcome this? When I "instructed" my freeradius to only load the freesitch dictionary I got this : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[257]: unknown option "Freeswitch"" How can I use the dictionary with the freeradius? (or other radius servers?) 4) The radius attributes included in the current requests are a) hard-coded, b) limited in number. I think many of us would like to use more attributes. Or even better define what to include (and what to put in them) using a config file (the same maybe?) 5) Does the module send accounting packets only for the a-leg of a call or for both legs? (Maybe that could be configurable too). If anyone is interested in the above questions/remarks please post a reply. I would really like to know how many of the mailing list users are also interested in FS radius support and your opinions on the matter. Regards, -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_radius_cdr questions and thoughts
the first one would not appear in the mailing list 2 hours after I had sent it so I figured there was something wrong and I resent it Brian West wrote: Did you mean to send this twice? /b On Jan 22, 2009, at 8:52 AM, Apostolos Pantsiopoulos wrote: I am trying to implement a radius based solution using FS. I have seen that the mod_radius_cdr module is actively maintained. so I have a few questions/remarks : 1) When I place a call and my radius server is down, the call blocks forever instead of just radius_timeout * radius_retries seconds (I have declared only one server). I would expect that FS would stop trying to send an Acc Start packet after some time and get on with the call. 2) I have also noticed that FS sends only 1 packet (I waited for a minute) instead of 3 (default in the config) since the first (and second) attempt failed. If my server was up (the port was responding) but it returned a req. failed answer would the above time-out be valid? 3) When I tried to load the dictionary.freeswitch to my freeradius server, it complained : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[33]: dict_addattr: Duplicate attribute name NAS-Port-Id" How can I overcome this? When I "instructed" my freeradius to only load the freesitch dictionary I got this : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[257]: unknown option "Freeswitch"" How can I use the dictionary with the freeradius? (or other radius servers?) 4) The radius attributes included in the current requests are a) hard-coded, b) limited in number. I think many of us would like to use more attributes. Or even better define what to include (and what to put in them) using a config file (the same maybe?) 5) Does the module send accounting packets only for the a-leg of a call or for both legs? (Maybe that could be configurable too). If anyone is interested in the above questions/remarks please post a reply. I would really like to know how many of the mailing list users are also interested in FS radius support and your opinions on the matter. Regards, -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_radius_cdr questions and thoughts
Hi, After some testing I came to the following conclusions : 1) The problem (timeouts and retries) I describe below only happens when there is no radius server responding on the other side. 2) It only happens when using the latest cvs version of radiusclient. If you use version 1.1.6 it works fine. I also read in the wiki (and found out myself by testing) that : "Currently, the module blocks the thread while it is sending the requests. This may cause threads to hang around longer than expected after a call, if your RADIUS servers are not reachable/responding." which I think is not desirable. Was this kind of behavior, been followed intentionally? I think that the NAS in most (if not all) implementations uses a non-blocking operation in order to proceed with the call. In that way there is not any significant delay (up to 15 seconds if radius is down) in the beginning of the call. Also, I noticed that if the radius acct packet fails, FS does not proceed with the call which is again -in my opinion - wrong. I think that the NAS should be able to continue with the call even if the Acct start or stop failed. For those directly involved in the maintenance of the mod_radius_cdr code : Is it relatively easy to change the blocking behavior of the module? Apostolos Pantsiopoulos wrote: Chris Parker wrote: On Thu, Jan 22, 2009 at 10:45 AM, Apostolos Pantsiopoulos mailto:r...@kinetix.gr>> wrote: I am trying to implement a radius based solution using FS. I have seen that the mod_radius_cdr module is actively maintained. so I have a few questions/remarks : 1) When I place a call and my radius server is down, the call blocks forever instead of just radius_timeout * radius_retries seconds (I have declared only one server). I would expect that FS would stop trying to send an Acc Start packet after some time and get on with the call. I have not seen this behavior. If you can duplicate this, and propose a patch, it would be gladly welcomed. I rebuilt and retried and the behavior persists. The call progress freezes and I get the following in the log : 2009-01-22 20:48:32 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/internal/9...@xxx.xxx.xxx.xxx) State ROUTING 2009-01-22 20:48:32 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/9...@xx.xxx.xxx.xxx SOFIA ROUTING 2009-01-22 20:48:32 [DEBUG] mod_radius_cdr.c:152 my_on_routing() [mod_radius_cdr] Entering my_on_routing After I hangup the client and issue a shutdown in FS I get the following : 2009-01-22 20:50:50 [CRIT] sofia.c:794 sofia_profile_thread_run() Waiting for 1 session(s) repeatedly and FS never exits. 2) I have also noticed that FS sends only 1 packet (I waited for a minute) instead of 3 (default in the config) since the first (and second) attempt failed. If my server was up (the port was responding) but it returned a req. failed answer would the above time-out be valid? I have not seen this behavior. The same here after the rebuild. 3) When I tried to load the dictionary.freeswitch to my freeradius server, it complained : Don't do that. The dictionary is for use with the radiusclient library. FreeRADIUS already includes a dictionary for FreeSWITCH VSAs ( you may need to uncomment it to have it loaded into FreeRADIUS ). I cannot find any reference to Freeswitch in the freeradius integrated dictionaries (in the share folder). Can you pinpoint the directory that a dictionary.freeswitch (or other FS related dictionary) resides? 4) The radius attributes included in the current requests are a) hard-coded, b) limited in number. I think many of us would like to use more attributes. Or even better define what to include (and what to put in them) using a config file (the same maybe?) This has been proposed. There isn't yet a mechanism, though the intent is to use a general purpose FS VSA for this. The code needs to be added to the mod_radius_cdr module to allow that to be a run_time configuration option. A general purpose VSA that holds only one value or many? Or a mix (array like)? 5) Does the module send accounting packets only for the a-leg of a call or for both legs? (Maybe that could be configurable too). If anyone is interested in the above questions/remarks please post a reply. I would really like to know how many of the mailing list users are also interested in FS radius support and your opinions on the matter. Again, patches are welcome. :) -Chris ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Re: [Freeswitch-users] mod_radius_cdr questions and thoughts
Please see in line comments. Anthony Minessale wrote: Sounds like having cake and eating it too. The risk is obvious when using radius or some other additional protocol for AAA that you will have trouble if the server is down. Radius was designed to be fast and redundant, so typically you would have 5 radius servers not one. I did not make mod_radius_cdr but I do pose this question: Where should the burden lie to make sure the calls are not delayed when choosing this option? Within the module. A separate thread could deal with the Acct Packet transmission without blocking the flow of the call. After a number of retries the thread should quit trying. If you make FS cache all the radius requests it could not complete in a timely manner, the whole point of keeping track of the exact time they occurred is lost and you could have calls that ended before the start packet ever was transmitted because they are cached in some process that will begin to swell with memory remembering all the requests it could not send. The acct start and stop packets contain the time info needed for billing purposes Even if the radius packets reaches the server 10 secs after the end of the call there is little harm done. The "real" start and stop times don't have to correspond to the times the packets arrived at the radius server. Then somehow it needs to gracefully catch up again when the radius server comes back This is the same reason I think that direct database CDR is a bad idea. If the retries*timeout time has passed the NAS should give up trying to send the packet. So there is not much catching up to do. The radius packets that failed while the radius server was down don't have to be retransmitted later. All the applications and users that use radius are comfortable with that fact. I for one use a x-checking mechanism (comparing CDRs with radius created CDRs) to verify the integrity of my calls. The real answer is that you are not allowed to have your radius server down at all, so you need more than one. I used to be in the dialup business and we had to have backup radius servers for the backup radius servers on a completely different network just in case not only the server was down but the network link to the server and it's backup server. It's like DNS, I can't ever be down or nothing works. I always use multiple radius servers (using different routes to my NASes). But sometimes there are other issues that could interfere with the NAS-Radius connectivity. On Tue, Jan 27, 2009 at 7:05 AM, Apostolos Pantsiopoulos mailto:r...@kinetix.gr>> wrote: Hi, After some testing I came to the following conclusions : 1) The problem (timeouts and retries) I describe below only happens when there is no radius server responding on the other side. 2) It only happens when using the latest cvs version of radiusclient. If you use version 1.1.6 it works fine. I also read in the wiki (and found out myself by testing) that : "Currently, the module blocks the thread while it is sending the requests. This may cause threads to hang around longer than expected after a call, if your RADIUS servers are not reachable/responding." which I think is not desirable. Was this kind of behavior, been followed intentionally? I think that the NAS in most (if not all) implementations uses a non-blocking operation in order to proceed with the call. In that way there is not any significant delay (up to 15 seconds if radius is down) in the beginning of the call. Also, I noticed that if the radius acct packet fails, FS does not proceed with the call which is again -in my opinion - wrong. I think that the NAS should be able to continue with the call even if the Acct start or stop failed. For those directly involved in the maintenance of the mod_radius_cdr code : Is it relatively easy to change the blocking behavior of the module? Apostolos Pantsiopoulos wrote: Chris Parker wrote: On Thu, Jan 22, 2009 at 10:45 AM, Apostolos Pantsiopoulos mailto:r...@kinetix.gr>> wrote: I am trying to implement a radius based solution using FS. I have seen that the mod_radius_cdr module is actively maintained. so I have a few questions/remarks : 1) When I place a call and my radius server is down, the call blocks forever instead of just radius_timeout * radius_retries seconds (I have declared only one server). I would expect that FS would stop trying to send an Acc Start packet after some time and get on with the call. I have not seen this behavior. If you can duplicate this, and propose a patch, it would be gladly welcomed. I rebuilt and retried and the behavior persists. The call progress freezes and I get the following in the log : 20
[Freeswitch-users] Call-Direction issue
Hi,
I am trying to include the Call-Direction variable (${direction}) in my
csv cdrs. But I get nothing. I tried printing the variable in the
console before a bridge :
But nothing there too. When I use uuid_dump during a call I can see the
Call-Direction variable. I tried call_direction as well but it doesn't
work either. Any ideas?
--
---
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: r...@kinetix.gr
---
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[Freeswitch-users] killall -HUP freeswitch
Hi, I am doing a "killall -HUP freeswitch" in order to achieve cdr-csv log rotation and the process gets killed instead. A few days ago it worked fine. I also noticed that a few days ago when I hit Ctrl+C freeswitch did not exit. Now, when I do this the process gets terminated immediately. What could be the cause of all this? I am using the svn 12548 trunk. I did not do any recompiling of FS. -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] killall -HUP freeswitch
I found out the cause : When the mod_opal module is loaded the FS process gets killed with a kill -HUP. I thought it would be good for everyone to know. Apostolos Pantsiopoulos wrote: > Hi, > > I am doing a "killall -HUP freeswitch" in order to achieve > cdr-csv log rotation and the process gets killed instead. A few > days ago it worked fine. I also noticed that a few days ago when I > hit Ctrl+C freeswitch did not exit. Now, when I do this the process gets > terminated immediately. What could be the cause of all this? I am using > the svn 12548 trunk. I did not do any recompiling of FS. > > > -- ------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Testing Freeswitch performance led to strange behavior
In the process of trying to use Freeswitch in a production environment I conducted a number of performance tests using various servers. It was then that I noticed some strange behavior from FS. When I stripped down the scenario I was using to a simple bridge scenario, I stumbled upon a strange behavior. The scenario as I stated is quite simple. |-| |-| | |--- Call from sipp--> | | | sipp | | FS| | | <-- Call back to sipp| | |-| |-| I did not use an RTP stream for my calls just to test the signaling alone. The sipp scenario is the standard uac.xml scenario that can be found integrated to sipp with the following options : Test FS 1: sipp :5060 -s -i -mi -ci -r 10 -d 5000 -l 100 -m 1000 -sf uac.xml Calls : 1000 Successful calls : 1000 Idle CPU during tests : ~(35-60) % (35 during the generation of new calls, 60 during the -l limit imposed by the test) Note : 985 of them had a duration (billsec) of 10 and 15 of them had a duration of 11. I tried raising the call rate and limit... Test FS 1: sipp :5060 -s -i -mi -ci -r 20 -d 5000 -l 200 -m 1000 -sf uac.xml Calls : 1000 Successful calls : 1000 Idle CPU during tests : ~0-30 % (0 during the generation of new calls, 30 during the -l limit imposed by the test) THIS IS WHAT MAKES ME WONDER : The distribution of the durations (billsec - not complete durations) : 183 calls with 10 secs billed duration 110 calls with 11 secs billed duration 238 calls with 12 secs billed duration 447 calls with 13 secs billed duration 22 calls with 14 secs billed duration The sipp scenario is simple "hangup the phone after 10 secs". So, why am I seeing these? Of course that has something to do with the stress the machine has been put through during the second test. But I can see it happening to less stressful conditions (i.e. 15 calls per second) to a smaller extend. I captured one of these calls and verified that when the sipp client hangs up exactly 10 secs after the call start, FS receives the BYE and replies with 200 OK. BUT it does not hang the second leg in a timely manner i.e. it sends a BYE to the sipp server side 1-4 seconds AFTER that. That explains the 11, 12, 13, 14 secs durations seen on the second test. What is more interesting is that I would expect to see in the CDRs the first and second leg to have different durations (since the a leg BYE was received and aknowledged by FS in the correct time) i.e. 10 and 14 secs accordingly. But what I get is the same duration for both legs (14 secs). This in my opinion is very dangerous on production environments (you get charged by your provider more seconds that you charge your clients - or - you falsely charge your clients with bigger durations although they hunged up corectly (and you acknowledged it)). NOTE No 1 : All the performance recommendations found in the wiki has been applied. In fact only the essential modules that could make this scenario work were loaded. NOTE No 2 : I tried using asterisk (as a point of reference - don't get me wrong - I am not trying to start a flame war here). And it succeeded doing on the same hardware 60 calls/sec with a channel limit of 400 sim. calls using only 50% of the cpu (maximum). No under any circumstances I have seen the behavior above (this inability to hang call legs in a timely manner). Even when I pushed asterisk to the limits (80 calls per second 600 max call limit) and it started failing on some calls it never failed to hangup the calls for both legs on exactly 10 secs. NOTE No 3 : As you can tell I was using a very small machine for my tests. When I moved the same tests to larger installations (Quad Core Opterons and Xeons) I got proportional results to the above. NOTE No 4 : The tests were performed in a LAN environment and since there was no RTP involved I think there were no bandwidth issues there. NOTE No 5 : The tests were performed using numerous SVN versions (latest : 13610), the stable version and the 1.0.4pre8 version. NOTE No 6 : Using the -hp switch made no noticeable change in behavior. I am not trying to complain for FS's performance (far from it). I am just somewhat disappointed seeing it performing in such a strange manner when under stress. I would prefer a design that drops the calls after a certain threshold than a design that incorrectly handles them all (I am aware of the max sessions per second in switch.conf.cml - but I am starting to see this behavior even with the cpu idling at about 80%). I don't know if anyone else had the same experience when testing Freeswitch. I can happily supply with all the test details (config files, captures etc) to all interested parties. -- --
Re: [Freeswitch-users] Testing Freeswitch performance led to strange behavior
Brian West wrote: > > On Jun 4, 2009, at 9:32 AM, Apostolos Pantsiopoulos wrote: > >> NOTE No 1 : All the performance recommendations found in the wiki has >> been applied. In fact only the essential modules that could make this >> scenario work >> were loaded. > > What are you testing against? What OS, Hardware, Distro and such? The small server tests were made on a 5-year old PC (32 bit, 3 Ghz P4, Cetnos 5.3). The large server 1 : Quad-Core AMD Opteron(tm) Processor 2350 HE (64 bit, Centos 5.3) The large server 2 : Dual-Core AMD Opteron(tm) Processor 2214 HE (64 bit, Centos 5.3) The large server 3 : Intel(R) Xeon(R) CPU E5345 @ 2.33GHz (32 bit, Centos 5.3) > >> NOTE No 2 : I tried using asterisk (as a point of reference - don't get >> me wrong - I am not trying to start a flame war here). And it succeeded >> doing on the same hardware 60 calls/sec with a channel limit of 400 >> sim. calls using only 50% of the cpu (maximum). No under any >> circumstances I have seen the behavior above (this inability to hang >> call legs in a timely manner). Even when I pushed asterisk to the limits >> (80 calls per second 600 max call limit) and it started failing on some >> calls it never failed to hangup the calls for both legs on exactly 10 >> secs. > > Load testing is a science and you can do it wrong most of the time > unless you know exactly what you're doing. If you're going against the > default dialplan its heavy and not something I would load test against. The dialplan : I think it is the simplest that can be used in this scenario. > > >> NOTE No 3 : As you can tell I was using a very small machine for my >> tests. When I moved the same tests to larger installations (Quad Core >> Opterons and Xeons) I got proportional results to the above. > > What are you testing on now? Hope its 64bit. Most of the platforms were 64 bit (although the results that I posted were from the small 32-bit server, the results from the 64-bit servers were proportional to those). In other words we needed a large call/sec rate for the high end servers but in any case the same phenomenon occured at around 60% idle cpu. -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Testing Freeswitch performance led to strange behavior
eeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_miness...@hotmail.com > <mailto:msn%3aanthony_miness...@hotmail.com> > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > <mailto:paypal%3aanthony.miness...@gmail.com> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > <mailto:sip%3a...@conference.freeswitch.org> > iax:gu...@conference.freeswitch.org/888 > <http://iax:gu...@conference.freeswitch.org/888> > googletalk:conf+...@conference.freeswitch.org > <mailto:googletalk%3aconf%2b...@conference.freeswitch.org> > pstn:213-799-1400 > > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Testing Freeswitch performance led to strange behavior
I am writing this to let you know that this behavior persists in the 1.0.4pre9. Could the calls/sec issue be due to the single threaded nature of Sofia? Because I am getting the feeling that the number of simultaneous channels doesn't really burdens FS, but many Calls/sec does. Apostolos Pantsiopoulos wrote: > Anthony Minessale wrote: >> FS uses async rtp timers so you may want to set rtp-timer-name=none in >> the profile param to simulate asterisk conditions. > > I tried that - although I am not using rtp in my scenario - with the > same results. > >> Also keep in mind that asterisk as an atvantage in a tiny crappy 32 bit >> single cpu box because that was what was popular when it was designed >> and the chance for race conditions is minimal because there is only 1 >> cpu. As you scale up to a 8 core 64 bit xeon you will set a drastic >> difference. > > Yes I know that this machine is not well suited for today's test needs. > But the issue occurs in every machine as long as it is pushed near (but > not quite near) to its limits. I have the same odd durations using a 64 > bit low end server. In this case I could achieve a better call/sec rate > than that of the crappy PC but around 50-60 calls/sec the same problem > showed up. I also used a Mosso Cloud Server (quad core - 64-bit) and the > same thing happened at a higher rate. > > >> I will be happy to investigate this issue a bit if you'd like but i do >> not have any box like you describe so if I can't find anything >> you may have to lend us your lab. > > I would appreciate it if you did. After all there this might be a > problem that has not surfaced yet but someday will as more and more > production boxes start using FS. So it would be better to investigate it > now. > I don't think lending you access to my old P4 PC would help you very much :) > If you have access to a normal 2-4 core system you can easily start > raising the sipp parameters until it starts happening. However if you > really think it is appropriate to use my test machines I'd be happy to > grant access to our low-end Opteron machine (just send me a personal > email). I cannot grant you access to larger systems because they are > used in production. > > I used the embedded sipp scenarios : > > on the UAS side : > > sipp -i -mi -ci -mp 8000 -sn uas > > on the UAC side : > > sipp :5060 -s 44050505-i -mi -ci -r 70 > -d 5000 -l 500 -m 2000 -sn uac > > The dialplan : > > > > > > > > > > data="absolute_codec_string=PCMU"/> > >data="sofia/gateway/sipp01/$1"/> > > > > > > > If you need anything else from the config just notify me. > > In order to verify that at some point the calls start having a > duration larger than the scenario's 5secs you can tcpdump on the sipp > machine or turn on the cdrs logging (I know that it degrades > performance, but as I said it is not a matter of when exactly it > starts happening, it is a matter that it DOES start happening). > > >> >> On Thu, Jun 4, 2009 at 12:47 PM, r...@kinetix.gr >> <mailto:r...@kinetix.gr> mailto:r...@kinetix.gr>> wrote: >> >> Michael Collins wrote: >> > >> > >> > The dialplan : >> > >> > >> > >> > >> > >> > >> > >> > > > expression="^.*$"> >> > >> > >> > You forgot the parens around .* >> > It should be expression="^(.*)$" if you plan to use $1 later in the >> > extension... >> > >> > >> > >> > >> > > > data="absolute_codec_string=PCMA"/> >> > > > data="sofia/gateway/sipp01/$1"/> >> > >> > ... like here ^^^ >> > :) >> > -MC >> >> You are right! Although, I don't think that would change the outcome of >> my test :) >> > >> > >> > >> > >> > >> > >> > >> > >> > >>
[Freeswitch-users] Proto specific hangup cause issue
The proto_specific_hangup_cause is missing on one of the two call legs. When the caller hangs up it is missing from the a-leg CDR. When the callee hangs up it is missing from the b-leg CDR. Is this nornal? And another question : what piece of info could inform me about who hanged up? -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: [EMAIL PROTECTED] --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Proto specific hangup cause issue
Both legs are SIP. From non-registered endpoints (if of any use). Michael S Collins wrote: I will do some research on this and let you know what I find out. Question: are these internal calls or pstn or ?? Just curious about your environment. Thanks, MC On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <[EMAIL PROTECTED]> wrote: The proto_specific_hangup_cause is missing on one of the two call legs. When the caller hangs up it is missing from the a-leg CDR. When the callee hangs up it is missing from the b-leg CDR. Is this nornal? And another question : what piece of info could inform me about who hanged up? -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: [EMAIL PROTECTED] --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: [EMAIL PROTECTED] --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Proto specific hangup cause issue
I am sending this second email to request/suggest/enquire about something relevant : Wouldn't it be useful to know which end of a specific call leg send the protocol specific hangup cause? Otherwise it would be difficult to understand what really happened. Michael S Collins wrote: I will do some research on this and let you know what I find out. Question: are these internal calls or pstn or ?? Just curious about your environment. Thanks, MC On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <[EMAIL PROTECTED]> wrote: The proto_specific_hangup_cause is missing on one of the two call legs. When the caller hangs up it is missing from the a-leg CDR. When the callee hangs up it is missing from the b-leg CDR. Is this nornal? And another question : what piece of info could inform me about who hanged up? -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: [EMAIL PROTECTED] --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: [EMAIL PROTECTED] --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Proto specific hangup cause issue
FreeSWITCH Version 1.0.trunk (10579) Brian West wrote: Did you say what SVN rev you're running. /b On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote: Both legs are SIP. From non-registered endpoints (if of any use). ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: [EMAIL PROTECTED] --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Proto specific hangup cause issue
I tested it and it works fine but it got me thinking...
Is just a copy of the cause to the other leg the correct way
to do it? Couldn't the two call legs hang up with different causes?
Especially when I could override the cause before it got send
to the e.g. calling side using e.g. the hangup command?
To make myself clear : I could have the b-leg (in a bridge hangup)
sending me a user busy code and I could send a circuit/channel
unavailable to my caller (a-leg), let's
say because I don't trust my terminator (b-leg) and his codes and I want to
enforce another one and send it to my originator so that he could retry
another
carrier.
What do you think?
Anthony Minessale wrote:
It's easy enough to set the value on both legs try r10614
It was only set on the opposing leg before but since it's harmless to
set it on both i did it for you.
On Fri, Dec 5, 2008 at 10:23 AM, Brian West <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> wrote:
Did you say what SVN rev you're running.
/b
On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:
> Both legs are SIP. From non-registered endpoints (if of any use).
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Re: [Freeswitch-users] Proto specific hangup cause issue
Any updates about the "which side hanged up" potential variable? Michael S Collins wrote: Makes sense. I will look into this. -MC On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: I am sending this second email to request/suggest/enquire about something relevant : Wouldn't it be useful to know which end of a specific call leg send the protocol specific hangup cause? Otherwise it would be difficult to understand what really happened. Michael S Collins wrote: I will do some research on this and let you know what I find out. Question: are these internal calls or pstn or ?? Just curious about your environment. Thanks, MC On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: The proto_specific_hangup_cause is missing on one of the two call legs. When the caller hangs up it is missing from the a-leg CDR. When the callee hangs up it is missing from the b-leg CDR. Is this nornal? And another question : what piece of info could inform me about who hanged up? -- ------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- --- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: [EMAIL PROTECTED] --- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Proto specific hangup cause issue
Not necessarily. For instance I got a "send_cancel" when the
calling party hanged up before the other party could pick up.
Also, shouldn't something like that be protocol/technology
neutral?
Anthony Minessale wrote:
sip_hangup_disposition will be set to recv_bye on the side that was
hungup.
On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos
<[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
Any updates about the "which side hanged up" potential variable?
Michael S Collins wrote:
Makes sense. I will look into this.
-MC
On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos
<[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
I am sending this second email to request/suggest/enquire about
something relevant :
Wouldn't it be useful to know which end of a specific call leg
send the protocol
specific hangup cause? Otherwise it would be difficult to
understand what really happened.
Michael S Collins wrote:
I will do some research on this and let you know what I find out.
Question: are these internal calls or pstn or ?? Just curious about
your environment.
Thanks,
MC
On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos < <mailto:[EMAIL PROTECTED]>[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>
wrote:
The proto_specific_hangup_cause is missing on one of the two
call legs. When the caller hangs up it is missing from the a-leg CDR.
When the callee hangs up it is missing from the b-leg CDR. Is this
nornal?
And another question : what piece of info could inform me about who
hanged up?
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Apostolos Pantsiopoulos
Kinetix Tele.com R & D
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Anthony Minessale II
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Re: [Freeswitch-users] Proto specific hangup cause issue
"I already added 2 patches for you right. Just be clear about what you
want."
And I am grateful of that.
"it is protocol neutral, that's why it starts with sip_"
I didn't know that. I thought that the sip_ variables are protocol
specific. So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?
"Maybe you should beat around the bush less with your "requirements" for
your application you are expecting me to support for you."
I am just trying to gather statistics for my providers as I would with
any VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a
general necessity for VoIP carriers. It is also very useful for
troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain
for, I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot
the issue better.
"Just be clear about what you want."
I want FS to reach that level of detailing and maturity in all aspects
so that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by
the community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The
patches that you made the last few days *were not intended for
me* exclusively but for *anyone* that will face the same situations
using FS. If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or
just use it as an announcement board.
I wish I was a c programmer and get involved with the project actively.
But I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing
and suggesting.
Anthony Minessale wrote:
it is protocol neutral, that's why it starts with sip_
the variable can be any of:
send_bye
recv_bye
send_cancel
send_refuse
using that value you can determine the information you asked. I
answered your specific question which was:
determining "which side hanged up". Maybe you should beat around the
bush less with your "requirements" for your application you are
expecting me to support for you.
I already added 2 patches for you right. Just be clear about what you
want.
On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos
<[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
Not necessarily. For instance I got a "send_cancel" when the
calling party hanged up before the other party could pick up.
Also, shouldn't something like that be protocol/technology
neutral?
Anthony Minessale wrote:
sip_hangup_disposition will be set to recv_bye on the side that
was hungup.
On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos
<[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
Any updates about the "which side hanged up" potential variable?
Michael S Collins wrote:
Makes sense. I will look into this.
-MC
On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos
<[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
I am sending this second email to request/suggest/enquire
about something relevant :
Wouldn't it be useful to know which end of a specific call
leg send the protocol
specific hangup cause? Otherwise it would be difficult to
understand what really happened.
Michael S Collins wrote:
I will do some research on this and let you know what I find out.
Question: are these internal calls or pstn or ?? Just curious about
your environment.
Thanks,
MC
On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos < <mailto:[EMAIL PROTECTED]>[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>
wrote:
The proto_specific_hangup_cause is missing on one of the two
call legs. When the caller hangs up it is missing from the a-leg CDR.
When the callee hangs up it is missing from the b-leg CDR. Is this
nornal?
And another question : what piece of info could inform me about who
hanged up?
--
---
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
---
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Re: [Freeswitch-users] Proto specific hangup cause issue
We are currently in the migration process from our current system to a FS based setup. We are in the process of adapting our billing and routing to FS. All the CDRs (and variables) related issues that we have been discussing on this mailing list come from the need to extract the same level of information from FS as we do with our current closed source proprietary system. So, we chose FS because of the versatility it provides in every aspect (event handling, config implementation etc.) and we strongly believe that all these additions/fixes would be beneficial to many potential FS users. We are at your disposal for more details in case you need more information about what exactly we are trying to do. Basically, our approach is from the VoIP carrier's point of view rather than the PBX user's/implementor's. So, the details that we asked to be introduced to FS come from real life issues that we have faced during the last few years with various platforms and troubleshooting experiences with other VoIP carriers. Michael Collins wrote: Thanks for your feedback. It definitely helps to know not only what you need FS to do but why you need it to do so. Do you have FS in production right now? Just curious. Thanks, MC On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos <[EMAIL PROTECTED]> wrote: "I already added 2 patches for you right. Just be clear about what you want." And I am grateful of that. "it is protocol neutral, that's why it starts with sip_" I didn't know that. I thought that the sip_ variables are protocol specific. So one would expect there to be an iax_hangup_disposition, woomera_hangup_disposition etc? "Maybe you should beat around the bush less with your "requirements" for your application you are expecting me to support for you." I am just trying to gather statistics for my providers as I would with any VoIP softswitch. (hangup causes per terminator per destination) I don't think that this is a specific "application" rather than a general necessity for VoIP carriers. It is also very useful for troubleshooting purposes : when I look at my CDRs to find a call that I got a complain for, I want to be able to tell if it was me or the provider who hanged up and gave a specific hangup cause, so that I can troubleshoot the issue better. "Just be clear about what you want." I want FS to reach that level of detailing and maturity in all aspects so that it could be the softswitch of choice by any VoIP entrepreneur (or hobbyist) and it is my strong belief that this can only be done by the community giving feedback to the programmers about what they find useful or not (i.e. experience from real-life situations). The patches that you made the last few days were not intended for me exclusively but for anyone that will face the same situations using FS. If you want the community to stop sending feedback about features/improvements you may as well close down this mailing list or just use it as an announcement board. I wish I was a c programmer and get involved with the project actively. But I am not. And as far as I can tell most of the registered users in this list aren't either. So they only way we can help is by testing and suggesting. Anthony Minessale wrote: it is protocol neutral, that's why it starts with sip_ the variable can be any of: send_bye recv_bye send_cancel send_refuse using that value you can determine the information you asked. I answered your specific question which was: determining "which side hanged up". Maybe you should beat around the bush less with your "requirements" for your application you are expecting me to support for you. I already added 2 patches for you right. Just be clear about what you want. On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <[EMAIL PROTECTED]> wrote: Not necessarily. For instance I got a "send_cancel" when the calling party hanged up before the other party could pick up. Also, shouldn't something like that be protocol/technology neutral? Anthony Minessale wrote: sip_hangup_disposition will be set to recv_bye on the side that was hungup. On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <[EMAIL PROTECTED]> wrote: Any updates about the "which side hanged up" potential variable? Michael S Collins wrote: Makes sense. I will look into this. -MC On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <[EMAIL PROTECTED]> wrote: I am sending this second email to request/suggest/enquire about something relevant : Wouldn't it be useful to know which end of a specific call leg send the protocol specific hangup cause? Otherwise it would be difficult to understand what really happened. Michael S Collins wrote: I will do some research on this and let you know what I find out. Question: are these internal calls or pstn
