Re: [Freeswitch-users] Destination Formats Expression
Thanks that will be a great help Jason White-14 wrote: Edmar Cruz darklio...@yahoo.com wrote: Is there a link or tutorial for the expressions format. Anything that describes Perl regular expressions should help, and for reference, see the pcre(3) manual page, and use the pcretest program to experiment. ___ 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 -- View this message in context: http://old.nabble.com/Destination-Formats-Expression-tp26840010p26840254.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] SIP Re-invite
Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ___ 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] Voicemail-Email
I get the same result with sendmail. This used to work in 1.0.3 , and
after upgrading to 1.0.4 (then some snapshots) then 1.0.5pre8 and the
problem is still there.
François
On Thu, 17 Dec 2009 17:33:58 +0100,
Oliver Schönbeck wrote:
Currently it is Version 1.0.trunk (15982)
VON: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] IM AUFTRAG VON Brian
West
GESENDET: Donnerstag, 17. Dezember 2009 17:17
AN:
freeswitch-users@lists.freeswitch.org
BETREFF: Re: [Freeswitch-users]
Voicemail-Email
What SVN rev. exactly?
/b
On Dec 17, 2009, at
10:13 AM, Oliver Schönbeck wrote:
Hello,
we are running freeswitch
1.0.trunk and are currently trying to get the mod_voicemail to send the
received messages to the user by using exim4 on a debian machine.
So
far we followed the instructions in the wiki article (
http://wiki.freeswitch.org/wiki/Mod_voicemail ).
I added some lines to
the bash script to enable some kind of logging:
#! /bin/bash
typeset
LOG=/tmp/${0##*/}.out
mv $LOG ${LOG}.old /dev/null 2
[[ -t 1
]] exec $LOG 2exim4 -t -v $LOG
If I run the script from the
command line everything is working as expected. If the script gets called
by freeswitch I get the following result in my logfile:
/usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation
fault (core dumped) exim4 -t -v $LOG
Has anybody seen similar
effects before?
Any advice whats going wrong is heavily appreciated.
Thanks
Oliver
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Re: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN
Should I open a JIRA for this? Best regards Peter Peter P GMX schrieb: Hello, we have the following scenario: A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For the called FS user, call forwarding has been enabled to another PSTN extension (B) . Result: The calling party does not hear any ringing tone. Here an Abstract of the SIP protocol we traced (PSTN(A) and PSTN(B) is in fact the same Patton Gateway): PSTN(A)INVITE===FS PSTN(A)===TRYING===FS FS===INVITE==PSTN(B) FS==TRYING===PSTN(B) FS==RINGING==PSTN(B) PSTN(A)==PROGRESS===FS FS===OK==PSTN(B) FSACKPSTN(B) PSTN(A)===OKFS PSTN(A)ACK==FS I would expect that FS answers RINGING back to PSTN(A). Instead it only answers SESSION PROGRESS. When PSTN(B) answers, they can hear each other, but there was no ringing tone to PSTN(A) before. Are there any hints to overcome this, besides playing early media to PSTN(A)? Best regards Peter ___ 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
[Freeswitch-users] LUA and return variables
Hi guys (and girls)! I'm working on a little bit of ENUM trickery and I tried doing some (illegal) nested conditions. :-) What I want to do is to first check enum with the ENUM application, then depending on the answer do some stuff. Say that the domain part of the ENUM answer is robin.nl, then I want to do action X instead of just briding the enum answer directly as I see in most examples. But I remembered that it wasn't allowed to do nested conditions. So what I did was stacked conditions. After that I read the dialplan wiki pages again and figured that my regexp never matches because variables I set during some phase of the extension I can't use in the same go as another condition. So, now my plan is to use LUA to do the regexp. I'll pass the enum answer to a lua script which will split the answer in a user and domain part and return those two to the main app. Then based on those two vars I'll do routing or other actions (like, prefix and then route). Is this how I'm supposed to do it? I can't find many examples on manipulating ENUM answers, other than bridging them directly. I can't change the way I do stuff to ENUM answers, because in most cases I'll just route them out the standard way. Anyone with experience on fiddling with ENUM answers? -- Robin Vleij ___ 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] Ringing after call has been rejected
I have an incomming call being answered by FreeSwitch and passed to IVR application which rejects the call. The call is never answered by FreeSwitch, but instead of hearing a busy signal, the caller hears ringing. Can anyone advise how I can get the user to hear a busy signal after call rejection instead of ringing. Here is the debug trace http://pastebin.freeswitch.org/11558 Thanks Brian -- View this message in context: http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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_conference scalability
Brian, You haven't said what codecs are being used yet. Are the listeners using a different codec to the speaker? If so, you're potentially doing transcoding on every single channel, which would make CPU usage skyrocket. -Steve 2009/12/17 Anthony Minessale anthony.miness...@gmail.com: What exactly is your test process? you should try increasing the interval in the conference profile to a bigger time slice maybe 30 40 or 60ms you could also increase the ptime to match as well. like brian said you could use mod_shout to broadcast the single speaker to icecast and let people listen with itunes/winamp On Thu, Dec 17, 2009 at 3:41 PM, Brian br...@proximosystems.com wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian br...@proximosystems.com wrote: I didn’t realize there was a policy about load testing questions. What forum should I have used for this? I didn’t get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production. On Thu, Dec 17, 2009 at 1:29 PM, Brian br...@proximosystems.com wrote: Hi Mike, I didn’t get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I’ve read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test – more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I’m doing wrong, but I don’t see what it could be. Brian. From: Michael Jerris [mailto:m...@jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I’m new to FreeSWITCH and I’m testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn’t scale nearly as well as
[Freeswitch-users] mod_xml_ldap compile issue.
Hi, I am having an issue getting mod_xml_ldap to compile properly cut-cut making all mod_xml_cdr making all mod_xml_ldap Creating mod_xml_ldap.la... /usr/bin/ld: /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a: could not read symbols: Bad value collect2: ld returned 1 exit status cat: .libs/mod_xml_ldap.log: No such file or directory make[5]: *** [mod_xml_ldap.la] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_xml_ldap-all] Error 1 make[2]: *** [all-recursive] Error 1 I notice the openldap library has been bumped up to .19 - not sure if that may have anything to do with it. At revision 15995 on a 2.6.31-15-generic Ubuntu x86_64 GNU/Linux notebook. mod_ldap compiles OK, but mod_xml_ldap fails as per the above. What am I doing working here ? Best Regards Keith ___ 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_conference scalability
I've got FS running on a 64 bit OS, and here is more info on the test
procedure.
I've got one server (primary) that hosts the speaker call (this is meant to
be a primary conference with a few speakers, but my test simplifies this to
just one speaker). I've got a second server (secondary) that hosts the
conference that all the listeners go into, and I have two other servers that
I use automate the listener calls. The goal is to have several secondary
servers to scale the listener side of things, but for this initial test I've
only got one secondary server.
The primary server dials into the secondary conference server so that the
listeners can hear the speaker conference on the primary server.
The automated listener servers start dialing into the listener conference at
a combined rate of 5 calls per second (i.e. 2.5 calls per second each). The
play an audio loop that represents noise on their end, which since they are
listeners, should be ignored anyway.
As I ramp up the automated listener calls, I manually call into the
conference from either my SIP phone, or from a land line using a DID that I
have directed to the conference.
All calls are using SIP with uLaw 8000hz codec. Also, I've set up the
profile for the listener conference to disable many of the events:
profile name=listener
param name=domain value=$${domain}/
param name=rate value=8000/
param name=moh-sound value=moh.wav/
param name=suppress-events
value=start-talking,stop-talking,energy-level,volume-level,gain-level,mute-
detect,energy-level-member,volume-in-member,volume-out-member,lock,unlock,fl
oor-change/
param name=caller-controls value=listener_controls/
/profile
I do have caller controls for the listener, since in my production I will
need to generate and handle events for listener DTMF.
To compare FreeSWITCH vs Asterisk, I just swap out the secondary conference
server and everything else stays the same.
Brian.
From: Brian West [mailto:br...@freeswitch.org]
Sent: Thursday, December 17, 2009 5:20 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_conference scalability
What exactly are you doing I know it goes better than that.. are you using
64bit?
/ b
On Dec 17, 2009, at 3:41 PM, Brian wrote:
I did a test with the trunk version for the one conference case, and it is
the same results as for 1.0.4. The audio failed at around 300 listeners.
Oddly though, it consumed less %CPU (240% instead of 300%), and yet the
audio still failed at the same number of listeners.
Brian.
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Re: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN
That depends if the call is answered and then you transfer it, you will HAVE to set the transfer_ringback variable you can't send a 180 to the thing or a progress and make it generate the ringback. You MUST do it yourself. You also fail to mention if the progress is a 180 or a 183 with sdp and media... or even better a 180 with sdp and media (silly sip people what were you thinking) either way... set the transfer_ringback variable. /b On Dec 18, 2009, at 4:00 AM, Peter P GMX wrote: Should I open a JIRA for this? Best regards Peter ___ 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] How can I join two freeswitch on two servers?
Hi, As far as I know, there are two ways to connect two freeswitch, by using ACL or using authentication. Also from this email history discussion, another solution is to create user in FS B directory,then treat server B as normal gateway by adding gateway definiton in FS A. So my question is how to connect FS A and FS B through ACL or through the way this email described. The information I pasted is about the last way. FS A: 192.168.129.168, caller id= 1001 FS B: 192.168.129.194, callee id= 1003, create 1101 for gateway configure In FS A add /conf/sip_proifles/external/gwfsa.xml include gateway name=gwfsa param name=username value=1101 param name=password value=1234 param name=proxy value=192.168.129.194:5060 param name=register value=false /gateway /include note: I delete and / for param cause it can't be displayed in this email. Both FS A and FS B are default configuration except creating id=1101 on FS B side. I'm confused if I connect two freeswitch by using ACLs, How do I confiugre data in both side ? Your kind help is highly appreciated. Seven Du wrote: I couldn't guess what you want, pastbin your full config and logs and give more detail of your story perhaps someone can help you. 2009/12/18 yvonne ding yhding2...@yahoo.ca: param name=username value=1101 param name=password value=1234 param name=proxy value=192.168.129.194:5060 param name=register value=false Hi, If I configure data as following, why FS A 1001 call FS B 1003 failed ? Thank you! FS A: 192.168.129.168, DN=1001 FS B: 192.168.129.194, DN=1003 In FS A add /conf/sip_proifles/external/gwfsa.xml include gateway name=gwfsa /gateway /include 1101 is configured in FS B /conf/directory/default/1101.xml, FS A don't have 1101 number Dan Le wrote: If you want FS server A to be able to call FS server B, you can set up a user account in server B's FS directory configs, and then just treat server B as a normal gateway by adding a gateway definition in server A. That will allow you to route calls to server B from A; to do the reverse, just mirror the configs the other direction. On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz darklio...@yahoo.com wrote: I like to connect two freeswitch, call each other, communicate and vice versa. Can you give me an example for that? Thanks -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 -- View this message in context: http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26832823.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 -- View this message in context: http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26844589.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] Presence across several networked FSs
I have found some ways to get presence, or rather BLF functions to work on Snom telephones in a distributed network with several FSs. I'll post a solution on the wiki when I have tested it further. Anyhow, I'm using the mod_event_multicast module with the following configuration: configuration name=event_multicast.conf description=Multicast Event settings param name=address value=225.1.1.1/ param name=port value=4242/ param name=bindings value=PRESENCE_IN CUSTOM sofia::register CUSTOM multicast::event/ /settings /configuration With this setting on all FSs, the registration table is also automatically updated thus listing all sets registered across all FSs. In the table sip_registrations (under the database for the profile used), the field status has the value: Registered if the UA is registered on another FS and the value Registered(UDP) if the UA is registered on the same FS. The field server_host, however, is the ip-address of local FS. Now comes the question: is there any way to let the field server_host show the server address of the server actually registered to? Or any other way using the existing modules to get the information about which FS the UAs are registered to? The information is going to be used for the routing decisions between networked FSs. /Jon ___ 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_xml_ldap compile issue.
On 12/18/2009 02:13 PM, Keith Laaks wrote: Hi, I am having an issue getting mod_xml_ldap to compile properly cut-cut making all mod_xml_cdr making all mod_xml_ldap Creating mod_xml_ldap.la... /usr/bin/ld: /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a: could not read symbols: Bad value collect2: ld returned 1 exit status cat: .libs/mod_xml_ldap.log: No such file or directory make[5]: *** [mod_xml_ldap.la] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_xml_ldap-all] Error 1 make[2]: *** [all-recursive] Error 1 I notice the openldap library has been bumped up to .19 - not sure if that may have anything to do with it. At revision 15995 on a 2.6.31-15-generic Ubuntu x86_64 GNU/Linux notebook. mod_ldap compiles OK, but mod_xml_ldap fails as per the above. What am I doing working here ? I had the same issue and MikeJ (one of the core developers) looked at it. Conclusion was that it is an openldap issue and iirc the solution is to libtoolize libraries/liblutil/Makefile.in so that when running configure a Makefile with proper compiler flags is generated in libraries/liblutil/ Patches welcome :) Regards, Patrick ___ 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_conference scalability
What is your dialplan on the secondary box?
On Dec 18, 2009, at 9:08 AM, Brian br...@proximosystems.com wrote:
I’ve got FS running on a 64 bit OS, and here is more info on the tes
t procedure.
I’ve got one server (primary) that hosts the speaker call (this is m
eant to be a primary conference with a few speakers, but my test sim
plifies this to just one speaker). I’ve got a second server (seconda
ry) that hosts the conference that all the listeners go into, and I
have two other servers that I use automate the listener calls. The g
oal is to have several secondary servers to scale the listener side
of things, but for this initial test I’ve only got one secondary ser
ver.
The primary server dials into the secondary conference server so
that the listeners can hear the speaker conference on the primary
server.
The automated listener servers start dialing into the listener
conference at a combined rate of 5 calls per second (i.e. 2.5 calls
per second each). The play an audio loop that represents noise on
their end, which since they are listeners, should be ignored anyway.
As I ramp up the automated listener calls, I manually call into the
conference from either my SIP phone, or from a land line using a DID
that I have directed to the conference.
All calls are using SIP with uLaw 8000hz codec. Also, I’ve set up th
e profile for the listener conference to disable many of the events:
profile name=listener
param name=domain value=$${domain}/
param name=rate value=8000/
param name=moh-sound value=moh.wav/
param name=suppress-events value=start-talking,stop-
talking,energy-level,volume-level,gain-level,mute-detect,energy-
level-member,volume-in-member,volume-out-member,lock,unlock,floor-
change/
param name=caller-controls value=listener_controls/
/profile
I do have caller controls for the listener, since in my production I
will need to generate and handle events for listener DTMF.
To compare FreeSWITCH vs Asterisk, I just swap out the secondary
conference server and everything else stays the same.
Brian.
From: Brian West [mailto:br...@freeswitch.org]
Sent: Thursday, December 17, 2009 5:20 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_conference scalability
What exactly are you doing I know it goes better than that.. are you
using 64bit?
/ b
On Dec 17, 2009, at 3:41 PM, Brian wrote:
I did a test with the trunk version for the one conference case, and
it is the same results as for 1.0.4. The audio failed at around 300
listeners. Oddly though, it consumed less %CPU (240% instead of
300%), and yet the audio still failed at the same number of listeners.
Brian.
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Re: [Freeswitch-users] ACLs through proxy
Hello Mathieu, I assumed that apply-proxy-acl was a modifier of auth-calls, so in my quick tests I just hard-coded the UA IP in the profile. param name=auth-calls value=true/ param name=apply-proxy-acl value=190.218.97.83/ !-- IP of UA -- And I get: 2009-12-18 09:14:28.250929 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 Rejected by user acl 190.218.97.83/32 Where 64.135.119.105 is the IP of my proxy. And actually this is a REGISTER, not an INVITE. I did a tcpdump, and I'm not seeing the X-AUTH-IP header in the register packet. I will be incommunicado for the rest of today, but when I get back online, I'll see if I can get my proxy to add the X-AUTH-IP to the REGISTER packet and see if that makes a difference. Thanks for your help! Bill Mathieu Rene wrote: From looking at sofia.c, if the ip address of the caller is in apply- proxy-acl, it'll look for the X-AUTH-IP header in the INVITE packet, and use that one for authentication. Is that what you did in your previous tests? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 17-Dec-09, at 11:02 PM, Bill W wrote: Hey Metik, Thanks for the reply, and the pointers for doing it with xml_curl. I'll guess have to do that in the short term, but in my opinion, having auth-acl be able to work through a proxy is very important as it is a vital part of a comprehensive security feature set. And it would be much simpler to implement from an end-user perspective than the alternative of doing it in xml_curl. As a matter of fact, I'm considering offering a bounty for that feature. What is the going rate for that kind of thing? Is anyone out there interested in coding this feature? Or chipping in for the bounty? Thanks, Bill Metik wrote: This may be difficult considering that ACL needs to consider the original src IP/URI. To do that it, freeswitch would need to do so using a header that retains that information (i.e. From, Via, Contact, etc.). Which I do not believe is currently possible using auth-acl or apply-proxy-acl. However, you should be able to emulate the behavior using mod_xml_curl (and validating against appropriate variables available when using it to authenticate the request). see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization -metik Bill W wrote: Hey Brian, I've been doing some testing and I am unable to get auth-calls to work through a proxy the way I want them to, even with setting apply-proxy-acl to either the endpoint IP or the proxy IP. I have a multi-tenant system with multiple domains with multiple users in each domain. And I want to restrict a user to an arbitrary CIDR and challenge them for a password. The arbitrary CIDR will vary from UA to UA, and is specified in the directory via the auth-acl parameter. TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of the proxy. Thanks, Bill Brian West wrote: it needs to be an ACL from acl.conf or a ip/cidr /b On Dec 17, 2009, at 5:41 AM, Bill W wrote: Okay, I added: param name=apply-proxy-acl value=true/ to my sofia profile and restarted sofia, and still no joy. I'm on FreeSWITCH Version 1.0.trunk (15764) I've got param name=auth-acl value=190.218.103.12/32/ param in the directory, but I'm still being rejected by the acl: 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 Rejected by user acl 190.218.103.12/32 Here's what I believe is the appropriate snippet of the debug output: http://pastebin.freeswitch.org/11531 Thoughts? Thanks, Bill ___ 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 ___ 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 ___ FreeSWITCH-users mailing list
Re: [Freeswitch-users] ACLs through proxy
You need to add that header manually in your OpenSIPS config, FreeSWITCH wont look in record-route/via to try to guess it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 18-Dec-09, at 10:53 AM, Bill W wrote: Hello Mathieu, I assumed that apply-proxy-acl was a modifier of auth-calls, so in my quick tests I just hard-coded the UA IP in the profile. param name=auth-calls value=true/ param name=apply-proxy-acl value=190.218.97.83/ !-- IP of UA -- And I get: 2009-12-18 09:14:28.250929 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 Rejected by user acl 190.218.97.83/32 Where 64.135.119.105 is the IP of my proxy. And actually this is a REGISTER, not an INVITE. I did a tcpdump, and I'm not seeing the X-AUTH-IP header in the register packet. I will be incommunicado for the rest of today, but when I get back online, I'll see if I can get my proxy to add the X-AUTH-IP to the REGISTER packet and see if that makes a difference. Thanks for your help! Bill Mathieu Rene wrote: From looking at sofia.c, if the ip address of the caller is in apply- proxy-acl, it'll look for the X-AUTH-IP header in the INVITE packet, and use that one for authentication. Is that what you did in your previous tests? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 17-Dec-09, at 11:02 PM, Bill W wrote: Hey Metik, Thanks for the reply, and the pointers for doing it with xml_curl. I'll guess have to do that in the short term, but in my opinion, having auth-acl be able to work through a proxy is very important as it is a vital part of a comprehensive security feature set. And it would be much simpler to implement from an end-user perspective than the alternative of doing it in xml_curl. As a matter of fact, I'm considering offering a bounty for that feature. What is the going rate for that kind of thing? Is anyone out there interested in coding this feature? Or chipping in for the bounty? Thanks, Bill Metik wrote: This may be difficult considering that ACL needs to consider the original src IP/URI. To do that it, freeswitch would need to do so using a header that retains that information (i.e. From, Via, Contact, etc.). Which I do not believe is currently possible using auth- acl or apply-proxy-acl. However, you should be able to emulate the behavior using mod_xml_curl (and validating against appropriate variables available when using it to authenticate the request). see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization -metik Bill W wrote: Hey Brian, I've been doing some testing and I am unable to get auth-calls to work through a proxy the way I want them to, even with setting apply-proxy-acl to either the endpoint IP or the proxy IP. I have a multi-tenant system with multiple domains with multiple users in each domain. And I want to restrict a user to an arbitrary CIDR and challenge them for a password. The arbitrary CIDR will vary from UA to UA, and is specified in the directory via the auth-acl parameter. TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of the proxy. Thanks, Bill Brian West wrote: it needs to be an ACL from acl.conf or a ip/cidr /b On Dec 17, 2009, at 5:41 AM, Bill W wrote: Okay, I added: param name=apply-proxy-acl value=true/ to my sofia profile and restarted sofia, and still no joy. I'm on FreeSWITCH Version 1.0.trunk (15764) I've got param name=auth-acl value=190.218.103.12/32/ param in the directory, but I'm still being rejected by the acl: 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 Rejected by user acl 190.218.103.12/32 Here's what I believe is the appropriate snippet of the debug output: http://pastebin.freeswitch.org/11531 Thoughts? Thanks, Bill ___ 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 ___ 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
Re: [Freeswitch-users] Voicemail-Email
oh really,
sendmail segfaults?
if another application is crashing you need to figure that out, whatever
used to work doesnt now so you need to figure out what it was and let us
know.
On Fri, Dec 18, 2009 at 3:51 AM, François Legal de...@thom.fr.eu.orgwrote:
I get the same result with sendmail. This used to work in 1.0.3 , and after
upgrading to 1.0.4 (then some snapshots) then 1.0.5pre8 and the problem is
still there.
François
On Thu, 17 Dec 2009 17:33:58 +0100, Oliver Schönbeck wrote:
Currently it is Version 1.0.trunk (15982)
*Von:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *Im Auftrag von *Brian West
*Gesendet:* Donnerstag, 17. Dezember 2009 17:17
*An:* freeswitch-users@lists.freeswitch.org
*Betreff:* Re: [Freeswitch-users] Voicemail-Email
What SVN rev. exactly?
/b
On Dec 17, 2009, at 10:13 AM, Oliver Schönbeck wrote:
Hello,
we are running freeswitch 1.0.trunk and are currently trying to get the
mod_voicemail to send the received messages to the user by using exim4 on a
debian machine.
So far we followed the instructions in the wiki article (
http://wiki.freeswitch.org/wiki/Mod_voicemail ).
I added some lines to the bash script to enable some kind of logging:
#! /bin/bash
typeset LOG=/tmp/${0##*/}.out
mv $LOG ${LOG}.old /dev/null 21
[[ -t 1 ]] echo Writing to logfile '$LOG'.
exec $LOG 21
exim4 -t -v $LOG
If I run the script from the command line everything is working as
expected. If the script gets called by freeswitch I get the following result
in my logfile:
/usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation
fault (core dumped) exim4 -t -v $LOG
Has anybody seen similar effects before?
Any advice whats going wrong is heavily appreciated.
Thanks
Oliver
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Re: [Freeswitch-users] mod_conference scalability
Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS. Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) François. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian br...@proximosystems.com wrote: I didn’t realize there was a policy about load testing questions. What forum should I have used for this? I didn’t get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production. On Thu, Dec 17, 2009 at 1:29 PM, Brian br...@proximosystems.com wrote: Hi Mike, I didn’t get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I’ve read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test – more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I’m doing wrong, but I don’t see what it could be. Brian. From: Michael Jerris [mailto:m...@jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I’m new to FreeSWITCH and I’m testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn’t scale nearly as well as I’d hoped (based on what I’ve read on how FreeSWITCH is supposed to be generally very scalable). Here’s my server
Re: [Freeswitch-users] mod_conference scalability
Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:8...@conference.freeswitch.orgsip%3a...@conference.freeswitch.orgThis is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde fdelawa...@wirelessmundi.com wrote: Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS. Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) François. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian br...@proximosystems.com wrote: I didn’t realize there was a policy about load testing questions. What forum should I have used for this? I didn’t get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production. On Thu, Dec 17, 2009 at 1:29 PM, Brian br...@proximosystems.com wrote: Hi Mike, I didn’t get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able
[Freeswitch-users] FreeSWITCH Weekly Conference Call Starting Shortly!
Hello everyone! Today's agenda is listed here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14 Also, we are going to be giving away goodies on some of the upcoming conferences, so call in and see what we've got in store. :) For the first 15 minutes we'll let everyone mingle and then we'll get into the agenda. Talk to you all soon! -Michael ___ 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] Ringing after call has been rejected
not answering it would be the best way. if you want to generate fake congestion you can use tone_stream:// or gentones On Fri, Dec 18, 2009 at 5:16 AM, bcxml bc...@hotmail.com wrote: I have an incomming call being answered by FreeSwitch and passed to IVR application which rejects the call. The call is never answered by FreeSwitch, but instead of hearing a busy signal, the caller hears ringing. Can anyone advise how I can get the user to hear a busy signal after call rejection instead of ringing. Here is the debug trace http://pastebin.freeswitch.org/11558 Thanks Brian -- View this message in context: http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:+19193869900 ___ 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] SIP Re-invite
I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ___ 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
[Freeswitch-users] Fwd: incoming call
Hi, i have up the freeswitch with domain(eg sipserver.domain.com) name instead of local ip, two clints are regitered with freeswitch using domain name(eg sipserver.domain.com), one client is making a call to other one, other clint receiving a invite request like this 173927 3120.658532 10.91.154.108 10.91.154.80 SIP/SDP Request: INVITE sip:1...@10.91.154.80:5061, with session description, but usually it should come with INVITE sip:1...@sipserver.domain.comsip%3a1...@sipserver.domain.com? what is changes i need to do for this? any idea? Regards-- Srinivasula Reddy K -- Srinivasula Reddy K ___ 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_conference scalability
It was of course just bad humor, I love both projects for what they are, and I agree that both have their own advantages and inconvenients. For example, accessing that same conference from a dahdi card could be another goal where Asterisk would be at an advantage, as chan_dahdi is still superior (in the more tested sense) than openzap+mod_openzap. I just use both projects separately or together depending on what's needed! I'm no banker nor do I understand the code, but many thanks for all those unpaid contributions providing an excellent alternative for free telephony. Your names really deserve being engraved in google's cache for eternity. :-) But still, I would like to see those numbers... François. On Fri, 2009-12-18 at 10:34 -0600, Anthony Minessale wrote: Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:8...@conference.freeswitch.org This is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde fdelawa...@wirelessmundi.com wrote: Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS. Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) François. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian br...@proximosystems.com wrote: I didn’t realize there was a policy about load testing questions. What forum should I have used for this? I didn’t get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re:
Re: [Freeswitch-users] ACLs through proxy
Honestly, several years ago I accomplished this by mod'ing SER (which became OpenSER which was then forked to OpenSIPS and Kamalio) and using one cluster of proxies for subscriber endpoints and another for infrastructure (so that I could keep RTP flows optimized yet support double NAT when required by an endpoint). Although the network looks different today. However, we were never quite happy about the lack of media failover (complicated NAT) and evaluated several commercial solutions until finding Covergence (which is now, for better or for worse since the jury is still out, owned by ACME Packet). At the time, they offered the best mix of security (their forte) yet scaled very well in comparison to their competitors that I had tested in our lab (ACME Packet, Kagoor, Netrake, NexTone, Kagoor, and Jasomi). In fact, they made a great decision, not unlike that of the FS developers, to implement a proven/stable SIP protocol stack. Nothing is perfect and that does not mean that we did not spend a considerable amount of time documenting bugs so that they could be addressed and it would work as it should We still use OpenSIPS for certain CSCF functionality (due to its speed and flexibility since it is not a B2BUA). Based on Mathieu's response (and he is definitely someone that would know), it looks like you should be able to easily append a X-AUTH-IP header (via OpenSIPS) containing the IP address of the endpoint and call it a day. -metik Bill W wrote: Hey Metik, That's exactly what I'm trying to do... load balance across multiple FS boxes, and have any machine in the cluster be able to reach a device behind a NAT firewall. Hence the need for the proxy. Also, I'm trying to keep the proxy relatively dumb and put all the logic in the FS boxes. True I could do the auth on the proxies as well, but then I'm setting up another authentication scheme in addition to what is on the FS boxes, and then integrating the databases so everything is consistent. I also have hosts that talk to the FS boxes directly, rather than through the proxy. So I can't get rid of auth_acl on FS either, even if I do implement it on the proxies. So my setup becomes much more complex and potentially brittle. And all we're really talking about for FreeSWITCH, conceptually speaking, is populating a variable with a different IP. We could even make it configurable, as to which IP is to be used for the auth-acl. What are you using for SBCs? (if you are allowed to divulge that) I'm currently using OpenSIPS for my proxy. Thanks, Bill Metik wrote: Why not simply implement this feature in the PROXY itself? FS has a pretty comprehensive security feature set for endpoints that directly register with it. Don't get me wrong, I do agree this is useful especially if you are going to be using your proxies to load balance across multiple FS boxes to create an ad-hoc cluster. I actually have session border controllers that have this feature and use it quite often. -metik Bill W wrote: Hey Metik, Thanks for the reply, and the pointers for doing it with xml_curl. I'll guess have to do that in the short term, but in my opinion, having auth-acl be able to work through a proxy is very important as it is a vital part of a comprehensive security feature set. And it would be much simpler to implement from an end-user perspective than the alternative of doing it in xml_curl. As a matter of fact, I'm considering offering a bounty for that feature. What is the going rate for that kind of thing? Is anyone out there interested in coding this feature? Or chipping in for the bounty? Thanks, Bill Metik wrote: This may be difficult considering that ACL needs to consider the original src IP/URI. To do that it, freeswitch would need to do so using a header that retains that information (i.e. From, Via, Contact, etc.). Which I do not believe is currently possible using auth-acl or apply-proxy-acl. However, you should be able to emulate the behavior using mod_xml_curl (and validating against appropriate variables available when using it to authenticate the request). see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization -metik Bill W wrote: Hey Brian, I've been doing some testing and I am unable to get auth-calls to work through a proxy the way I want them to, even with setting apply-proxy-acl to either the endpoint IP or the proxy IP. I have a multi-tenant system with multiple domains with multiple users in each domain. And I want to restrict a user to an arbitrary CIDR and challenge them for a password. The arbitrary CIDR will vary from UA to UA, and is specified in the directory via the auth-acl parameter. TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of the proxy. Thanks, Bill Brian West wrote: it needs to be an
Re: [Freeswitch-users] mod_conference scalability
yes, I understand. My reply was to the thread in general not directed at you =p On Fri, Dec 18, 2009 at 11:41 AM, François Delawarde fdelawa...@wirelessmundi.com wrote: It was of course just bad humor, I love both projects for what they are, and I agree that both have their own advantages and inconvenients. For example, accessing that same conference from a dahdi card could be another goal where Asterisk would be at an advantage, as chan_dahdi is still superior (in the more tested sense) than openzap+mod_openzap. I just use both projects separately or together depending on what's needed! I'm no banker nor do I understand the code, but many thanks for all those unpaid contributions providing an excellent alternative for free telephony. Your names really deserve being engraved in google's cache for eternity. :-) But still, I would like to see those numbers... François. On Fri, 2009-12-18 at 10:34 -0600, Anthony Minessale wrote: Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:8...@conference.freeswitch.orgsip%3a...@conference.freeswitch.orgThis is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde fdelawa...@wirelessmundi.com wrote: Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS. Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) François. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian br...@proximosystems.com wrote: I didn’t realize there was a policy about load testing questions. What forum should I have used for this? I didn’t get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks,
Re: [Freeswitch-users] SIP Re-invite
Mike, There are 2 traces in there. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ___ 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
Re: [Freeswitch-users] LUA and return variables
On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij vi...@fx-services.com wrote:
Hi guys (and girls)!
I'm working on a little bit of ENUM trickery and I tried doing some
(illegal) nested conditions. :-)
What I want to do is to first check enum with the ENUM application,
then depending on the answer do some stuff. Say that the domain part
of the ENUM answer is robin.nl, then I want to do action X instead of
just briding the enum answer directly as I see in most examples.
But I remembered that it wasn't allowed to do nested conditions. So
what I did was stacked conditions. After that I read the dialplan wiki
pages again and figured that my regexp never matches because variables
I set during some phase of the extension I can't use in the same
go as another condition. So, now my plan is to use LUA to do the
regexp.
I'll pass the enum answer to a lua script which will split the answer
in a user and domain part and return those two to the main app. Then
based on those two vars I'll do routing or other actions (like, prefix
and then route).
Is this how I'm supposed to do it? I can't find many examples on
manipulating ENUM answers, other than bridging them directly. I can't
change the way I do stuff to ENUM answers, because in most cases I'll
just route them out the standard way.
Anyone with experience on fiddling with ENUM answers?
One thing you can do is create an extension that does the enum look up and
then transfers the call back into the dialplan. You could set up a separate
context that handles just the enum checking. Your condition would just need
to match whatever var you put the enum return val in. So if your var name is
enum_res then you can transfer like this after your enum lookup:
action application=transfer data=${enum_res} XML my_enum_context/
Then create a context named my_enum_context and match for the condition(s)
you need, like:
condition field=${enum_res} expression=robin\.nl
do stuff
/condition
Then have a different extension for other values of enum_res...
This is just one way to do it without using a scripting lang. you can by all
means use Lua as well.
-MC
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Re: [Freeswitch-users] SIP Re-invite
Mike, There are 2 traces in there. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ___ 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
[Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header
Is it possible to allow/deny REGISTER requests based on the User-Agent header? I need to know/manage what devices are registering. Best Regards, Jerry ___ 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_conference scalability
I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. My scenario is not a hypothetical one of having robots call the conference in a way that probably does not match reality. In fact, this will very much reflect the reality of the application Im building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. Im trying to find a real solution to a real problem. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Friday, December 18, 2009 11:34 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org This is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde fdelawa...@wirelessmundi.com wrote: Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS. Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) François. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian br...@proximosystems.com wrote: I didnt realize there was a policy about load testing questions. What forum should I have used for this? I didnt get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you
Re: [Freeswitch-users] LUA and return variables
I use a similar method (transfer to XML dialplan based on the value of
${enum_route_1}) to determine if the SIP URI is native to a particular
FS instance or if it needs to be sent off-net and it works well.
-metik
Michael Collins wrote:
On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij vi...@fx-services.com
mailto:vi...@fx-services.com wrote:
Hi guys (and girls)!
I'm working on a little bit of ENUM trickery and I tried doing some
(illegal) nested conditions. :-)
What I want to do is to first check enum with the ENUM application,
then depending on the answer do some stuff. Say that the domain part
of the ENUM answer is robin.nl http://robin.nl, then I want to
do action X instead of
just briding the enum answer directly as I see in most examples.
But I remembered that it wasn't allowed to do nested conditions. So
what I did was stacked conditions. After that I read the dialplan wiki
pages again and figured that my regexp never matches because variables
I set during some phase of the extension I can't use in the same
go as another condition. So, now my plan is to use LUA to do the
regexp.
I'll pass the enum answer to a lua script which will split the answer
in a user and domain part and return those two to the main app. Then
based on those two vars I'll do routing or other actions (like, prefix
and then route).
Is this how I'm supposed to do it? I can't find many examples on
manipulating ENUM answers, other than bridging them directly. I can't
change the way I do stuff to ENUM answers, because in most cases I'll
just route them out the standard way.
Anyone with experience on fiddling with ENUM answers?
One thing you can do is create an extension that does the enum look up
and then transfers the call back into the dialplan. You could set up a
separate context that handles just the enum checking. Your condition
would just need to match whatever var you put the enum return val in.
So if your var name is enum_res then you can transfer like this
after your enum lookup:
action application=transfer data=${enum_res} XML my_enum_context/
Then create a context named my_enum_context and match for the
condition(s) you need, like:
condition field=${enum_res} expression=robin\.nl
do stuff
/condition
Then have a different extension for other values of enum_res...
This is just one way to do it without using a scripting lang. you can
by all means use Lua as well.
-MC
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Re: [Freeswitch-users] mod_conference scalability
Brian, Now that you know the scale freeswotch scales to in you scenario, and having designed a mult-server solution can you not add more server to scale further? As freeswitch continues to improve retest and revise your architecture design. Sent from my iPhone On Dec 18, 2009, at 11:14 AM, Brian br...@proximosystems.com wrote: I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. My scenario is not a hypothetical one of “having robots call the con ference in a way that probably does not match reality”. In fact, thi s will very much reflect the reality of the application I’m building . Only instead of 300 listeners, I need to scale to over 2000 listen ers minimum – per event, with possibly more than one concurrent even t. I want to pack as many listeners on one server as I can. I’m tryi ng to find a real solution to a real problem. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Friday, December 18, 2009 11:34 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:8...@conference.freeswitch.org This is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde fdelawa...@wirelessmundi.co m wrote: Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS. Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) François. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian br...@proximosystems.com wrote: I didn’t realize there was a policy about load testing questions. What forum should I have used for this? I didn’t get the chance to test on FS trunk yet, but when I do I w ill provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of
Re: [Freeswitch-users] LUA and return variables
On 12/18/09 7:18 PM, Michael Collins wrote: Hi Michael, One thing you can do is create an extension that does the enum look up and then transfers the call back into the dialplan. You could set up a Cool idea, didn't think about that! separate context that handles just the enum checking. Your condition would just need to match whatever var you put the enum return val in. So if your var name is enum_res then you can transfer like this after your enum lookup: Right, makes sense. Going to try a bit in that direction. Do the enum lookup and then transfer to an enum handling context. Simple, should have thought about that. :) This is just one way to do it without using a scripting lang. you can by all means use Lua as well. My main question there really was, since I'm not able to work on vars I set in an extention, will that work if I return vars from a script? It should really, but I was asking to make sure it would. I think in that design, the script would have been like three rules or something, but keeping it in the dialplan is nicer, I think (even though it says don't do magic in the dialplan, do it in scripts on the wiki). I'll report back when I managed to fiddle something together. /Robin ___ 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_conference scalability
Brian, there was not one insulting word in anything I have said and as this is a community mailing list my replies are always voiced to address the public in general not you specifically, like I already mentioned in my last post. If you open a public forum on a FAQ be prepared to hear our policy. Indeed many people do unrealistic load testing and most people with strong will find it insulting when a group of people have a set of standard policy by which they try to deal with making a penny jar for all the 2 cents worth of input we get on a daily basis. I can't begin to iterate over all the cases we endure on a weekly basis. additionally 90% of bug reports are on older releases and we always make people reproduce their issues on SVN trunk because 3 core devs and a handful of helpers can't maintain 20 versions of the code. I gave you some really suggestions yesterday let me repaste it, I fail to see any insults: --- What exactly is your test process? you should try increasing the interval in the conference profile to a bigger time slice maybe 30 40 or 60ms you could also increase the ptime to match as well. like brian said you could use mod_shout to broadcast the single speaker to icecast and let people listen with itunes/winamp --- I have to get in these fights with people constantly so I guess that is part of my job and my biggest mistake is spending so much time trying to explain myself. - Show quoted text - On Fri, Dec 18, 2009 at 1:33 PM, Michael Collins m...@freeswitch.org wrote: On Fri, Dec 18, 2009 at 11:14 AM, Brian br...@proximosystems.com wrote: I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of load testing or researching a new solution which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) My scenario is not a hypothetical one of “having robots call the conference in a way that probably does not match reality”. In fact, this will very much reflect the reality of the application I’m building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum – per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I’m trying to find a real solution to a real problem. That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
Re: [Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header
could be possible with a code change, open a bounty on jira and someone may do it On Fri, Dec 18, 2009 at 12:35 PM, Jerry Richards jerry.richa...@teotech.com wrote: Is it possible to allow/deny REGISTER requests based on the User-Agent header? I need to know/manage what devices are registering. Best Regards, Jerry ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:+19193869900 ___ 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] Ringing after call has been rejected
Try the following: action application=hangup data=USER_BUSY/ I don't know whether it will work in your case, but here we use it to reject a call while we want to signal that the remote party is busy. Regards, __Yehavi: 2009/12/18 bcxml bc...@hotmail.com I have an incomming call being answered by FreeSwitch and passed to IVR application which rejects the call. The call is never answered by FreeSwitch, but instead of hearing a busy signal, the caller hears ringing. Can anyone advise how I can get the user to hear a busy signal after call rejection instead of ringing. Here is the debug trace http://pastebin.freeswitch.org/11558 Thanks Brian -- View this message in context: http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] mod_conference scalability
Hi Michael, Thanks for the invite, but I can't make it on the call. Anyway, I'm not sure if discussing my specific case is meant for that type of call, is it? After Brian's suggestion to use shoutcast and local streams, I was looking at the code for those modules. I'm not familiar with shoutcast or icecast capabilities, so I don't know if they can just pass though my audio stream unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on the source server, and then back from mp3 to uLaw (or whatever phone codec) on the other server. I was wondering if maybe there was a way to make a stream out of an existing channel, and have all the other channels just listen to that stream. It would be sort of halfway between conference and shoutcast. I would call in to the secondary server like I already do, but only instead of entering into a conference as a speaker, the channel would just start producing a local audio stream for the listener channels to tap into. It would avoid the need to have another piece of software to manage (shoutcast or icecast), and my support team would be happier... However, I would still need to do tests for the streaming idea to see how that scales... Brian. From: Michael Collins [mailto:m...@freeswitch.org] Sent: Friday, December 18, 2009 2:33 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability On Fri, Dec 18, 2009 at 11:14 AM, Brian br...@proximosystems.com wrote: I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of load testing or researching a new solution which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) My scenario is not a hypothetical one of having robots call the conference in a way that probably does not match reality. In fact, this will very much reflect the reality of the application I'm building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum - per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I'm trying to find a real solution to a real problem. That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael ___ 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_conference scalability
I am more than sure there is probably plenty of room for conference optimizations it's just a big task. We don't have a test labbed up and an urgency to work on it. If you really want us to pursue trying to improve the performance perhaps you can contact us at consult...@freeswitch.org and provide us with access your test environment and let us investigate the possibility of making improvements. On Fri, Dec 18, 2009 at 2:16 PM, Brian br...@proximosystems.com wrote: Hi Michael, Thanks for the invite, but I can’t make it on the call. Anyway, I’m not sure if discussing my specific case is meant for that type of call, is it? After Brian’s suggestion to use shoutcast and local streams, I was looking at the code for those modules. I’m not familiar with shoutcast or icecast capabilities, so I don’t know if they can just pass though my audio stream unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on the source server, and then back from mp3 to uLaw (or whatever phone codec) on the other server. I was wondering if maybe there was a way to make a stream out of an existing channel, and have all the other channels just listen to that stream. It would be sort of halfway between conference and shoutcast. I would call in to the secondary server like I already do, but only instead of entering into a conference as a speaker, the channel would just start producing a local audio stream for the listener channels to tap into. It would avoid the need to have another piece of software to manage (shoutcast or icecast), and my support team would be happier... However, I would still need to do tests for the streaming idea to see how that scales... Brian. *From:* Michael Collins [mailto:m...@freeswitch.org] *Sent:* Friday, December 18, 2009 2:33 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] mod_conference scalability On Fri, Dec 18, 2009 at 11:14 AM, Brian br...@proximosystems.com wrote: I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of load testing or researching a new solution which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) My scenario is not a hypothetical one of “having robots call the conference in a way that probably does not match reality”. In fact, this will very much reflect the reality of the application I’m building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum – per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I’m trying to find a real solution to a real problem. That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
Re: [Freeswitch-users] mod_conference scalability
Hi Brian, Have a look at this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop - I took a quick look through the code and couldn't see any reason why you shouldn't have a bunch of eavesdroppers listening to a single caller. I'd be surprised if this didn't perform a lot better for your application. Cheers -- Dave I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. My scenario is not a hypothetical one of “having robots call the conference in a way that probably does not match reality”. In fact, this will very much reflect the reality of the application I’m building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum – per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I’m trying to find a real solution to a real problem. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Friday, December 18, 2009 11:34 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:8...@conference.freeswitch.org This is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde fdelawa...@wirelessmundi.com wrote: Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS. Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) François. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian br...@proximosystems.com wrote: I didn’t realize there was a policy about load testing questions. What forum should I have used for this? I didn’t get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's
Re: [Freeswitch-users] SIP Re-invite
Thank you Mike for your suggestion on IRC. We did what you recommend and found out it's the iptables issue that we thought it was not there at the beginning since we saw the first 2 invites from the far end fine, but somehow it has something to do with the 3rd invite. I did close the Jira that I thought it was a bug. Thank you again for the community and your support. Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ___ 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
Re: [Freeswitch-users] mod_conference scalability
Sounds like a plan. We will pursue it through the consult...@freeswith.org route. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Friday, December 18, 2009 3:30 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I am more than sure there is probably plenty of room for conference optimizations it's just a big task. We don't have a test labbed up and an urgency to work on it. If you really want us to pursue trying to improve the performance perhaps you can contact us at consult...@freeswitch.org and provide us with access your test environment and let us investigate the possibility of making improvements. On Fri, Dec 18, 2009 at 2:16 PM, Brian br...@proximosystems.com wrote: Hi Michael, Thanks for the invite, but I can't make it on the call. Anyway, I'm not sure if discussing my specific case is meant for that type of call, is it? After Brian's suggestion to use shoutcast and local streams, I was looking at the code for those modules. I'm not familiar with shoutcast or icecast capabilities, so I don't know if they can just pass though my audio stream unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on the source server, and then back from mp3 to uLaw (or whatever phone codec) on the other server. I was wondering if maybe there was a way to make a stream out of an existing channel, and have all the other channels just listen to that stream. It would be sort of halfway between conference and shoutcast. I would call in to the secondary server like I already do, but only instead of entering into a conference as a speaker, the channel would just start producing a local audio stream for the listener channels to tap into. It would avoid the need to have another piece of software to manage (shoutcast or icecast), and my support team would be happier... However, I would still need to do tests for the streaming idea to see how that scales... Brian. From: Michael Collins [mailto:m...@freeswitch.org] Sent: Friday, December 18, 2009 2:33 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability On Fri, Dec 18, 2009 at 11:14 AM, Brian br...@proximosystems.com wrote: I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of load testing or researching a new solution which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) My scenario is not a hypothetical one of having robots call the conference in a way that probably does not match reality. In fact, this will very much reflect the reality of the application I'm building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum - per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I'm trying to find a real solution to a real problem. That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com
Re: [Freeswitch-users] packaging preference question
I think that sounds like a good idea. It would also keep permission management simple. -William King On Fri, 2009-12-18 at 17:22 -0500, Frank Carmickle wrote: Hello all The packaging folk are interested in knowing if anyone has a problem with having the install set up the user and group to freeswitch:freeswitch. This would be the default on debs rpms and ports packaging. The freeswitch user would be added to daemon and audio groups. The FusionPBX packaging can then add www-data/apache to the freeswitch group. Any objections? --FC ___ 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
[Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter)
So, it's been a while since I mentioned this project, but its finally nearing the point where it's going to be able to go into production (and replace my old asterisk-based platform) so I decided to dredge it up again. Briefly, spice telephony is a call/contact center platform that leverages FS for VoIP, IVRs, call recording, etc. It also supports handing email/voicemail contacts (chat is planned, too). Here's some features: * Skill based routing * Priority, unified queues * Web based administration, agent interaction (using the dojo toolkit) * Supervisory drag and drop interface for managing agents/call flow * Queue 'recipes' - ability to play announcements, send to voicemail, modify skills or priority based on certain conditions (queue time, media type, hour of day, # of available agents, etc). * Integration API for importing agents/clients out of a CRM/AD/whatever * Detailed CDRs recording every step of a call (IVR, Queue, Ring, Transfer, Wrapup, etc). The project is implemented in Erlang (erlang.org) and thus allows spice-telephony to be deployed as a distributed system (multiple nodes aggregated into a single system). Calls can come into any node and, skills permitting, can be offered to any agent on the local node or any of the remote nodes. Nodes can also operate independantly if isolated by a netsplit or simply deployed standalone. CDRs and config files are stored in erlang's distributed database, mnesia, and CDRs can be output in parallel to any node configured to do so (so you can have all your call data in multiple places without having to do SQL replication). Erlang's fault tolerant nature also allows the platform to be very robust, entire subsystems can fail at runtime and be automatically restarted by supervisor process, and the entire erlang node can be automatically restarted if the node crashes. There's a lot more than mentioned above, so I'd encourage anyone interested to grab the latest release from: http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz and look at the install guide: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony_Install_Guide You'll need an erlang version = R13B01 and ruby's 'rake' installed, you shouldn't need much of anything else. It *does* work on windows but I don't recommend it (I can try to help you get it working though). There's also some more information available here: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony The documentation is a little sparse, but I'll do my best to answer any questions. Any feedback is appreciated. Andrew ___ 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] [ANN] Spice Telephony 0.9.5 released (FS based callcenter)
I've been asked to provide some screenshots, so here's some of the agent/supervisor interface: http://eagle.bsd.st/~andrew/cpxshots/ Hopefully the image names are self-explanatory. In the ringing picture, that URL pop is a configurable URL that can be used to integrate with a CRM, in my case our own CRM - spicecsm. The URL supports interpolation for variables like callerid, clientid, call type, etc. The supervisor view is a little hard to describe via static images, but you're able to drag and drop agents into another profile (empty profiles are hidden when not dragging an agent), drag agents onto an agent to send them the call, and there's also various right click menus available. Oh, and I forgot to mention this before; this system is in 'live testing' and the goal is to do a final deployment sometime in January. Andrew ___ 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] ACLs through proxy
Hey Metik,
Thanks so much for your insights and your help. And yes, I was able to
append the X-AUTH-IP header with no problem. But that didn't solve the
issue. After some more research, it appears that the problem isn't with
auth-calls at all.
I disabled all auth-call directives in every sip profile and the
registration through a proxy is still being rejected.
I looked in sofia_reg.c and if auth_acl is defined, sofia_reg checks the
ip variable against the auth_acl cidr.
if (auth_acl) {
if (!switch_check_network_list_ip(ip, auth_acl)) {
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_WARNING, IP %s Rejected by user acl %s\n, ip, auth_acl);
ret = AUTH_FORBIDDEN;
goto end;
}
So I guess the question is, is it possible to control what gets put into
the ip variable?
Thanks,
Bill
Metik wrote:
Honestly, several years ago I accomplished this by mod'ing SER (which
became OpenSER which was then forked to OpenSIPS and Kamalio) and using
one cluster of proxies for subscriber endpoints and another for
infrastructure (so that I could keep RTP flows optimized yet support
double NAT when required by an endpoint). Although the network looks
different today.
However, we were never quite happy about the lack of media failover
(complicated NAT) and evaluated several commercial solutions until
finding Covergence (which is now, for better or for worse since the jury
is still out, owned by ACME Packet). At the time, they offered the best
mix of security (their forte) yet scaled very well in comparison to
their competitors that I had tested in our lab (ACME Packet, Kagoor,
Netrake, NexTone, Kagoor, and Jasomi). In fact, they made a great
decision, not unlike that of the FS developers, to implement a
proven/stable SIP protocol stack. Nothing is perfect and that does not
mean that we did not spend a considerable amount of time documenting
bugs so that they could be addressed and it would work as it should
We still use OpenSIPS for certain CSCF functionality (due to its speed
and flexibility since it is not a B2BUA).
Based on Mathieu's response (and he is definitely someone that would
know), it looks like you should be able to easily append a X-AUTH-IP
header (via OpenSIPS) containing the IP address of the endpoint and call
it a day.
-metik
Bill W wrote:
Hey Metik,
That's exactly what I'm trying to do... load balance across multiple FS
boxes, and have any machine in the cluster be able to reach a device
behind a NAT firewall. Hence the need for the proxy. Also, I'm trying
to keep the proxy relatively dumb and put all the logic in the FS boxes.
True I could do the auth on the proxies as well, but then I'm setting up
another authentication scheme in addition to what is on the FS boxes,
and then integrating the databases so everything is consistent.
I also have hosts that talk to the FS boxes directly, rather than
through the proxy. So I can't get rid of auth_acl on FS either, even if
I do implement it on the proxies. So my setup becomes much more
complex and potentially brittle.
And all we're really talking about for FreeSWITCH, conceptually
speaking, is populating a variable with a different IP. We could even
make it configurable, as to which IP is to be used for the auth-acl.
What are you using for SBCs? (if you are allowed to divulge that) I'm
currently using OpenSIPS for my proxy.
Thanks,
Bill
Metik wrote:
Why not simply implement this feature in the PROXY itself?
FS has a pretty comprehensive security feature set for endpoints that
directly register with it.
Don't get me wrong, I do agree this is useful especially if you are
going to be using your proxies to load balance across multiple FS boxes
to create an ad-hoc cluster. I actually have session border controllers
that have this feature and use it quite often.
-metik
Bill W wrote:
Hey Metik,
Thanks for the reply, and the pointers for doing it with xml_curl.
I'll guess have to do that in the short term, but in my opinion, having
auth-acl be able to work through a proxy is very important as it is a
vital part of a comprehensive security feature set. And it would be
much simpler to implement from an end-user perspective than the
alternative of doing it in xml_curl.
As a matter of fact, I'm considering offering a bounty for that feature.
What is the going rate for that kind of thing?
Is anyone out there interested in coding this feature? Or chipping in
for the bounty?
Thanks,
Bill
Metik wrote:
This may be difficult considering that ACL needs to consider the
original src IP/URI. To do that it, freeswitch would need to do so
using a header that retains that information (i.e. From, Via, Contact,
etc.). Which I do not believe is currently possible using auth-acl or
apply-proxy-acl.
However,
[Freeswitch-users] Park with Pre Answer
Is there any way to park a channel without causing pre-answer (resulting is a SIP 183 Session Progress)? Thanks, Ron ___ 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
