Re: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database
Hi
Look at Contrib of source
http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/diegoviola/
some pre-paid examples
Ram
On Wed, Dec 16, 2009 at 12:27 AM, Senaka Amarakeerthi
senaka...@gmail.comwrote:
Dear Sir,
I have successfully installed freeSWITCH and it works fine in passthrough
mode. I installed nibblebill and it deduct money from the accounts database
and it works fine. but I have two problems.
1. Calls can be initiated even though there is a minus value in accounts
database
2. Calls doesn't hangup when it goes to minus values.
Any answers are greatly appreciated.
This is my dialplan:
action application=nibblebill data=flush/
extension name=hangup
condition field=destination_number expression=^(hangup)$
action application=playback data=no_more_funds.wav/
action application=hangup/
/condition
/extension
extension name=Omega_Out
condition field=caller_id_number expression=^(\d{4})$/
condition field=destination_number expression=^(\d{11})$
action application=set data=nibble_rate=0.0448/
action application=set data=nibble_account=${accountcode}/
action application=set data=bypass_media=true/
action application=bridge
data={absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1/
/condition
/extension
This is the configuration file;
configuration name=nibblebill.conf description=Nibble Billing
settings
!-- See http://wiki.freeswitch.org/index.php?title=Mod_nibblebill for
help with these options --
!-- Information for connecting to your database --
!-- The database table where your CASH column is located --
!-- The column name where we store the value of the account --
!-- The column name for the unique ID identifying the account --
!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
bill only at end of call) --
!-- By default, warn a caller when their balance is at $5.00. You can
set this to a negative number. --
!-- By default, terminate a caller when their balance hits $0.00. You
can set this to a negative number. --
!-- If a call goes beyond a certain dollar amount, flag or terminate it
--
/settings
/configuration
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[Freeswitch-users] BLF on Grandstream GXP2020
Hallo All! I need information about setup BLF on GXP2010/2020 phones with Freeswitch. I search in Freeswitch Wiki and maillist archives but find no usable information. ___ 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 to set the Session Name on a SDP?
I just found that this is related to the username of the profile. It needs to be set as parameter. Oscav wrote: Hi, Is it possible to set (rewrite) the Session Name in the SDP of a 183 progress sent to inbound ? Many thanks -- View this message in context: http://old.nabble.com/How-to-set-the-Session-Name-on-a-SDP--tp26815554p26826579.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] ACLs through proxy
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 Brian West wrote: use apply-proxy-acl on the sofia profile. /b On Dec 15, 2009, at 10:58 PM, Bill W wrote: However, having the proxy in the path effectively negates using IP based ACLS. ___ 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 nibblebill deduct money but no hangup at zero and can call without money in database
Dear Ram,
Thank you for the reply. To work with your code I hope that Mod cdr
should be there. But wiki says that its not functional. What should I
do.
Thanks
Senaka
On Thu, Dec 17, 2009 at 7:29 PM, ram talk2...@gmail.com wrote:
Hi
Look at Contrib of source
http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/diegoviola/
some pre-paid examples
Ram
On Wed, Dec 16, 2009 at 12:27 AM, Senaka Amarakeerthi senaka...@gmail.com
wrote:
Dear Sir,
I have successfully installed freeSWITCH and it works fine in passthrough
mode. I installed nibblebill and it deduct money from the accounts
database
and it works fine. but I have two problems.
1. Calls can be initiated even though there is a minus value in accounts
database
2. Calls doesn't hangup when it goes to minus values.
Any answers are greatly appreciated.
This is my dialplan:
action application=nibblebill data=flush/
extension name=hangup
condition field=destination_number expression=^(hangup)$
action application=playback data=no_more_funds.wav/
action application=hangup/
/condition
/extension
extension name=Omega_Out
condition field=caller_id_number expression=^(\d{4})$/
condition field=destination_number expression=^(\d{11})$
action application=set data=nibble_rate=0.0448/
action application=set data=nibble_account=${accountcode}/
action application=set data=bypass_media=true/
action application=bridge
data={absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1/
/condition
/extension
This is the configuration file;
configuration name=nibblebill.conf description=Nibble Billing
settings
!-- See http://wiki.freeswitch.org/index.php?title=Mod_nibblebill for
help with these options --
!-- Information for connecting to your database --
!-- The database table where your CASH column is located --
!-- The column name where we store the value of the account --
!-- The column name for the unique ID identifying the account --
!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
bill only at end of call) --
!-- By default, warn a caller when their balance is at $5.00. You can
set this to a negative number. --
!-- By default, terminate a caller when their balance hits $0.00. You
can set this to a negative number. --
!-- If a call goes beyond a certain dollar amount, flag or terminate
it
--
/settings
/configuration
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Re: [Freeswitch-users] Sofia performance
with the scenario below can we get the better performance: We create one profile for incoming call listening on 5060 as profile1 we create two profile for outgoing calls as profile2 on 5050 and profile3 on 5051 now we are receiving all calls on profile1:5060, but while bridging them to vendors we divide them, half to profile2:5050 and half to profile3:5051, something like: action application=bridge data=sofia/profile2/x...@x.x.x.x/ action application=bridge data=sofia/profile3/x...@x.x.x.x/ Will it make any difference? Thanks On Sun, Dec 13, 2009 at 11:37 PM, Anthony Minessale anthony.miness...@gmail.com wrote: Sep processes does better than sep profiles. We need to push the sofia devs to work on a better concurrancy scheme but they are too busy with other nokia duties these days so were stuck with what we got for now. About 400cps on a good day On Dec 13, 2009 4:05 PM, Jay Binks jaybi...@gmail.com wrote: I'm interested in what the upper limit would be, when expecting a performance improvement with sofia profiles. For example let's say I were to direct connect to customers ( layer 2 ) with a .1q trunk coming in to fs and a Sofia profile for each customer. Am I going to hit a bottleneck at 20,50,100,500 ??? Guess it's hardware limited , but any thoughts ? J On 14/12/2009, at 4:36, Anthony Minessale anthony.miness...@gmail.com wrote: Here is my standa... ___ 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
Re: [Freeswitch-users] Scanning my firewall for open UDP ports?
I don't have access to a remote computer from which I could log on and run nmap. I'll see if I can get a shell access somewhere. Thank you. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26827581.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] Scanning my firewall for open UDP ports?
Just for your information there is a version of nmap for windows. So you can do the test from your desktop. Оригинално писмо От: Fred-145 Относно: Re: [Freeswitch-users] Scanning my firewall for open UDP ports? До: freeswitch-users@lists.freeswitch.org Изпратено на: Четвъртък, 2009, Декември 17 14:54:49 EET I don't have access to a remote computer from which I could log on and run nmap. I'll see if I can get a shell access somewhere. Thank you. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26827581.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_shout.so: undefined symbol: ogg_sync_wrote
Hi Mike, This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can setup ssh access for you to check things out. In case this wasn't apparent I am trying to install FS from trunk. Thanks, Neil On Wed, Dec 16, 2009 at 9:14 PM, Michael Jerris m...@jerris.com wrote: strange, can someone file a bug on this on jira.freeswitch.org and contact me off list with ssh info so I can troubleshoot this on your box. Thanks Mike On Dec 16, 2009, at 9:56 AM, Neil Patel wrote: I'm also experiencing this problem, and I have verified I have libogg, libvorbis, and their dev packages installed. I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not listed in the dependency lib list. Is this related? -Neil On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris m...@jerris.com wrote: looks like ogg devel packages are installed but ogg lib is not? On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: FreeSWITCH seems to be unable to read MP3 files, citing that it's an unknown format. Looking through the log, I found this during startup: 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_sync_wrote** There don't seem to be any compile-time errors, yet I can't seem to eliminate this issue. Any help would be appreciated. ___ 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
Re: [Freeswitch-users] ACLs through proxy
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
Re: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote
Works on my CentOS 5.4 box just fine... /b On Dec 17, 2009, at 7:34 AM, Neil Patel wrote: Hi Mike, This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can setup ssh access for you to check things out. In case this wasn't apparent I am trying to install FS from trunk. Thanks, Neil ___ 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] detecting rtp packet for zombie channels
We need more info... svn rev, gcore, back trace and what not... please see the reporting bugs link on the wiki. http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Dec 16, 2009, at 11:53 PM, Juan Backson wrote: Hi I have rtp-timeout-sec set to 300 s but I am still getting calls with duration of 1 day long. Is there any other ways to check for zombie channels? jb ___ 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 to set the Session Name on a SDP?
Why are you needing to change it? /b On Dec 17, 2009, at 5:21 AM, Oscav wrote: I just found that this is related to the username of the profile. It needs to be set as parameter. ___ 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 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 setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I’ve set file logging to “notice” level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from “top”, it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. ___ 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] Polycom 501 conferencing with FreeSwitch
Its software, anything is possible with enough time and effort. Mike On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: After some discussions with Polycom support it seems that their conferencing support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the latest and is not compatible with the latest one). Any idea whether it is possible to program Freeswitch to support this draft? Thanks, __Yehavi: ___ 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
if you don't see it in sofia siptrace but do see it in tcpdump capture then something very ugly is going on. Either sofia has hung up completely and is not listening on that port anymore (can other calls go through?) or the packet you see in tcpdump is not really going to the right port. Can you confirm which one? Mike On Dec 16, 2009, at 6:29 PM, DJB wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. ___ 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
Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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
[Freeswitch-users] Small delay in registration validity
It seems to me, in previous revisions of FS, we could successfully call a registered user as soon as his terminal gets 200 OK for REGISTER. But after testing recent revisions, it seems we must wait a little (I wait 1 second) otherwise a call to bridge would end with this: 2009-12-17 22:46:14.155163 [ERR] switch_ivr_originate.c:2249 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] Similar thing is happening when the terminal unregisters: after unregistration an immediate call to bridge sofia/profile/user%domain will succeed. Has anything changed recently in the way registration works that could explain this? br, takeshi ___ 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
Is the packet capture running on the FS box itself? On Thu, Dec 17, 2009 at 9:36 AM, Michael Jerris m...@jerris.com wrote: if you don't see it in sofia siptrace but do see it in tcpdump capture then something very ugly is going on. Either sofia has hung up completely and is not listening on that port anymore (can other calls go through?) or the packet you see in tcpdump is not really going to the right port. Can you confirm which one? Mike On Dec 16, 2009, at 6:29 PM, DJB wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. ___ 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] mod_shout.so: undefined symbol: ogg_sync_wrote
if you contact me offlist, or better, join #freeswitch on irc.freenode.net and ping me (MikeJ) Mike On Dec 17, 2009, at 8:34 AM, Neil Patel wrote: Hi Mike, This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can setup ssh access for you to check things out. In case this wasn't apparent I am trying to install FS from trunk. Thanks, Neil On Wed, Dec 16, 2009 at 9:14 PM, Michael Jerris m...@jerris.com wrote: strange, can someone file a bug on this on jira.freeswitch.org and contact me off list with ssh info so I can troubleshoot this on your box. Thanks Mike On Dec 16, 2009, at 9:56 AM, Neil Patel wrote: I'm also experiencing this problem, and I have verified I have libogg, libvorbis, and their dev packages installed. I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not listed in the dependency lib list. Is this related? -Neil On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris m...@jerris.com wrote: looks like ogg devel packages are installed but ogg lib is not? On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: FreeSWITCH seems to be unable to read MP3 files, citing that it's an unknown format. Looking through the log, I found this during startup: 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_sync_wrote** There don't seem to be any compile-time errors, yet I can't seem to eliminate this issue. Any help would be appreciated. ___ 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 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
The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB djbin...@yahoo.com wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. -- *From:* Anthony Minessale anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Wed, December 16, 2009 3:42:48 PM *Subject:* Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 -- 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] detecting rtp packet for zombie channels
Are you doing proxy or bypass meda? 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 12:53 AM, Juan Backson wrote: Hi I have rtp-timeout-sec set to 300 s but I am still getting calls with duration of 1 day long. Is there any other ways to check for zombie channels? jb On Wed, Dec 16, 2009 at 10:52 PM, Brian West br...@freeswitch.org wrote: Why not just set rtp-timeout-sec on the sofia profile and it'll do that for you. Unless something else is going on. /b On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: Hi, I am having problem with around 1 % of the channels always get zombilized. What I want to do is to have a background thread that regularly check all the channels that have been in existance for like 1 hr, and then check to see if there is any RTP coming in and going out. If there is no RTP, then I just hangup that channel. Does anyone know if there is anyway to do that in a freeswitch module? Which API can I use to accomplish this purpose? Alternatively, is there anyway to configure freeswitch so that it will hangup the calls where there is no media in and out for so many seconds? Thanks, jb ___ 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
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] How to debug TLS handshake errors?
You could try ssldump: http://www.rtfm.com/ssldump/ On Thu, Dec 17, 2009 at 12:16 AM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Hello, I am trying to debug a TLS handshake error between FreeSwitch and some ATA. When setting the loglevel to 9 I get only a message that TLS handshake failed. Is there some other debug command to show what happens during the TLS handshake process? Thanks! __Yehavi: ___ 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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] Polycom 501 conferencing with FreeSwitch
I'll rephrase my question: Has anyone done that, or should I dig into it? After all, Polycom is quite common... Thanks, __Yehavi: 2009/12/17 Michael Jerris m...@jerris.com Its software, anything is possible with enough time and effort. Mike On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: After some discussions with Polycom support it seems that their conferencing support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the latest and is not compatible with the latest one). Any idea whether it is possible to program Freeswitch to support this draft? Thanks, __Yehavi: ___ 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] detecting rtp packet for zombie channels
sip session timers is the standardized way to handle this. On Thu, Dec 17, 2009 at 10:00 AM, Mathieu Rene mrene_li...@avgs.ca wrote: Are you doing proxy or bypass meda? 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 12:53 AM, Juan Backson wrote: Hi I have rtp-timeout-sec set to 300 s but I am still getting calls with duration of 1 day long. Is there any other ways to check for zombie channels? jb On Wed, Dec 16, 2009 at 10:52 PM, Brian West br...@freeswitch.org wrote: Why not just set rtp-timeout-sec on the sofia profile and it'll do that for you. Unless something else is going on. /b On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: Hi, I am having problem with around 1 % of the channels always get zombilized. What I want to do is to have a background thread that regularly check all the channels that have been in existance for like 1 hr, and then check to see if there is any RTP coming in and going out. If there is no RTP, then I just hangup that channel. Does anyone know if there is anyway to do that in a freeswitch module? Which API can I use to accomplish this purpose? Alternatively, is there anyway to configure freeswitch so that it will hangup the calls where there is no media in and out for so many seconds? Thanks, jb ___ 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 -- 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] mod_voicemail question
Hello Micheal
On Dec 15, 2009, at 12:09 PM, Michael Collins wrote:
Hi all,
What is the difference between the mod_voicemail vm_message_ext parameter
and the file-extension parameter?
vm_message_ext is a channel variable:
http://wiki.freeswitch.org/wiki/Mod_voicemail#vm_message_ext
file-extension is a parameter of the voicemail module:
http://wiki.freeswitch.org/wiki/Mod_voicemail#file-extension
The former sets for a specific user, the latter for mod_voicemail in general.
Ahh, thanks for clearing this up for me! Now I understand the difference.
I want all my voicemail in .WAV format except for a couple of extensions
which need to be in MP3.
I'm getting strange results playing with these settings, for example, after
logging into the voicemail, it will say You have 1 new message. First
message at date and time, and then instead of the voicemail message there
will be silence and a long pause. Then it will repeat the message count and
loop this behavior. During the silence, I seem to be able to press keys to
trigger voicemail events, like for example I am allowed to delete the message
(although it isn't playing the message to me, and I am instead hearing
silence).
Any ideas?
Is this perhaps a recording of silence, so that you might actually be
listening to a message?
-MC
Nope, turns out that according to the FS logs it was trying to play a
{uuid}.WAV file that it was expecting to still be there, but which was deleted
from a previous checking of voicemail. It's like the database of new messages
was out of sync with the message sound files in the mailbox on the server. I
deleted the 'ghost' voicemails from my mailbox and now things are back to
normal. That could have been a result of my experimentations, I doubt it was a
problem with FS.
Thanks for your help and keep up the outstanding work! I love FreeSWITCH. :-)
Steve___
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[Freeswitch-users] Voicemail-Email
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] Small delay in registration validity
The sql is sorted into transactions to boost performance so it waits for either 500 statements to execute or 500ms to elapse to accumulate as many sql stmts as possible into the transaction. set sql-in-transactions to false in your profile or make a patch to make the 500ms configurable On Thu, Dec 17, 2009 at 9:53 AM, mayamatakeshi mayamatake...@gmail.comwrote: It seems to me, in previous revisions of FS, we could successfully call a registered user as soon as his terminal gets 200 OK for REGISTER. But after testing recent revisions, it seems we must wait a little (I wait 1 second) otherwise a call to bridge would end with this: 2009-12-17 22:46:14.155163 [ERR] switch_ivr_originate.c:2249 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] Similar thing is happening when the terminal unregisters: after unregistration an immediate call to bridge sofia/profile/user%domain will succeed. Has anything changed recently in the way registration works that could explain this? br, takeshi ___ 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'd be suspicious of: (a) the CSeq going backwards from 4 to 3 between re-invites 2 and 3; (b) the branch on the Via tag changing (c) (not sure about this one) the SDP session ID and version changing for what's the same session. --Dave Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. __ From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 ___ 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
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] SIP Re-invite
The trace that I pasted on the pastebin was from our analyzer,Tektronix spectra2 that was sitting between FS and customer. I also had the FS sip trace on and compare with the trace from Spectra when I found out about the 3rd re-invite was missing from FS. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB djbin...@yahoo.com wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn: +19193869900 +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 -- 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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
It only happened to the calls from this customer that keeps sending re-invite every 30 minutes, since their switch is expecting a reply back from those re-invite and FS did not respond back to those re-invite. Thank you. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 7:36:44 AM Subject: Re: [Freeswitch-users] SIP Re-invite if you don't see it in sofia siptrace but do see it in tcpdump capture then something very ugly is going on. Either sofia has hung up completely and is not listening on that port anymore (can other calls go through?) or the packet you see in tcpdump is not really going to the right port. Can you confirm which one? Mike On Dec 16, 2009, at 6:29 PM, DJB wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. ___ 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
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] Small delay in registration validity
On Fri, Dec 18, 2009 at 1:12 AM, Anthony Minessale anthony.miness...@gmail.com wrote: The sql is sorted into transactions to boost performance so it waits for either 500 statements to execute or 500ms to elapse to accumulate as many sql stmts as possible into the transaction. set sql-in-transactions to false in your profile or make a patch to make the 500ms configurable Thanks. To change the param sql-in-transactions is enough for me (just during tests). I tested setting it to false and the behavior is as expected. I have updated the wiki: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#sql-in-transactions On Thu, Dec 17, 2009 at 9:53 AM, mayamatakeshi mayamatake...@gmail.comwrote: It seems to me, in previous revisions of FS, we could successfully call a registered user as soon as his terminal gets 200 OK for REGISTER. But after testing recent revisions, it seems we must wait a little (I wait 1 second) otherwise a call to bridge would end with this: 2009-12-17 22:46:14.155163 [ERR] switch_ivr_originate.c:2249 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] Similar thing is happening when the terminal unregisters: after unregistration an immediate call to bridge sofia/profile/user%domain will succeed. Has anything changed recently in the way registration works that could explain this? br, takeshi ___ 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 ___ 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] Mirroring wiki with wget for offline browsing?
Hello I'm no wget expert, and figured I should ask here first: I'd like to download the whole wiki using wget for off-line reading. Using the following didn't work: wget -m -np http://wiki.freeswitch.org/wiki/Main_Page If I move the wiki/ directory to the root directory of my web server, and try to open http://localhost/wiki/Main_Page, FireFox tries to download the page with this dialog box: You have chosen to open Main_Page which is a: application/octet-stream I assume wget can do this, but I don't know enough. Has someone succeeded in downloading the whole wiki with wget and could give the right switches to use? Thank you. -- View this message in context: http://old.nabble.com/Mirroring-wiki-with-wget-for-offline-browsing--tp26831043p26831043.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] Mirroring wiki with wget for offline browsing?
I would rather you not do that with wget you beat the hell out of the wiki resources... how often do you do this? I would try doing a printable version. /b On Dec 17, 2009, at 10:56 AM, Fred-145 wrote: Hello I'm no wget expert, and figured I should ask here first: I'd like to download the whole wiki using wget for off-line reading. Using the following didn't work: wget -m -np http://wiki.freeswitch.org/wiki/Main_Page If I move the wiki/ directory to the root directory of my web server, and try to open http://localhost/wiki/Main_Page, FireFox tries to download the page with this dialog box: You have chosen to open Main_Page which is a: application/octet-stream I assume wget can do this, but I don't know enough. Has someone succeeded in downloading the whole wiki with wget and could give the right switches to use? 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] Mirroring wiki with wget for offline browsing?
I only tried once. Maybe someone used to wget could generate a PDF in case people need an offline copy? -- View this message in context: http://old.nabble.com/Mirroring-wiki-with-wget-for-offline-browsing--tp26831043p26831566.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] Building on Windows
On Thu, Dec 17, 2009 at 05:02:10PM -, Dave Stevenson wrote: Hi, I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but.., I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as Unsupported, although the Wiki says that you only need VC++2005. What does unsupported mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? Install VS 2008 if at all possible (express edition is free). 2005 support isn't maintained much if at all, so a lot of newer modules stand a good chance of not having support. 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like make current and make sounds may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? Tortoise SVN is fine and is probably the de-facto client for windows. 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 4. There was one fatal error in the build of mod_opal (missing file) (Some examples of the warnings and the error are shown below :-) Try with VS 2008 and see if they go away. 5. How do I specify which options (e.g., mod_flite, to be included iin the build. You can enable the different sub projects somehow in the UI, I always forget exactly how but just click around in VS and you'll find it. 6. How do I build the sounds etc. ? The sounds are a subproject too IIRC. 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] SIP Re-invite
Anthony/Michael, I finally have a complete traces of a call at http://pastebin.freeswitch.org/11539 There are 2 traces in there from within the same box. 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 Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB djbin...@yahoo.com wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 -- 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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
[Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl
Hi,I am trying to define Gateways (for inbound and outbound calls via SIP
provider) within Directory (under internal sample profile) using XML CURLBut
I am getting this warning:2009-12-17 14:42:03.294519 [WARNING] sofia_reg.c:2115
Gateway 'MyGW' not found.
And
sofia status gateway MyGWAPI CALL [sofia(status gateway MyGW)] output:Invalid
Gateway!
This is my configuration (overlook language details )
section name=\directory\+ domain name=\10.0.0.124\+ user
id=\test\+ gateways+ gateway name=\MyGW\+
param name=\username\ value=\234wf423\/+param
name=\password\ value=\pwdpwd\/+ param name=\realm\
value=\testvoip.com\/+ param name=\proxy\
value=\my.provider.com\/+param name=\register\ value=\true\/+
/gateway+ /gateways+params+ param
name=\password\ value=\1234\/+param name=\dial-string\
value=\{presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}\/+
/params+variables+ variable
name=\register-gateway\ value=\MyGW\/+variable
name=\accountcode\ value=\test\/+ variable name=\user_context\
value=\mycontext\/+variable name=\effective_caller_id_name\
value=\Test User\/+variable name=\effective_caller_id_number\
value=\1234\/+ /variables+ /user+ /domain+
/section+ /document;
User id test is able to register and call other internal users
In my sip_profiles/internal.xml I have:
!-- indicator to parse the directory for domains with parse=true to get
gateways--domain name=all parse=true/!-- indicator to parse the
directory for domains with parse=true to get gateways and alias every domain
to this profile --domain name=all alias=true parse=true/
Can you help me with this issue?
Thank youPaulo
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Re: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch
I have not seen anyone mention it. Mike On Dec 17, 2009, at 11:07 AM, Yehavi Bourvine wrote: I'll rephrase my question: Has anyone done that, or should I dig into it? After all, Polycom is quite common... Thanks, __Yehavi: 2009/12/17 Michael Jerris m...@jerris.com Its software, anything is possible with enough time and effort. Mike On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: After some discussions with Polycom support it seems that their conferencing support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the latest and is not compatible with the latest one). Any idea whether it is possible to program Freeswitch to support this draft? ___ 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] [freeswitch-users] Gateway in Directory Loaded via xml curl
I'm going to guess you removed these lines from your profile:
domains
domain name=all alias=false parse=true/
/domains
parse=true causes the profile to parse the domain looking for gateways and
register them..
/b
On Dec 17, 2009, at 11:18 AM, Paulo Vicentini wrote:
Hi,
I am trying to define Gateways (for inbound and outbound calls via SIP
provider) within Directory (under internal sample profile) using XML CURL
But I am getting this warning:
2009-12-17 14:42:03.294519 [WARNING] sofia_reg.c:2115 Gateway 'MyGW' not
found.
And
sofia status gateway MyGW
API CALL [sofia(status gateway MyGW)] output:
Invalid Gateway!
This is my configuration (overlook language details )
section name=\directory\+
domain name=\10.0.0.124\+
user id=\test\+
gateways+
gateway name=\MyGW\+
param name=\username\ value=\234wf423\/+
param name=\password\ value=\pwdpwd\/+
param name=\realm\ value=\testvoip.com\/+
param name=\proxy\ value=\my.provider.com\/+
param name=\register\ value=\true\/+
/gateway+
/gateways+
params+
param name=\password\ value=\1234\/+
param name=\dial-string\
value=\{presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}\/+
/params+
variables+
variable name=\register-gateway\ value=\MyGW\/+
variable name=\accountcode\ value=\test\/+
variable name=\user_context\ value=\mycontext\/+
variable name=\effective_caller_id_name\ value=\Test User\/+
variable name=\effective_caller_id_number\ value=\1234\/+
/variables+
/user+
/domain+
/section+
/document;
User id test is able to register and call other internal users
In my sip_profiles/internal.xml I have:
!-- indicator to parse the directory for domains with parse=true to
get g
ateways--
domain name=all parse=true/
!-- indicator to parse the directory for domains with parse=true to
get g
ateways and alias every domain to this profile --
domain name=all alias=true parse=true/
Can you help me with this issue?
Thank you
Paulo
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Re: [Freeswitch-users] SIP Re-invite
I am sorry; here is the complete one: http://pastebin.freeswitch.org/11540 Thank you. From: DJB djbin...@yahoo.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 9:35:27 AM Subject: Re: [Freeswitch-users] SIP Re-invite Anthony/Michael, I finally have a complete traces of a call at http://pastebin.freeswitch.org/11539 There are 2 traces in there from within the same box. 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 Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB djbin...@yahoo.com wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 -- 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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] How can I join two freeswitch on two servers?
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--tp24045824p26831042.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
Can you post the full packets with Ethernet, IP, UDP headers as well, or upload a pcap file? I'll add the change in 'Max-Forwards' from 70 to 69 between the two packets to my things to be suspicious of list. --Dave The trace that I pasted on the pastebin was from our analyzer,Tektronix spectra2 that was sitting between FS and customer. I also had the FS sip trace on and compare with the trace from Spectra when I found out about the 3rd re-invite was missing from FS. Thank you. __ From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB djbin...@yahoo.com wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. __ From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn: +19193869900 +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 -- Anthony Minessale II FreeSWITCH
Re: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9
I found the issue with this. I did an svn checkout from the trunk, and then
I did a local svn export to another local folder. For some reason, the svn
export did not include the libs/openzap folder (which was not the case when
I got 1.0.5pre8). Must I do a separate svn export from the libs/openzap
folder?
Best Regards,
Jerry
-Original Message-
From: Brian West [mailto:br...@freeswitch.org]
Sent: Wednesday, December 16, 2009 2:28 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9
Need siptrace with this type sofia profile siptrace on replace
with your profile.
/b
On Dec 16, 2009, at 4:23 PM, Jerry Richards wrote:
I upgraded to the latest 1.0.5pre9 and now if I try to call from an
internal phone to an external number on my Sangoma PRI, I get a 502 Bad
Gateway
reply. Below is the console loglevel 7 output. It says the
destination is out-of-order. I'm not sure what this means. Any help is
appreciated.
2009-12-16 14:10:46.410656 [DEBUG] sofia.c:5285 0 acls to check for
proxy
2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5303 network ip is a proxy
[0]
2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5331 IP 192.168.72.32
Rejected by acl domains. Falling back to Digest auth.
2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5285 0 acls to check for
proxy
2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5303 network ip is a proxy
[0]
2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5331 IP 192.168.72.32
Rejected by acl domains. Falling back to Digest auth.
2009-12-16 14:10:46.457607 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/5...@192.168.72.141:5060
[e58e763f-7688-4600-aa70-481bbc359f58]
2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3787 Channel
sofia/internal/5...@192.168.72.141:5060 entering state [received][100]
2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3798 Remote SDP:
v=0
o=TC 1100638826 1100638826 IN IP4 192.168.72.32 s=session c=IN IP4
192.168.72.32 t=0 0 m=audio 1760 RTP/AVP 0 18 4 101 a=rtpmap:0
PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000/1
a=ptime:20
a=ptime:20
2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3923
(sofia/internal/5...@192.168.72.141:5060) State Change CS_NEW -
CS_INIT
2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/5...@192.168.72.141:5060 [BREAK]
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/5...@192.168.72.141:5060) Running State Change CS_INIT
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/5...@192.168.72.141:5060) State INIT
2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:83
sofia/internal/5...@192.168.72.141:5060 SOFIA INIT
2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:111
(sofia/internal/5...@192.168.72.141:5060) State Change CS_INIT -
CS_ROUTING
2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/5...@192.168.72.141:5060 [BREAK]
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/5...@192.168.72.141:5060) State INIT going to sleep
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/5...@192.168.72.141:5060) Running State Change
CS_ROUTING
2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/5...@192.168.72.141:5060) State ROUTING
2009-12-16 14:10:46.458582 [DEBUG] mod_sofia.c:132
sofia/internal/5...@192.168.72.141:5060 SOFIA ROUTING
2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:78
sofia/internal/5...@192.168.72.141:5060 Standard ROUTING
2009-12-16 14:10:46.458582 [INFO] mod_dialplan_xml.c:408 Processing
Anonymous-93491028 in context default
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-unloop] continue=false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (PASS)
[unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (FAIL)
[unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-tod_example] continue=true
Dialplan: day of week[4] =~ 2-6 (PASS)
Dialplan: hour[14] =~ 9-18 (PASS)
Dialplan: sofia/internal/5...@192.168.72.141:5060 Date/Time Match
(PASS) [tod_example] break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Action
set(open=true)
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-holiday_example] continue=true
Dialplan: month[12] =~ 1 (FAIL)
Dialplan: sofia/internal/5...@192.168.72.141:5060 Date/Time Match
(FAIL) [holiday_example] break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-Mediant1000] continue=false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (FAIL)
[Mediant1000]
destination_number(93491028) =~ /^8(\d+)$/ break=on-false
Dialplan:
Re: [Freeswitch-users] Building on Windows
On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: On Thu, Dec 17, 2009 at 05:02:10PM -, Dave Stevenson wrote: Hi, I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but.., I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as Unsupported, although the Wiki says that you only need VC++2005. What does unsupported mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? Install VS 2008 if at all possible (express edition is free). 2005 support isn't maintained much if at all, so a lot of newer modules stand a good chance of not having support. We maintain it as far as things that work now shouldn't break, but we rarely test it and only fix things when people supply patches or let me know there is a problem so I can address it. 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like make current and make sounds may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? Tortoise SVN is fine and is probably the de-facto client for windows. make current and such are all for the unix build only, on the msvc (at least 2008) build they are all built right into the solution ] 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 2005 has slightly different warning settings than are even available in 2008 so I get these from time to time. If you open up a bug on jira.freeswitch.org for me with details I can try to get them corrected. 4. There was one fatal error in the build of mod_opal (missing file) (Some examples of the warnings and the error are shown below :-) Try with VS 2008 and see if they go away. I think this is due to missing dependencies. I don't think I had automation to download the right svn versions of opal. 5. How do I specify which options (e.g., mod_flite, to be included iin the build. You can enable the different sub projects somehow in the UI, I always forget exactly how but just click around in VS and you'll find it. You can adjust this in the configuration managaer 6. How do I build the sounds etc. ? The sounds are a subproject too IIRC. I think think might only be in the 2008 versions, I can't recall to be sure, but there are targets you can build that will install them. Mike ___ 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] Getting 502 Bad Gateway with 1.0.5.pre9
This would have nothing to do with receiving a 502 on sip. /b On Dec 17, 2009, at 12:08 PM, Jerry Richards wrote: I found the issue with this. I did an svn checkout from the trunk, and then I did a local svn export to another local folder. For some reason, the svn export did not include the libs/openzap folder (which was not the case when I got 1.0.5pre8). Must I do a separate svn export from the libs/openzap folder? 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
[Freeswitch-users] Handling REFER...
Hello everyone, I've got two profiles running: s2s and trunk. The context for s2s is defined as s2s-in. The context for trunk is defined as trunk-in. trunk is bound to 192.168.168.3. recv 481 bytes from udp/[192.168.168.76]:5065 at 18:43:37.309706: REFER sip:mod_so...@192.168.168.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 To: NONAME sip:19415551...@192.168.168.3;tag=BagvZeKSrj7yH From: sip:9412848...@192.168.168.76:5065;transport=udp;tag=203332153_1430350929_10 Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Max-Forwards: 70 Refer-To: sip:6463959...@192.168.168.3 Contact: sip:s...@192.168.168.76:5065;transport=udp Content-Length: 0 send 592 bytes to udp/[192.168.168.76]:5065 at 18:43:37.316093: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 From: sip:9412848...@192.168.168.76:5065;transport=udp;tag=203332153_1430350929_10 To: NONAME sip:9415551...@192.168.168.3;tag=BagvZeKSrj7yH Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Contact: sip:mod_so...@192.168.168.3:5060 User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 FS routed this to the s2s-in context, even though it was sent to the trunk profile. Shouldn't it have ended up in trunk-in? For the time being I wrote some crazy dialplan for s2s-in to transfer the call to trunk-in but I'm wondering what could be going on here. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] detecting rtp packet for zombie channels
Please try on SVN trunk. I might toss a PRE10 sooner. /b On Dec 17, 2009, at 1:05 PM, Juan Backson wrote: Hi, I am using 1.0.4 (exported) with proxy media, and I have enable-timer = true and minimum-session-expires=120. Is this the correct way of setting the sip session timers? thanks, jb ___ 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 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 setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to notice level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from top, it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. ___ 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
If you're going to have that many listeners then it would be best to use something like shoutcast to broadcast the stream out to a local stream on various different boxes... then tie the callers into a stream... when they have questions uuid_transfer them into the conf.. then back to the stream when done. This would scale to very large numbers because you could split it out into 100's of boxes if needed. /b On Dec 17, 2009, at 1:29 PM, Brian 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. ___ 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 to debug TLS handshake errors?
I am trying Audiocodes and Vegastream ATAs, and work with either the manufacturer or the local representative here. On SNOM I managed to make it work, and will try Polycom soon (once I manage to grab one unit from our users...). Thanks, __yehavi: 2009/12/17 Brian West br...@freeswitch.org Also what device are you using? I haven't tested with many so far... Polycom, Snom and a few others do TLS (see interop page on wiki) others do it wrong. /b On Dec 17, 2009, at 10:04 AM, Kristian Kielhofner wrote: You could try ssldump: http://www.rtfm.com/ssldump/ ___ 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
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 setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I’ve set file logging to “notice” level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from “top”, it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. ___ 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 -- 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] mod_conference scalability
We are always doing enhancements and yes there are some real scalability enhancements in trunk compared to 1.0.4, I am just not sure if they effect conference significantly or not. I would guess that trunk is actually more stable than 1.0.4 at the moment. Give it a try and find out. Mike On Dec 17, 2009, at 2:29 PM, Brian 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 setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I’ve set file logging to “notice” level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from “top”, it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. ___ 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
Re: [Freeswitch-users] Building on Windows
The sounds projects (which do the downloads and extraction) are not present for 2005. Also alot of the newer modules dont have build support either. I would suggest you use VS2008 Express Michael Jerris wrote: On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: On Thu, Dec 17, 2009 at 05:02:10PM -, Dave Stevenson wrote: Hi, I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but.., I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as Unsupported, although the Wiki says that you only need VC++2005. What does unsupported mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? Install VS 2008 if at all possible (express edition is free). 2005 support isn't maintained much if at all, so a lot of newer modules stand a good chance of not having support. We maintain it as far as things that work now shouldn't break, but we rarely test it and only fix things when people supply patches or let me know there is a problem so I can address it. 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like make current and make sounds may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? Tortoise SVN is fine and is probably the de-facto client for windows. make current and such are all for the unix build only, on the msvc (at least 2008) build they are all built right into the solution ] 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 2005 has slightly different warning settings than are even available in 2008 so I get these from time to time. If you open up a bug on jira.freeswitch.org for me with details I can try to get them corrected. 4. There was one fatal error in the build of mod_opal (missing file) (Some examples of the warnings and the error are shown below :-) Try with VS 2008 and see if they go away. I think this is due to missing dependencies. I don't think I had automation to download the right svn versions of opal. 5. How do I specify which options (e.g., mod_flite, to be included iin the build. You can enable the different sub projects somehow in the UI, I always forget exactly how but just click around in VS and you'll find it. You can adjust this in the configuration managaer 6. How do I build the sounds etc. ? The sounds are a subproject too IIRC. I think think might only be in the 2008 versions, I can't recall to be sure, but there are targets you can build that will install them. Mike ___ 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://n2.nabble.com/Building-on-Windows-tp4182382p4183177.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] [Windows] Stable enough for production use?
I run FreeSWITCH on a Windows Server 2008 R2 (x64) box with several analog lines and it works very well. mercutioviz wrote: And we shouldn't be using 1.0.4 anyway, should we? ;) -MC On Wed, Dec 16, 2009 at 3:26 PM, Moises Silva moises.si...@gmail.comwrote: I've been using FreeSWITCH on Windows lately and seems to work pretty well. Sangoma has been testing more and more lately the Windows drivers with FreeSWITCH, and I think you should be just fine.I have not tested 1.0.4 though, always using trunk, if you are going to be using PSTN lines (and therefore requiring openzap) I think it would be a good idea for you to use trunk and latest wanpipe drivers. -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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 -- View this message in context: http://n2.nabble.com/Windows-Stable-enough-for-production-use-tp4174199p4183200.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
Yes, while it is true that does make a profound difference but if he has many listeners and not very many talkers... just tapping into the conference and streaming that audio out would scale well. /b On Dec 17, 2009, at 1:50 PM, Steve Underwood wrote: I don't think you have mentioned which codecs are involved. This can have a profound effect. Steve ___ 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 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 setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to notice level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from top, it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. ___ 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] [freeswitch-users] Gateway in Directory Loaded via xml curl
Hi,FS was sending (while loading modules) such request: [purpose] = gateways
But I was not aware of that...so that I am replying FS with my Gateways now...
But now I am wondering...suppose I have 1000 domains and two different gateways
per domain (2K Gateways) Should I reply FS request with such huge XML on
startup?
Thanks for your backings
PauloFrom: br...@freeswitch.org
Date: Thu, 17 Dec 2009 11:44:15 -0600
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded
via xml curl
I'm going to guess you removed these lines from your profile:
domains
domain name=all alias=false parse=true/
/domains
parse=true causes the profile to parse the domain looking for gateways and
register them..
/b
On Dec 17, 2009, at 11:18 AM, Paulo Vicentini wrote:Hi,I am trying to define
Gateways (for inbound and outbound calls via SIP provider) within Directory
(under internal sample profile) using XML CURLBut I am getting this
warning:2009-12-17 14:42:03.294519 [WARNING] sofia_reg.c:2115 Gateway 'MyGW'
not found.
And
sofia status gateway MyGWAPI CALL [sofia(status gateway MyGW)] output:Invalid
Gateway!
This is my configuration (overlook language details )
section name=\directory\+ domain name=\10.0.0.124\+ user
id=\test\+ gateways+ gateway name=\MyGW\+
param name=\username\ value=\234wf423\/+param
name=\password\ value=\pwdpwd\/+ param name=\realm\
value=\testvoip.com\/+ param name=\proxy\
value=\my.provider.com\/+param name=\register\ value=\true\/+
/gateway+ /gateways+params+ param
name=\password\ value=\1234\/+param name=\dial-string\
value=\{presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}\/+
/params+variables+ variable
name=\register-gateway\ value=\MyGW\/+variable
name=\accountcode\ value=\test\/+ variable name=\user_context\
value=\mycontext\/+variable name=\effective_caller_id_name\
value=\Test User\/+variable name=\effective_caller_id_number\
value=\1234\/+ /variables+ /user+ /domain+
/section+ /document;
User id test is able to register and call other internal users
In my sip_profiles/internal.xml I have:
!-- indicator to parse the directory for domains with parse=true to get
gateways--domain name=all parse=true/!-- indicator to parse the
directory for domains with parse=true to get gateways and alias every domain
to this profile --domain name=all alias=true parse=true/
Can you help me with this issue?
Thank youPaulo
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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 setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I’ve set file logging to “notice” level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from “top”, it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps
Re: [Freeswitch-users] mod_conference scalability
Hi Brian, I imagine that one of the issues is that you're using a complex sledgehammer (mod_conference) to crack a simple nut - that of having multiple listeners listening to a single speaker. As far as I am aware, FreeSWITCH doesn't have anything built in which will allow this kind of simple audio path switching - maybe someone more knowledgeable than me will correct me if I'm wrong? I presented some stuff at ClueCon which would address this kind of simple application and ought to scale well beyond what you've seen with FS or Asterisk. It's still pretty basic [I'd do more with it if I wasn't so busy joshing with the other Brian on Facebook], and has never been deployed in anger but, if you're interested, drop me a note off-list. --Dave 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 setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I’ve set file logging to “notice” level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1
Re: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl
In your case don't store them in the domain put them in the gateways tags on the profile directly. /b On Dec 17, 2009, at 2:46 PM, Paulo Vicentini wrote: Hi, FS was sending (while loading modules) such request: [purpose] = gateways But I was not aware of that...so that I am replying FS with my Gateways now... But now I am wondering...suppose I have 1000 domains and two different gateways per domain (2K Gateways) Should I reply FS request with such huge XML on startup? Thanks for your backings Paulo ___ 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] How to overcome 415 Unsupported Media Type
I try to attach Bravis video conference clients to Freeswitch. Those video conference clients are really working good (Multilingual clients for testing ca be downloaded here: http://www.bravis.eu/). Some big companies here in Germany use them in large installations. They are based on SIP, but do not use any publicly known codecs. Normally they are maintained and routed via our OpenSIPS server, but I would like to integrate them into our Freeswitch system. That way I do not have to manage 2 SIP servers for phone calls and video conferencing calls. However the SIP message does not provide Content-Type: application/sdp. Instead it provides Content-Type: application/BRAVIS. The clients register successfully but they do not invite. Freeswitch answers SIP/2.0 415 Unsupported Media Type. I have added bypass_media=true into the dialplan and inbound-late-negotiation true in the internal profile but this didn't help. I think Freeswitch complains about the content-type. Is there any way how I may overcome this? Here is a sample Invite INVITE sip:835...@sip5.mydomain.com SIP/2.0. From: myname sip:835...@sip5.mydomain.com;tag=5c5c3ef6bbe9de119f1aa11f7ca41a5f. To: sip:835...@sip5.mydomain.com. Via: SIP/2.0/UDP 217.xxx.xxx.xx6:5530;iid=9931;branch=z9hG4bKc4583ef6bbe9de119f1aa11f7ca41a5f;uas-addr=217.24.11.190;rport. CSeq: 4711 INVITE. Call-ID: 2-ee3d3ef6-bbe9-de11-9fa1-a11f7ca41a5f. Contact: myname sip:835...@217.xxx.xxx.xx6:5530. User-Agent: BRAVIS/1.5.20.27.4585 (Linux 2.6.31-16-generic; generic; Ubuntu 9.10; i686; de; 8). Max-Forwards: 70. Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS. Supported: 100rel. Content-Type: application/BRAVIS. Content-Length: 174. ACAABAAAFDAAABAAMEFBHCGLACAACAAACPKNBHGOAPLDABAAFAAADBABAAPPELAFAACAAAHDHCGGGMHIPPUPOPBEKHHHAPLDOPBEKHHHAPLDABAADCABAAADFBMDHOAEAAGIGPHDHEAAPPJFKGAPLHHNKF. 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] Performance Tuning
Looking at Performance Tune my Freeswitch http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations Is refers to the following: Turn off every module you don't need Turn presence off in the profiles libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles mod_cdr_csv is slower than mod_xml_cdr How do I change each one any references on Wiki? ___ 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] Performance Tuning
1. http://wiki.freeswitch.org/wiki/Modules.conf.xml http://wiki.freeswitch.org/wiki/Modules.conf.xml2. http://wiki.freeswitch.org/wiki/Sofia.conf.xml#manage-presence http://wiki.freeswitch.org/wiki/Sofia.conf.xml#manage-presence3. http://wiki.freeswitch.org/wiki/Getting_Started_Guide#SIP_Profiles http://wiki.freeswitch.org/wiki/Getting_Started_Guide#SIP_Profiles Might not be entirely helpful, but basically you can use either the external or internal profiles and change the ports, etc., as required. 4. You can disable mod_cdr_csv and enable mod_xml_cdr based on #1. Thanks, Vinuth. On Fri, Dec 18, 2009 at 3:08 AM, Ujjval Karihaloo ujj...@simplesignal.comwrote: Looking at Performance Tune my Freeswitch http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations Is refers to the following”: Turn off every module you don't need Turn presence off in the profiles libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles mod_cdr_csv is slower than mod_xml_cdr How do I change each one ….any references on Wiki? ___ 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] Performance Tuning
libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles If you only have one provider for your trunk is it possible to set up multiple profiles for enhanced performance? For example if I have multiple DDIs from the provider can I set up a different profile for each one? Or maybe based on some some sort of a pattern? ___ 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
Hello Oliver,
I have the same on Ubuntu wth newest trunk.
Best regards
Peter
Oliver Schönbeck schrieb:
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 $LOG
Has anybody seen similar effects before?
Any advice whats going wrong is heavily appreciated.
Thanks
Oliver
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Re: [Freeswitch-users] mod_conference scalability
What exactly are you doing I know it goes better than that.. are you using 64bit? / b On Dec 17, 2009, at 3:41 PM, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Voicemail-Email
yah it's exim segfaulting because you have to configure it to emulate
sendmail per the wiki page.
On Thu, Dec 17, 2009 at 4:17 PM, Peter P GMX prometheus...@gmx.net wrote:
Hello Oliver,
I have the same on Ubuntu wth newest trunk.
Best regards
Peter
Oliver Schönbeck schrieb:
Hello,
we are running freeswitch 1.0.trunk and are currently trying to get
the mod_voicemail to send the received messages to the user by using
exim4 on a debian machine.
So far we followed the instructions in the wiki article (
http://wiki.freeswitch.org/wiki/Mod_voicemail ).
I added some lines to the bash script to enable some kind of logging:
#! /bin/bash
typeset LOG=/tmp/${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
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 I’d hoped (based on what I’ve read on how FreeSWITCH is supposed to be generally very scalable). Here’s my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I’ve set file logging to “notice” level. My
[Freeswitch-users] sip message logging and analysis
I bit off topic but. Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier's first response is that we dropped the call. But this is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? ___ 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 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
Re: [Freeswitch-users] Handling REFER...
Thanks for the hint! force_transfer_context and force_transfer_dialplan. I've updated the wiki (I'll add an example once I test it). On Thu, Dec 17, 2009 at 5:06 PM, Anthony Minessale anthony.miness...@gmail.com wrote: The calls inherit the context from the parent, I think there is a var you can set on the chan to pick what context to use in a transfer like transfer_context or something grep the code for it On Dec 17, 2009 1:07 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Hello everyone, I've got two profiles running: s2s and trunk. The context for s2s is defined as s2s-in. The context for trunk is defined as trunk-in. trunk is bound to 192.168.168.3. recv 481 bytes from udp/[192.168.168.76]:5065 at 18:43:37.309706: REFER sip:mod_so...@192.168.168.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 To: NONAME sip:19415551...@192.168.168.3;tag=BagvZeKSrj7yH From: sip:9412848...@192.168.168.76:5065;transport=udp;tag=203332153_1430350929_10 Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Max-Forwards: 70 Refer-To: sip:6463959...@192.168.168.3 Contact: sip:s...@192.168.168.76:5065;transport=udp Content-Length: 0 send 592 bytes to udp/[192.168.168.76]:5065 at 18:43:37.316093: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 From: sip:9412848...@192.168.168.76:5065;transport=udp;tag=203332153_1430350929_10 To: NONAME sip:9415551...@192.168.168.3;tag=BagvZeKSrj7yH Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Contact: sip:mod_so...@192.168.168.3:5060 User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 FS routed this to the s2s-in context, even though it was sent to the trunk profile. Shouldn't it have ended up in trunk-in? For the time being I wrote some crazy dialplan for s2s-in to transfer the call to trunk-in but I'm wondering what could be going on here. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 message logging and analysis
Frank, Probably the cleanest (albeit non-FreeSWITCH) way to implement this would be to use OpenSIPS/SER/etc between you and the carrier with the siptrace module. But that's probably more work than you want. There's always tcpdump with a decent filter (udp port 5060 and host x.x.x.x) and then something like http://www.badpenguin.co.uk/files/pcap-util2 Both will allow you to search for BYEs and who is sending them. Also keep in mind that they (or you) may just be dropping the RTP without ever sending a BYE. Setting the various RTP timeouts in FreeSWITCH can help with that. You can then look for logs/events (are there any for RTP timeout?) to see who's dropping RTP. On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact fr...@impactfax.com wrote: I bit off topic but… Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier’s first response is that we dropped the call. But this is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? ___ 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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] Handling REFER...
On Thu, Dec 17, 2009 at 3:59 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Thanks for the hint! force_transfer_context and force_transfer_dialplan. I've updated the wiki (I'll add an example once I test it). I love it when users go all Chuck Norris and Rambo in answering their questions AND documenting the info! Thanks KK. -MC ___ 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 message logging and analysis
On Thu, Dec 17, 2009 at 4:01 PM, Frank @ Impact fr...@impactfax.com wrote: I bit off topic but… Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier’s first response is that we dropped the call. But this is aday later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? Jason Garland's ClueCon2009 videos about tcpdump and wireshark cover the thought of doing a rotating log file so that it captures a bunch of stuff but doesn't go over X number of megabytes... I don't recall exactly where in his videos that part appears, but here are the links to those vids. Hope it helps! -MC Look at this video first: http://www.viddler.com/explore/cluecon/videos/33/ Then check this one if you need more info: http://www.viddler.com/explore/cluecon/videos/8/ ___ 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 message logging and analysis
I'm using VQManager (there is a 30 day trial) and it's useful for seeing who does what / when per call; it's very easy to install... From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Frank @ Impact Sent: Thursday, December 17, 2009 4:02 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] sip message logging and analysis I bit off topic but... Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier's first response is that we dropped the call. But this is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? ___ 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] Handling REFER...
Also when can we expect little KK's running around? :P Congrats on the marriage /b On Dec 17, 2009, at 6:27 PM, Michael Collins wrote: I love it when users go all Chuck Norris and Rambo in answering their questions AND documenting the info! Thanks KK. -MC ___ 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 message logging and analysis
So is wireshark UI and its free! :P /b On Dec 17, 2009, at 6:33 PM, Chris Fowler wrote: I’m using VQManager (there is a 30 day trial) and it’s useful for seeing who does what / when per call; it’s very easy to install… ___ 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 message logging and analysis
i agree with christian, though i would use tshark. you can actually get the fields you want (method and callid) and store them in a dB. then you need to match them with a query. it is simple but Lots of work. look into -e and -E of tshark separate the fields by , have fun! David El 18/12/2009, a las 01:27, Kristian Kielhofner kristian.kielhof...@gmail.com escribió: Frank, Probably the cleanest (albeit non-FreeSWITCH) way to implement this would be to use OpenSIPS/SER/etc between you and the carrier with the siptrace module. But that's probably more work than you want. There's always tcpdump with a decent filter (udp port 5060 and host x.x.x.x) and then something like http://www.badpenguin.co.uk/files/pcap-util2 Both will allow you to search for BYEs and who is sending them. Also keep in mind that they (or you) may just be dropping the RTP without ever sending a BYE. Setting the various RTP timeouts in FreeSWITCH can help with that. You can then look for logs/events (are there any for RTP timeout?) to see who's dropping RTP. On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact fr...@impactfax.com wrote: I bit off topic but… Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier’s first response is that we dropped the call. But thi s is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? ___ 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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] sip message logging and analysis
I'm using contrib/seven/sip/sip2db.rb 2009/12/18 David Villasmil david.villasmil.w...@gmail.com: i agree with christian, though i would use tshark. you can actually get the fields you want (method and callid) and store them in a dB. then you need to match them with a query. it is simple but Lots of work. look into -e and -E of tshark separate the fields by , have fun! David El 18/12/2009, a las 01:27, Kristian Kielhofner kristian.kielhof...@gmail.com escribió: Frank, Probably the cleanest (albeit non-FreeSWITCH) way to implement this would be to use OpenSIPS/SER/etc between you and the carrier with the siptrace module. But that's probably more work than you want. There's always tcpdump with a decent filter (udp port 5060 and host x.x.x.x) and then something like http://www.badpenguin.co.uk/files/pcap-util2 Both will allow you to search for BYEs and who is sending them. Also keep in mind that they (or you) may just be dropping the RTP without ever sending a BYE. Setting the various RTP timeouts in FreeSWITCH can help with that. You can then look for logs/events (are there any for RTP timeout?) to see who's dropping RTP. On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact fr...@impactfax.com wrote: I bit off topic but… Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier’s first response is that we dropped the call. But thi s is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? ___ 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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 ___ 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?
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
Re: [Freeswitch-users] Creating Default Accounts on Directory
Please check your dialplan to match the new extension. You are looking for dialplan/default.xml extension Local_Extension. Check the cond destination_number, it should give you a good hint. Regards, JM On Fri, Dec 18, 2009 at 12:56 AM, Edmar Cruz darklio...@yahoo.com wrote: Hi Sir, I want to create a new xml file on the default directory of freeswitch where 1000.xml is located, sample i created 9387821.xml and copy the contents of the 1000.xml. The problem is when I used the account 9387821.xml and call 1000.xml it doesn't work the message in freeswitch it always CS_DESTROY... Please help me this with issue thanks... Edmar -- View this message in context: http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p26838457.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 -- João Mesquita FreeSWITCH™ Solutions t: +1 (646) 4959927 ___ 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 message logging and analysis
Some providers do retain call data for diagnostic purposes and to to aid in troubleshooting. Why not politely ask them if they could provide you with a sip trace themselves or forward along the evidence that supported their conclusion. They should be willing to help you solve a problem that may potentially be of benefit to their other customers that report similar issues. Otherwise, as others suggest, you could simply capture the signaling and media traffic from the FS box itself using tcpdump (e.g. tcpdump -i eth0 -s 0 -w debug.pcap host 127.0.0.1 ) or ngrep (-d eth0 -W byline -O /tmp/debug.pcap host 127.0.0.1) and analyze the resulting file in Wirehark (Statistics-Voip Calls or Telephony-Voip Calls in the current version). If your provider is using a session border controller or does not have a distributed architecture, then you can replace 127.0.0.1 with the appropriate address. If not, then simply don't use the host filter at all (it will result in a larger capture file). I would just keep in mind that if an upstream device (NAT router, firewall, etc.) is wreaking havoc with session refreshes by dropping re-INVITEs or UPDATEs (associated with session refreshing), you may not see them because of your vantage point. The reason I typically recommend using the -i (tcpdump) and -d (ngrep) switch is to avoid linux 'cooked' captures (more of a personal preference since I occasionally do have to convert or merge captures). If you only have SSH access to your FS box, you may want to use tcpdump or ngrep along with screen. tshark (tty/cli vesion of Wireshark) and sipgrep are also extremely useful. The later requires ngrep and a couple perl modules but I believe it is included with FS in the contrib or scripts directory--I forget which). -metik Frank @ Impact wrote: I bit off topic but… Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier’s first response is that we dropped the call. But this is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? ___ 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] Creating Default Accounts on Directory
Hi Sir, Not working condition field=destination_number expression=^(10[01][0-9])$ i set this to condition field=destination_number expression=^(80[1][0-9])$ to call 801.xml up to 809.xml on the dialplan/default.xml same thing... Thanks, Edmar João Mesquita-4 wrote: Please check your dialplan to match the new extension. You are looking for dialplan/default.xml extension Local_Extension. Check the cond destination_number, it should give you a good hint. Regards, JM On Fri, Dec 18, 2009 at 12:56 AM, Edmar Cruz darklio...@yahoo.com wrote: Hi Sir, I want to create a new xml file on the default directory of freeswitch where 1000.xml is located, sample i created 9387821.xml and copy the contents of the 1000.xml. The problem is when I used the account 9387821.xml and call 1000.xml it doesn't work the message in freeswitch it always CS_DESTROY... Please help me this with issue thanks... Edmar -- View this message in context: http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p26838457.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 -- João Mesquita FreeSWITCH™ Solutions t: +1 (646) 4959927 ___ 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/Creating-Default-Accounts-on-Directory-tp26838457p26838750.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] ACLs through proxy
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
Re: [Freeswitch-users] ACLs through proxy
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 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 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
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 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, 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 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
[Freeswitch-users] Destination Formats Expression
Hi Everyone, Is there a link or tutorial for the expressions format. Example: condition field=destination_number expression=^(10[01][0-9])$ 10 - default number [01[- second number that start only on 0 or 1; [0-9] - 0 to 9 can be use Is there any? Thanks, Edmar -- View this message in context: http://old.nabble.com/Destination-Formats-Expression-tp26840010p26840010.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] how does FS failover or load balance outbound calls between tow proxy
Hi All I have a FS cluster behind two OpenSIPS proxy, the incoming calls is load balance and failover to FS cluster by OpenSips, It works well. The problem is, the outbound calls from FS must also route throw then OpenSIPS servers. So, does FS servers can loadbalance the outbound calls between the two OpenSIPS servers and failover if one of the Opensips server is down? -- Lei.Tang lei.tl...@gmail.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] Destination Formats Expression
On Dec 17, 2009, at 11:34 PM, Jason White ja...@jasonjgw.net 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. http://wiki.freeswitch.org/wiki/Regular_Expression -MC ___ 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
