Re: [Freeswitch-users] Strange Performance when using as SBC
If you don't have to transcode, using proxy media mode will still save you some CPU time. This is 1/2 way between bypass media and the default media interactive mode. The other draw back to this mode is if you are using FS to clean up RTP and DTMF you loose those functions but they are not needed in most use cases. As far as the log level goes, I found that once I had things stable setting the loglevel to helped a good deal... Info is probably a bit too high of a loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you insist on leaving logging turned on... On a busy system these can and will generate a good deal of activity (and disk IO if using mod_logfile) Ken From: rod kawa...@laposte.net Reply-To: freeswitch-users@lists.freeswitch.org Date: Mon, 02 Feb 2009 11:36:35 +0400 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Ken, 1) I'd like to use FS to hide topology, so bypass media is not possible 2) done 3) done 4) not used 5) i'm using this ins switch.xml - param name=loglevel value=info/, if you think an other log level is more suitable. Regarding logging, I can see in console and in the freeswitch.log that there is still a lot of NOTICE logging, see below: 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8721 (sofia/internal/s...@10.10.10.1:5060) Ended 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/s...@10.10.10.1:5060 [CS_HANGUP] 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8722 (sofia/external/9...@10.10.20.100) Ended 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/external/9...@10.10.20.100 [CS_HANGUP] 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() Channel [sofia/external/9...@10.10.20.100] has been answered 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() Changing codec ptime to 30. I bet you have a linksys/sipura =D Do you have any idea where I can switch off this kind of logging. I thought it should be in /dialplan/internal.xml, but I see that in internal.xml - param name=debug value=0/ thanks a lot for your suggestion. regards, rod Ken Rice wrote: Dont forget there are several things you can do to increase performance... 1) where possible use bypass media or media proxy modes 2) mount freeswitch/db as a ram drive (if you are using voicemail with the internal FS DBs you'll need a way to make this persistant across reboots) 3) see the wiki for setting reasonable ulimits 4) (this is my oppinion others may vary) dont use mod_cdr_csv 5) turn off (or reduce logging) in switch.conf.xml all of these thing can greatly improve performance. On Mon, Feb 2, 2009 at 1:04 AM, rod kawa...@laposte.net mailto:kawa...@laposte.net wrote: Thanks Anthony, the setup is like this: sipp server FS 1 FS2 FS1 is the AMD CPU that has only one extension in dialplan that bridges to FS2. is the first extension in FS2 dialplan that plays moh, FS2 has no CPU pbm. FS1 is maxing out at 60 bridged calls without your option -hp. Using -hp, I'm now able to bridge 200 concurrent calls (a great improvement) and the system is still reactive. CPU load is high but not 100% and as the system responds well, I think that doesn't matter. The 2GB of memory are completely consumed (top command shows 700MB for FS process). I understand that FS1 server is not the best hardware platform, and I'm waiting for new 4 cores server for testing. I will update those numbers when testing with the new hardware. regards, rod. Anthony Minessale wrote: Which of the 2 machines has the load issue? You said it was one box calling the other. You have 2 major things against you, single CPU and AMD, but you should at least be able to get in the vicinity of 800-1000 calls on a box like that. Are you calling the default ? It's not really an appropriate extension for load testing. On the terminating box you should set up a manual extension that is the first one in the dial plan to play a wav file from preferably a ram disk or /tmp If you do plan on using this in production accept nothing less than a multi-core intel machine with at least 4 cores, the more cores the better because that parallel processing is where FS gets it's atvantage. On Fri, Jan 30, 2009 at 5:56 AM, rod kawa...@laposte.net mailto:kawa...@laposte.net mailto:kawa...@laposte.net mailto:kawa...@laposte.net wrote: Dear list, I've been playing with freeswitch for some time (2 months) and the fact is that I'm very pleased with the functionnalities of this software. I'd
Re: [Freeswitch-users] How do I set my FS internal ip address to a static value.
you need to add this setting to sofia.conf.xml param name=auto-restart value=false/ You'll also need to edit the sofia profiles and input the exact IP you wish it to bind to. The params are sip-ip and rtp-ip. /b On Feb 1, 2009, at 3:24 PM, c...@eugeneweb.com wrote: Hi Gang, I've been struggleing with this also. Actually I can get it to bind to my address, the problem is it randomly drops my calls. :-( I have a FS running on a box with a static IP and I can start a call between two extensions and it will go for hours. Then I add anther interface say eth0:0 with a new static IP and reconfigure my phones and FS to use that, and the calls drop after about 15-20 mins. Though it's pretty random. Here is my setup. I have Debian Linux 2.6.23.1 kernel, and freeswitch-1.0.1. Here is my /etc/network/interfaces: # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) # The loopback interface auto lo iface lo inet loopback # The first network card - this entry was created during the Debian installation auto eth0 eth0:0 iface eth0 inet dhcp iface eth0:0 inet static address 192.168.0.249 netmask 255.255.255.0 gateway 192.168.0.254 The only change I made to the FS config is in Vars.xml. I added this line close to the top: X-PRE-PROCESS cmd=set data=local_ip_v4=192.168.0.249/ Here is the console log of the call being dropped: freeswi...@archive sofia status API CALL [sofia(status)] output: NameType Data State = = = = = = = = = = = = = = = = = = = = = = = = = = = == external profile sip:mod_so...@67.171.158.226:5080 RUNNING (0) internal profile sip:mod_so...@192.168.0.249:5060 RUNNING (2) nat profile sip:mod_so...@67.171.158.226:5070 RUNNING (0) default alias internal ALIASED outbound alias external ALIASED 192.168.0.249 alias internal ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = = == 3 profiles 3 aliases freeswi...@archive 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 sofia_glue_restart_all_profiles() Reload XML [Success] 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM Reloaded 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 sofia_read_frame() Hangup sofia/internal/ 1...@192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 switch_ivr_multi_threaded_bridge() Hangup sofia/internal/1...@192.168.0.249 [CS_EXECUTE] [NORMAL_CLEARING] 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 6 (sofia/internal/ 1...@192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) Ended 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/ 1...@192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes [CS_HANGUP] 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 5 (sofia/internal/1...@192.168.0.249 ) Ended 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1...@192.168.0.249 [CS_HANGUP] 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [192.168.0.249] for profile [internal] 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [default] for profile [internal] 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started Profile internal [sofia_reg_internal] 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [outbound] for profile [external] 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started Profile external [sofia_reg_external] 2009-02-01 13:23:20 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started Profile nat [sofia_reg_nat] sofia status API CALL [sofia(status)] output: NameType Data State = = = = = = = = = = = = = = = = = = = = = = = = = = = == external profile sip:mod_so...@67.171.158.226:5080
[Freeswitch-users] mod_voicemail: Limiting the number how often a menu is repeated
Hello, today I searched for a way to limit the number of menu repeatings in mod_voicemail to let's say 3 times and when it reached the limit voicemail should abort. But I couldn't find a hint. Any ideas? regards helmut ___ 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] Q931 decoding Update
Hello, today I uploaded a little patch for openzap concerning missed linking of the pcap library. So loading ozmod_isdn failed with some kind of unknown symbol pcap_flush_dump error message. This keeps mod_openzap from loading at FS startup. regards helmut ___ 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] Call Variable not available when call hangup
the leg you are running the script on is not hungup, the other leg of the call is. If it was hungup you would not be executing the script. Asterisk and the h ext and the whole dead-agi thing are all poor design showing it's teeth. We do not support anything like it. You can however try this: (see the link below) http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks-giving-me-headaches-p21614840.html On Mon, Feb 2, 2009 at 6:53 AM, shehzad p pmh...@gmail.com wrote: Is there any settings that when call hangup control can be transferred to another context and these CDR values can be accessible there? (just like in Asterisk, h extension) shehzad p wrote: Hi all, I need to process some CDR variables in Dialplan, like call duration, Answered time etc. but when I place info application after bridge, it is not listing them properly as below: === Caller-Channel-Created-Time: [1233573341672157] Caller-Channel-Answered-Time: [1233573342712939] Caller-Channel-Hangup-Time: [0] == Here Hangup time is 0, So how can I find actual values? --I know that we can use xml_cdr or cdr_csv, but my current need is to get those values from dialplan itself so that can be passed to some script... thanks, msp -- View this message in context: http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21789152.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Strange Performance when using as SBC
Yes, that's exactly the issue: libtool. Provided that you use libtool 1.5.22-4 or some other 1.5.x version, FS seems to work fine with Intrepid. -- Raul On Mon, 2009-02-02 at 14:10 -0600, Brian West wrote: Its too bleeding edge and you had better know what you're doing if you use it. It comes with libtool 2.2 which you can't use to build FreeSWITCH. /b On Feb 2, 2009, at 2:01 PM, pe...@networkoblivion.com wrote: What is wrong with Intrepid? ___ 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] Strange Performance when using as SBC
On Tue, 03 Feb 2009 09:48:14 Raul Fragoso wrote: In addition to libtool, you may have issues with the latest packages of gcc and some other tools that FS will need. In any case, it's better to not use Intrepid at all ;-) Use Hardy as suggested and you will be happy. You shouldn't have any issues. I've used Intrepid on a VM to compile and test FreeSWITCH quite a bit and haven't run across any issues at all after downgrading libtool. That said I would also recommend Hardy LTS for production servers. hads -- http://nicegear.co.nz VoIP, DVB and other Linux compatible hardware. ___ 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] Strange Performance when using as SBC
Thanks rod for a quick answer, FS is installed on Ubuntu Server. I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to communicate with TDM but this all depends how much calls it can take, or maybe we can also do something in clustering environment ( I am not sure about it). But thanks again and any further help will be highly appreciated. Kind Regards Saeed Ahmed Tariq -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 1:53 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Saeed, I just created an account to share my setup on the wiki. I will detail all the steps for a clean install of a debian64 lenny with FS used as a SBC (next step is to try the new LCR module :) )and what I'm doing do stress the server. I wrote nothing at this time so please be patient, I'm waiting for my new hardware so that I will detail as much as possible what I'll do. For beginning I suggest you reading the start page on the wiki, especially these pages: -http://wiki.freeswitch.org/wiki/Getting_Started_Guide -http://wiki.freeswitch.org/wiki/Dialplan_XML maybe you could tell more about the linux distribution you're using so that I can give you some pointers for sipp... regards. rod. Saeed Ahmed wrote: Hi Rod, Could you please share how you configured Sipp FS to create a test environment? Especially the dial plan, sofia settings etc..., actually I am a newbie. I want to test it on a single FS machine. Kind Regards Saeed -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 11:00 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Ken, Jay, thanks for pointing to proxy media, I will test. Ken, you are right, I was brain damaged (a stupid mistake) when setting INFO cause this kind of level could be very verbose. I'm switching to CRIT or ERR. Thanks guys, rod. thanks for Ken Rice wrote: If you don't have to transcode, using proxy media mode will still save you some CPU time. This is 1/2 way between bypass media and the default media interactive mode. The other draw back to this mode is if you are using FS to clean up RTP and DTMF you loose those functions but they are not needed in most use cases. As far as the log level goes, I found that once I had things stable setting the loglevel to helped a good deal... Info is probably a bit too high of a loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you insist on leaving logging turned on... On a busy system these can and will generate a good deal of activity (and disk IO if using mod_logfile) Ken From: rod kawa...@laposte.net Reply-To: freeswitch-users@lists.freeswitch.org Date: Mon, 02 Feb 2009 11:36:35 +0400 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Ken, 1) I'd like to use FS to hide topology, so bypass media is not possible 2) done 3) done 4) not used 5) i'm using this ins switch.xml - param name=loglevel value=info/, if you think an other log level is more suitable. Regarding logging, I can see in console and in the freeswitch.log that there is still a lot of NOTICE logging, see below: 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8721 (sofia/internal/s...@10.10.10.1:5060) Ended 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/s...@10.10.10.1:5060 [CS_HANGUP] 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8722 (sofia/external/9...@10.10.20.100) Ended 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/external/9...@10.10.20.100 [CS_HANGUP] 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() Channel [sofia/external/9...@10.10.20.100] has been answered 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() Changing codec ptime to 30. I bet you have a linksys/sipura =D Do you have any idea where I can switch off this kind of logging. I thought it should be in /dialplan/internal.xml, but I see that in internal.xml - param name=debug value=0/ thanks a lot for your suggestion. regards, rod Ken Rice wrote: Dont forget there are several things you can do to increase performance... 1) where possible use bypass media or media proxy modes 2) mount freeswitch/db as a ram drive (if you are using voicemail with the internal FS DBs you'll need a way to make this persistant across
Re: [Freeswitch-users] Q931 decoding Update
We need to add more than this including detection in openzap configure.in if libpcap is available (headers and lib) and if not, disabling the functionality. MIke On Feb 2, 2009, at 1:04 PM, Helmut Kuper wrote: Hello, today I uploaded a little patch for openzap concerning missed linking of the pcap library. So loading ozmod_isdn failed with some kind of unknown symbol pcap_flush_dump error message. This keeps mod_openzap from loading at FS startup. regards helmut ___ 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] Strange Performance when using as SBC
Thanks Rod, Its really helpful contribution. @Nextone: I don't want to say much about it, but simply I am not happy with it, have you heard someone satisfied with NX who also owns it? Kind Regards Saeed Ahmed Tariq -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 3:33 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Saeed, Here is a first draft of what I did to install FS on my server. Configuration are not present, they'll be in a next release :p http://wiki.freeswitch.org/wiki/SBC_Setup My aim is to setup FS as a SBC, I hope this page could be a great startup point for others. I will update regularly based on what I did. Saeed, why are you replacing your Nextone, it's said to be one of the best commercial SBC on the market. regards. Saeed Ahmed wrote: Thanks rod for a quick answer, FS is installed on Ubuntu Server. I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to communicate with TDM but this all depends how much calls it can take, or maybe we can also do something in clustering environment ( I am not sure about it). But thanks again and any further help will be highly appreciated. Kind Regards Saeed Ahmed Tariq -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 1:53 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Saeed, I just created an account to share my setup on the wiki. I will detail all the steps for a clean install of a debian64 lenny with FS used as a SBC (next step is to try the new LCR module :) )and what I'm doing do stress the server. I wrote nothing at this time so please be patient, I'm waiting for my new hardware so that I will detail as much as possible what I'll do. For beginning I suggest you reading the start page on the wiki, especially these pages: -http://wiki.freeswitch.org/wiki/Getting_Started_Guide -http://wiki.freeswitch.org/wiki/Dialplan_XML maybe you could tell more about the linux distribution you're using so that I can give you some pointers for sipp... regards. rod. Saeed Ahmed wrote: Hi Rod, Could you please share how you configured Sipp FS to create a test environment? Especially the dial plan, sofia settings etc..., actually I am a newbie. I want to test it on a single FS machine. Kind Regards Saeed -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 11:00 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Ken, Jay, thanks for pointing to proxy media, I will test. Ken, you are right, I was brain damaged (a stupid mistake) when setting INFO cause this kind of level could be very verbose. I'm switching to CRIT or ERR. Thanks guys, rod. thanks for Ken Rice wrote: If you don't have to transcode, using proxy media mode will still save you some CPU time. This is 1/2 way between bypass media and the default media interactive mode. The other draw back to this mode is if you are using FS to clean up RTP and DTMF you loose those functions but they are not needed in most use cases. As far as the log level goes, I found that once I had things stable setting the loglevel to helped a good deal... Info is probably a bit too high of a loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you insist on leaving logging turned on... On a busy system these can and will generate a good deal of activity (and disk IO if using mod_logfile) Ken From: rod kawa...@laposte.net Reply-To: freeswitch-users@lists.freeswitch.org Date: Mon, 02 Feb 2009 11:36:35 +0400 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Ken, 1) I'd like to use FS to hide topology, so bypass media is not possible 2) done 3) done 4) not used 5) i'm using this ins switch.xml - param name=loglevel value=info/, if you think an other log level is more suitable. Regarding logging, I can see in console and in the freeswitch.log that there is still a lot of NOTICE logging, see below: 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8721 (sofia/internal/s...@10.10.10.1:5060) Ended 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel
[Freeswitch-users] Phonebooth?
I got a group email from Anders Brownworth this weekend regarding him donating Phonebooth to the FreePBX project? Wonder what the impact will be.. -- - Jason Gehman General Manager North Voice Communications www.NorthVC.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] LDAP Integration
On Jan 31, 2009, at 4:15 AM, John Skopis (Lists) wrote: Leon de Rooij wrote: Hi John, I've been trying to get your mod_xml_ldap module running, but didn't get very far yet.. What is the official way to get the module built ? The official way to build all fs modules is to uncomment the entry in modules.conf. If you want to build a specific module there are targets make mod_name-clean make mod_name-install Thanks, I'll try that. as for mod_xml_ldap, I really do not feel that it is as quality as I would expect a production quality module to be. I understand, it's just that I'm very interested in it as we're using ldap everywhere over here. I tried modifying trunk/freeswitch.spec so that XML_INT_MODULES contains xml_int/mod_xml_ldap There's also a directories/mod_ldap in DISABLED_MODULES in the same file, but I don't suppose it's necessary to enable it, or is it ? mod_ldap is a separate module, implementing the directory interface, not to be confused with the directory, which is queried for user + domain configuration (e.g., conf/directory/default.xml). perhaps it should be renamed to mod_dbi? The mod_xml_ldap doesn't get built by running make make or dpkg- buildpackage from trunk/ Also I tried building it from the module directory itself, but then I get the following error: fsbuil...@sv:~/trunk/src/mod/xml_int/mod_xml_ldap$ make Compiling mod_xml_ldap.c... cc1: warnings being treated as errors mod_xml_ldap.c: In function 'xml_ldap_search': mod_xml_ldap.c:356: warning: cast from pointer to integer of different size make[1]: *** [mod_xml_ldap.o] Error 1 make: *** [all] Error 1 I have been working on a new module called mod_entity that works off a simple description of an xml entitiy (domain, user, extension, condition, action, anti-action currently) querying a db backend via the directory interface for fields used to build the entity. It still needs a bit of work but I am hoping to get a patch together this weekend. I will post it to the freeswitch-dev list asking for comments. Off the top of my head at least the wishlist TODO is: implement connection pooling for mod_directory implement a cache either as a module used by an xml_int mod or in switch_xml to cache a switch_xml_t (Also I had to apt-get install libsasl2 libsasl2-dev, otherwise make from this dir errored with missing sasl/sasl.h) Can you see what I'm doing wrong ? (I'm using svn rev 11560) thanks regards, Leon On Jan 6, 2009, at 4:55 AM, John Skopis (Lists) wrote: Vinicius Kobashi wrote: hi ppl. i tried hard to make it work, but still i couldnt find a complete openldap scheme that provides these information, and i still could't find out where to put these configuration... can anyone help me? thankz! vinicius escreveu: thankz! ill set my openldap to provide these information.. but these about these binding settings... where should i set them? best regards John Skopis (Lists) wrote: vinicius wrote: hi ppl.. i tried to find something at google, but i couldnt manage to find anything. i still dont know what to do to make the mod_xml_ldap work. i couldnt find information about how to build a config file for the module, and where to store it... can anyone give me a help? Be advised mod_xml_ldap is probably not production quality and will undoubtedly change, eventually at least. Here is what I used once: bindings binding name=directory !--%s is populated with the extension -- param name=filter value=(FSid=%s) bindings=directory/ !--basedn for the searches %s is replaced with domain-- param name=basedn value=ou=people,dc=example / param name=url value=ldap://172.16.75.129; / param name=binddn value=cn=admin,dc=example / param name=bindpass value=secret / trans !-- we need to translate these attrs into FS attrs -- tran name=id mapfrom=FSid / tran name=mailbox mapfrom=FSmailbox / tran name=password mapfrom=FSPassword / tran name=vm-password mapfrom=FSvm-password / tran name=email-addr mapfrom=FSemail-addr / tran name=vm-email-all-messages mapfrom=FSvm-email-all- messages / tran name=vm-delete-file mapfrom=FSvm-delete-file / tran name=vm-attach-file mapfrom=FSvm-attach-file / /trans /binding binding name=configuration param name=filter value=(%s=%s) bindings=configuration/ param name=basedn value=name=%s,dc=example / param name=url value=ldap://172.16.75.129; / param name=binddn value=cn=admin,dc=example / param name=bindpass value=secret / /binding /bindings which should/probably/might work with ldap objects like these: dn: cn=John Skopis,ou=people,dc=example objectClass: person objectClass:
Re: [Freeswitch-users] Strange Performance when using as SBC
Rod, that wiki article is Awesome ! real good to see guides with start to finish steps. cant wait to see the next installment of your guide :) Jay On Tue, Feb 3, 2009 at 12:33 AM, rod kawa...@laposte.net wrote: Hi Saeed, Here is a first draft of what I did to install FS on my server. Configuration are not present, they'll be in a next release :p http://wiki.freeswitch.org/wiki/SBC_Setup My aim is to setup FS as a SBC, I hope this page could be a great startup point for others. I will update regularly based on what I did. Saeed, why are you replacing your Nextone, it's said to be one of the best commercial SBC on the market. regards. ___ 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 do I set my FS internal ip address to a static value.
Hi Gang, I've been struggleing with this also. Actually I can get it to bind to my address, the problem is it randomly drops my calls. :-( I have a FS running on a box with a static IP and I can start a call between two extensions and it will go for hours. Then I add anther interface say eth0:0 with a new static IP and reconfigure my phones and FS to use that, and the calls drop after about 15-20 mins. Though it's pretty random. Here is my setup. I have Debian Linux 2.6.23.1 kernel, and freeswitch-1.0.1. Here is my /etc/network/interfaces: # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) # The loopback interface auto lo iface lo inet loopback # The first network card - this entry was created during the Debian installation auto eth0 eth0:0 iface eth0 inet dhcp iface eth0:0 inet static address 192.168.0.249 netmask 255.255.255.0 gateway 192.168.0.254 The only change I made to the FS config is in Vars.xml. I added this line close to the top: X-PRE-PROCESS cmd=set data=local_ip_v4=192.168.0.249/ Here is the console log of the call being dropped: freeswi...@archive sofia status API CALL [sofia(status)] output: Name Type Data State = external profile sip:mod_so...@67.171.158.226:5080 RUNNING (0) internal profile sip:mod_so...@192.168.0.249:5060 RUNNING (2) nat profile sip:mod_so...@67.171.158.226:5070 RUNNING (0) defaultalias internal ALIASED outboundalias external ALIASED 192.168.0.249alias internal ALIASED = 3 profiles 3 aliases freeswi...@archive 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 sofia_glue_restart_all_profiles() Reload XML [Success] 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM Reloaded 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 sofia_read_frame() Hangup sofia/internal/1...@192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 switch_ivr_multi_threaded_bridge() Hangup sofia/internal/1...@192.168.0.249 [CS_EXECUTE] [NORMAL_CLEARING] 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 6 (sofia/internal/1...@192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) Ended 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1...@192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes [CS_HANGUP] 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 5 (sofia/internal/1...@192.168.0.249) Ended 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1...@192.168.0.249 [CS_HANGUP] 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [192.168.0.249] for profile [internal] 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [default] for profile [internal] 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started Profile internal [sofia_reg_internal] 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [outbound] for profile [external] 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started Profile external [sofia_reg_external] 2009-02-01 13:23:20 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started Profile nat [sofia_reg_nat] sofia status API CALL [sofia(status)] output: Name Type Data State = external profile sip:mod_so...@67.171.158.226:5080 RUNNING (0) internal profile sip:mod_so...@192.168.0.249:5060 RUNNING (0) outboundalias external ALIASED 192.168.0.249alias internal ALIASED nat profile sip:mod_so...@67.171.158.226:5070 RUNNING (0) defaultalias internal ALIASED = 3 profiles 3 aliases There is an older thread that says one
Re: [Freeswitch-users] Call Variable not available when call hangup
As I told you on IRC, the call is not completed at that stage... So there is no hangup time... You must post process the call or figure out your own start answer and stop times From: shehzad p pmh...@gmail.com Reply-To: freeswitch-users@lists.freeswitch.org Date: Mon, 2 Feb 2009 04:07:57 -0800 (PST) To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Call Variable not available when call hangup Hi all, I need to process some CDR variables in Dialplan, like call duration, Answered time etc. but when I place info application after bridge, it is not listing them properly as below: === Caller-Channel-Created-Time: [1233573341672157] Caller-Channel-Answered-Time: [1233573342712939] Caller-Channel-Hangup-Time: [0] == Here Hangup time is 0, So how can I find actual values? --I know that we can use xml_cdr or cdr_csv, but my current need is to get those values from dialplan itself so that can be passed to some script... thanks, msp -- View this message in context: http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p 21788550.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] Conference dialing and uuid
Loopback will not work in that case either. If the far end plays ringback inband you should hear that if you use the conference dial api call. /b On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: Aaah ok. Thanks for clearing that up. So using loopback is still the only real workable sollution for me, since that generates ringback from and alternative endpoint and plays it into the conference. I might play with some javascript that streams ring into the channel eventually but for now the string comparisons at least get me the right uuid. Thank you again, Sias ___ 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] Strange Performance when using as SBC
Hi Rod, Could you please share how you configured Sipp FS to create a test environment? Especially the dial plan, sofia settings etc..., actually I am a newbie. I want to test it on a single FS machine. Kind Regards Saeed -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 11:00 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Ken, Jay, thanks for pointing to proxy media, I will test. Ken, you are right, I was brain damaged (a stupid mistake) when setting INFO cause this kind of level could be very verbose. I'm switching to CRIT or ERR. Thanks guys, rod. thanks for Ken Rice wrote: If you don't have to transcode, using proxy media mode will still save you some CPU time. This is 1/2 way between bypass media and the default media interactive mode. The other draw back to this mode is if you are using FS to clean up RTP and DTMF you loose those functions but they are not needed in most use cases. As far as the log level goes, I found that once I had things stable setting the loglevel to helped a good deal... Info is probably a bit too high of a loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you insist on leaving logging turned on... On a busy system these can and will generate a good deal of activity (and disk IO if using mod_logfile) Ken From: rod kawa...@laposte.net Reply-To: freeswitch-users@lists.freeswitch.org Date: Mon, 02 Feb 2009 11:36:35 +0400 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Ken, 1) I'd like to use FS to hide topology, so bypass media is not possible 2) done 3) done 4) not used 5) i'm using this ins switch.xml - param name=loglevel value=info/, if you think an other log level is more suitable. Regarding logging, I can see in console and in the freeswitch.log that there is still a lot of NOTICE logging, see below: 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8721 (sofia/internal/s...@10.10.10.1:5060) Ended 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/s...@10.10.10.1:5060 [CS_HANGUP] 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8722 (sofia/external/9...@10.10.20.100) Ended 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/external/9...@10.10.20.100 [CS_HANGUP] 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() Channel [sofia/external/9...@10.10.20.100] has been answered 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() Changing codec ptime to 30. I bet you have a linksys/sipura =D Do you have any idea where I can switch off this kind of logging. I thought it should be in /dialplan/internal.xml, but I see that in internal.xml - param name=debug value=0/ thanks a lot for your suggestion. regards, rod Ken Rice wrote: Dont forget there are several things you can do to increase performance... 1) where possible use bypass media or media proxy modes 2) mount freeswitch/db as a ram drive (if you are using voicemail with the internal FS DBs you'll need a way to make this persistant across reboots) 3) see the wiki for setting reasonable ulimits 4) (this is my oppinion others may vary) dont use mod_cdr_csv 5) turn off (or reduce logging) in switch.conf.xml all of these thing can greatly improve performance. On Mon, Feb 2, 2009 at 1:04 AM, rod kawa...@laposte.net mailto:kawa...@laposte.net wrote: Thanks Anthony, the setup is like this: sipp server FS 1 FS2 FS1 is the AMD CPU that has only one extension in dialplan that bridges to FS2. is the first extension in FS2 dialplan that plays moh, FS2 has no CPU pbm. FS1 is maxing out at 60 bridged calls without your option -hp. Using -hp, I'm now able to bridge 200 concurrent calls (a great improvement) and the system is still reactive. CPU load is high but not 100% and as the system responds well, I think that doesn't matter. The 2GB of memory are completely consumed (top command shows 700MB for FS process). I understand that FS1 server is not the best hardware platform, and I'm waiting for new 4 cores server for testing. I will update those numbers when testing with the new hardware. regards, rod. Anthony Minessale wrote: Which of the 2 machines has the load issue? You said it was one box calling the other. You have 2 major things against you, single CPU and AMD, but you should at least be able to get in the vicinity of 800-1000 calls on a box like that.
[Freeswitch-users] FreeSwitch setup as a Dumb SBC
Hi Guys,
I've been working at setting up a couple of FreeSwitch nodes as a topology
hiding SBCs that handles both ingress traffic from my
providers/peers and pass traffic up to an openser router that then routes
call across the cluster of SBCs through which they reach the destination.
I have OpenSIPS/SER setup doing DB route lookups and ENUM with LCR/Serial
forking etc.
My question is what would be the best way to send a call out to a
destination choosen by the OpenSER router?
For example:
SIP Provider -- SBC --- OpenSER ( route lookup returns
123.123.123.4 as dest ) -- SBC --- 123.123.123.4
I was thinking something along the lines of adding a X-Route-To:
+1nxxnxxx...@123.123.123.4 with openser
and then something like this in the SBC.
context name=from-sipcore
extension name=outboundroute
action application=bridge
data=sofia/external/${sip_h_X-Route-To} /
/extension
/context
Is this a wise approach, is there anything I could do to do this better?
I'd like to keep the logic in the SBCs as simple as possible.
I am pretty familiar with SIP but my knowledge fades when it gets into the
nitty gritty of routing. ie the Contact: and Via: headers
and all that good stuff.
I should also state I have two profiles defined one for the internal/private
core network and one for the outside external network.
Any thoughts on this at all would be greatly appreciated.
Am I missing something in the SIP spec that would allow for this is a
standardized way?
Regards,
-Adam
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[Freeswitch-users] Application language to support C or C++?
Hi All, I saw the applications using FreeSwitch library can be written in JavaScript, Perl, Python and Lua but I need to use Linux C or C++ for applications, Is FreeSwitch can supported it? Where can I get the sample codes? My Linux platform is base on Fedora. Thanks a lot. Jason ___ 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] Application language to support C or C++?
FreeSwitch is written in C mainly and some things like mod_opal are written in C++, you can create your own modules in C... Grab the source and look around its pretty straight forward From: lee jason ja...@voicesession.com Reply-To: freeswitch-users@lists.freeswitch.org Date: Tue, 3 Feb 2009 10:21:25 +0800 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Application language to support C or C++? Hi All, I saw the applications using FreeSwitch library can be written in JavaScript, Perl, Python and Lua but I need to use Linux C or C++ for applications, Is FreeSwitch can supported it? Where can I get the sample codes? My Linux platform is base on Fedora. Thanks a lot. Jason ___ 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] FreeSwitch setup as a Dumb SBC
Yes you can do that, but there is nothing that says you cant have FreeSWITCH
just do those lookups and ENUM (FS Supports ENUM out of the box) and do the
exact same thing so it would work like Provider - ingress SBC - egress
SBC/Registration Server - customer
Loosing a whole hop in the process
From: Adam Long ajl...@worldlink.net
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Mon, 2 Feb 2009 22:05:17 -0500
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] FreeSwitch setup as a Dumb SBC
Hi Guys,
I¹ve been working at setting up a couple of FreeSwitch nodes as a topology
hiding SBCs that handles both ingress traffic from my
providers/peers and pass traffic up to an openser router that then routes
call across the cluster of SBCs through which they reach the destination.
I have OpenSIPS/SER setup doing DB route lookups and ENUM with LCR/Serial
forking etc.
My question is what would be the best way to send a call out to a
destination choosen by the OpenSER router?
For example:
SIP Provider -- SBC --- OpenSER ( route lookup returns
123.123.123.4 as dest ) -- SBC --- 123.123.123.4
I was thinking something along the lines of adding a ³X-Route-To:
+1nxxnxxx...@123.123.123.4² with openser
and then something like this in the SBC
context name=from-sipcore
extension name=outboundroute
action application=bridge
data=sofia/external/${sip_h_X-Route-To} /
/extension
/context
Is this a wise approach, is there anything I could do to do this better?
I¹d like to keep the logic in the SBCs as simple as possible.
I am pretty familiar with SIP but my knowledge fades when it gets into the
nitty gritty of routing ie the Contact: and Via: headers
and all that good stuff.
I should also state I have two profiles defined one for the internal/private
³core² network and one for the outside ³external² network.
Any thoughts on this at all would be greatly appreciated.
Am I missing something in the SIP spec that would allow for this is a
standardized way?
Regards,
-Adam
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Re: [Freeswitch-users] FreeSwitch setup as a Dumb SBC
Hi Adam,
I'm in the process of using FS as a SBC. For the route lookup, I do it
using OpenSER carrierroute, without having to flow through
SBC---Openser---SBC. I'm using carrierroute at this time cause I need
more than 200 000 routing entries and carrierroute has been tested with
twice this number.
Here is the setup:
- install openser and carrierroute and make openser listening on
127.0.0.1:5062 (for example) on your SBC
- populate carrierroute table
What I do to use carrierroute module from FS is to use a specific
X-header (X-LOOKUP).
In the dialplan, in the default context, I have something like this:
extension name=LOOKUP_ROUTE
condition field=destination_number expression=(\d+)$
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=export data=sip_h_X-ROUTE=LOOKUP/
action application=bridge
data=sofia/internal/${sip_req_us...@127.0.0.1:5062/
action application=export
data=sip_h_X-ROUTE=${sip_redirect_contact_host_0}/
action application=transfer data=${destination_number} XML ROUTING/
/condition
/extension
The process is simple:
the export sip_h_X-ROUTE=LOOKUP had a sip header X-ROUTE=LOOKUP
then I bridge the call to 127.0.0.1:5062 (openser process)
In openser I have a route block that checks the presence of header
LOOKUP and openser sends a 604: unable to route call if the prefix is
not found, or a 302: with the IP of the gateway found
In FS, you can get the IP using the variable
${sip_redirect_contact_host_0}. Then I transfer this to the context
ROUTING, where the check condition is based on the LOOKUP header that
has been rewritten with this variable.
I will document all this setup (installation of openser/carrierroute and
config file of FS and openser) on a wiki page I start writing yesterday,
so please be indulgent and patient.
The next step is to test the scalability of this.
I'm a very bad programmer, so that's the only way for me to contribute
to FS, and as I see many people interested for an SBC setup, I think it
could be great if we share our work/knowlegde.
The wiki page is there:
http://wiki.freeswitch.org/wiki/SBC_Setup
regards,
rod.
Adam Long wrote:
Hi Guys,
I’ve been working at setting up a couple of FreeSwitch nodes as a
topology hiding SBCs that handles both ingress traffic from my
providers/peers and pass traffic up to an openser router that then
routes call across the cluster of SBCs through which they reach the
destination.
I have OpenSIPS/SER setup doing DB route lookups and ENUM with
LCR/Serial forking etc.
My question is what would be the best way to send a call out to a
destination choosen by the OpenSER router?
For example:
SIP Provider -- SBC --- OpenSER ( route lookup returns
123.123.123.4 as dest ) -- SBC --- 123.123.123.4
I was thinking something along the lines of adding a “X-Route-To:
+1NXXNXX@ mailto:+1NXXNXX@123.123.123.4” with openser
and then something like this in the SBC…
context name=from-sipcore
extension name=outboundroute
action application=bridge data=sofia/external/${sip_h_X-Route-To} /
/extension
/context
Is this a wise approach, is there anything I could do to do this better?
I’d like to keep the logic in the SBCs as simple as possible.
I am pretty familiar with SIP but my knowledge fades when it gets into
the nitty gritty of routing… ie the Contact: and Via: headers
and all that good stuff.
I should also state I have two profiles defined one for the
internal/private “core” network and one for the outside “external”
network.
Any thoughts on this at all would be greatly appreciated.
Am I missing something in the SIP spec that would allow for this is a
standardized way?
Regards,
-Adam
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Re: [Freeswitch-users] Conference dialing and uuid
Actually loopback does work. however as I said it generates a pair of extra channels. Hmmm I was trying to generate and extra call to a JS script that generated a teletone ring in an on_ring_execute for the second call however it seems to stop execution of the call itself. Event though I use api commands to originate and then transfer it into the conference so that I have direct access to its uuid. I think changeing the moh might be a bit simpler however and elimite some CoreDB stuff I was doing to keep track of the calls ring generating call (what a sentance). On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: you could set the conference moh sound to be tone_stream::// with the teletone spec for ring sound and it use ignore_early_media=true in your originates so the first caller would hear ringback until the 2nd one arrived. On Mon, Feb 2, 2009 at 4:29 AM, Brian West [1]br...@freeswitch.org wrote: Loopback will not work in that case either. If the far end plays ringback inband you should hear that if you use the conference dial api call. /b On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: Aaah ok. Thanks for clearing that up. So using loopback is still the only real workable sollution for me, since that generates ringback from and alternative endpoint and plays it into the conference. I might play with some javascript that streams ring into the channel eventually but for now the string comparisons at least get me the right uuid. Thank you again, Sias ___ Freeswitch-users mailing list [2]freeswitch-us...@lists.freeswitch.org [3]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:[4]http://lists.freeswitch.org/mailman/options/freeswitch-u sers [5]http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH [6]http://www.freeswitch.org/ ClueCon [7]http://www.cluecon.com/ AIM: anthm [8]MSN:anthony_miness...@hotmail.com GTALK/JABBER/[9]PAYPAL:anthony.miness...@gmail.com IRC: [10]irc.freenode.net #freeswitch FreeSWITCH Developer Conference [11]sip:8...@conference.freeswitch.org [12]iax:gu...@conference.freeswitch.org/888 [13]googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 References 1. mailto:br...@freeswitch.org 2. mailto:Freeswitch-users@lists.freeswitch.org 3. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 4. http://lists.freeswitch.org/mailman/options/freeswitch-users 5. http://www.freeswitch.org/ 6. http://www.freeswitch.org/ 7. http://www.cluecon.com/ 8. mailto:msn%3aanthony_miness...@hotmail.com 9. mailto:paypal%3aanthony.miness...@gmail.com 10. http://irc.freenode.net/ 11. mailto:sip%3a...@conference.freeswitch.org 12. http://iax:gu...@conference.freeswitch.org/888 13. mailto:googletalk%3aconf%2b...@conference.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
