Re: [Freeswitch-users] Question about odbc support
Hello, hm kind of unclear Question. So I'm looking for a way to get the affected number of rows after executing a delete statement via ODBC. There is a function called SQLRowCount(), but I didn't found a switch_odbc_* function in FS which allows me to call it. On 18.11.2009 19:21, Helmut Kuper wrote: Hi, does anybody know how to check the affected rows caused by delete, insert or update sql statements in FS? To do this with sqlite3 there is a function called switch_core_db_changes(). 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 -- Mit freundlichen Grüßen Helmut Kuper Geschäftseinheit FD - Lösungen für Finanzdienstleister Telefax: (0441) 8000-2799 mailto:helmut.ku...@ewetel.de ___ EWE TEL GmbH Cloppenburger Straße 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Geschäftsführung: Hans-Joachim Iken (Vorsitzender), Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___ ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Question about odbc support
Hello, hm kind of unclear Question. So I'm looking for a way to get the affected number of rows after executing a delete statement via ODBC. There is a function called SQLRowCount(), but I didn't found a switch_odbc_* function in FS which allows me to call it. On 18.11.2009 19:21, Helmut Kuper wrote: Hi, does anybody know how to check the affected rows caused by delete, insert or update sql statements in FS? To do this with sqlite3 there is a function called switch_core_db_changes(). 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] XML config file parsing
Samuel Mukoti samuelmuk...@gmail.com wrote: I'm a new freeswitch user and am wondering what people do when setting options in the freeswitch config files. Do people use special tools, XML editors etc or is it just vi/emacs/Kate? Emacs has an XML editing mode; Vim may have extensions for handling XML as well. However, I have not found it necessary to invoke the XML features of an editor; just treating the configuration files as plain text is sufficient. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] store registration info in memcache
I believe OBDC is the official way.. however id love look at doing this in a higher performance way, without the single point of failure.. local memcache, in front of OBDC or something ?? not 100% sure of it, but just using a single central database is a little bit of a concern in a carrier environment. Jay On Thu, Nov 19, 2009 at 7:14 PM, Leon de Rooij l...@scarlet-internet.nlwrote: Hi, Not that I know of, but you can use odbc to store registrations and share it that way.. regards, Leon On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: Hi, Is there anyway to store registration info in memcache instead of sqlite? By doing that, it is possible for multiple freeswitch to share the same user registration info. Is there anyway I can intercept the registration success/failure event and write stuff to memcache myself? thanks, woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 -- Sincerely Jay ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] store registration info in memcache
If we could access mod_memcache for registration information that would be ideal and highly robust, since you can share memcache with external applications. Lon On Thu, Nov 19, 2009 at 2:07 AM, Leon de Rooij l...@scarlet-internet.nl wrote: Well, you can of course easily have a loadbalancer with failover in front of your sql servers and have them replicate to each other. Freeswitch will reconnect if a connection goes down. Perhaps failover is also possible directly through odbc ? Does anyone know if that's possible ? regards, Leon On Nov 19, 2009, at 10:33 AM, jay binks wrote: I believe OBDC is the official way.. however id love look at doing this in a higher performance way, without the single point of failure.. local memcache, in front of OBDC or something ?? not 100% sure of it, but just using a single central database is a little bit of a concern in a carrier environment. Jay On Thu, Nov 19, 2009 at 7:14 PM, Leon de Rooij l...@scarlet-internet.nl wrote: Hi, Not that I know of, but you can use odbc to store registrations and share it that way.. regards, Leon On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: Hi, Is there anyway to store registration info in memcache instead of sqlite? By doing that, it is possible for multiple freeswitch to share the same user registration info. Is there anyway I can intercept the registration success/failure event and write stuff to memcache myself? thanks, woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 -- Sincerely Jay ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Extension Configuration - XML File Entries for Group configuration
Hi, Can someone please help me understand a little more about Group configuration ? I believe that Group Membership is configured in the \conf\directory\default.xml file I've done this and the caller groups seem to work fine. However, each extension in the \conf\directory\default directory, e.g., 111.xml also has an entry for callgroup Can someone explain what the difference in these two options is please ? regards Dave___ FreeSWITCH-users mailing list FreeSWITCH-users@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] store registration info in memcache
I'd have to double check all the sql used for registration, but I doubt memcache is expressive enough to act as the registration store. For instance, you can't get a list of registrations from it (sofia status profile internal). memcache is a keystore only. That being said, one could use memcache as a umm.. well cache like it is designed as a front end to the real odbc database. Consult memcache first then hit the db. Doing anything like that would require moving much of mod_memcache up into core, something I promised I would do at one point but never got around to doing -- lack of time and motivation and no strong use case IMO. On Thu, Nov 19, 2009 at 4:23 AM, Lon Baker l...@kickasspixels.com wrote: If we could access mod_memcache for registration information that would be ideal and highly robust, since you can share memcache with external applications. Lon On Thu, Nov 19, 2009 at 2:07 AM, Leon de Rooij l...@scarlet-internet.nl wrote: Well, you can of course easily have a loadbalancer with failover in front of your sql servers and have them replicate to each other. Freeswitch will reconnect if a connection goes down. Perhaps failover is also possible directly through odbc ? Does anyone know if that's possible ? regards, Leon On Nov 19, 2009, at 10:33 AM, jay binks wrote: I believe OBDC is the official way.. however id love look at doing this in a higher performance way, without the single point of failure.. local memcache, in front of OBDC or something ?? not 100% sure of it, but just using a single central database is a little bit of a concern in a carrier environment. Jay On Thu, Nov 19, 2009 at 7:14 PM, Leon de Rooij l...@scarlet-internet.nl wrote: Hi, Not that I know of, but you can use odbc to store registrations and share it that way.. regards, Leon On Nov 19, 2009, at 9:40 AM, Woody Dickson wrote: Hi, Is there anyway to store registration info in memcache instead of sqlite? By doing that, it is possible for multiple freeswitch to share the same user registration info. Is there anyway I can intercept the registration success/failure event and write stuff to memcache myself? thanks, woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 -- Sincerely Jay ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 -- -Rupa ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 implement mod_lcr + mod_limit
Using lcr_auto_route + limit isn't really possible at this point. It is on the list of things to do but is more complex than it seems on it's surface. mod_lcr just constructs dial strings, it doesn't do any call control. It does provide enough information to do what you want via a scripting language like lua. mod_lcr sets channel vars lcr_route_count which tells you how many routes there are. It also sets lcr_route_N (where N is 1 to lcr_route_count) which contains each lcr route. You can then iterate over the routes, set limit try to bridge and loop until success. Arguably this should be done from within FS so that you could just use lcr_auto_route (assuming mod_lcr can pull limit info from the routes db). That is the plan but a workable solution hasn't magically appeared yet. On Mon, Nov 16, 2009 at 1:29 AM, Ahmed Munir ahmedmunir...@gmail.com wrote: Hi, I've worked on setup for carriers routing using mod_lcr + mod_nibble + mod_xml_curl and mod_xml_cdr. The setup is working fine as I desired. Now I want to include mod_limit in to my setup. As I read the wiki pages of mod_limit I want to know how can I limit the calls per destination basis while running mod_lcr? Because LCR is routing to different carriers, how can I call mod_limit in mod_lcr? Kindly advise this issue soon. -- Regards, Ahmed Munir ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] XML config file parsing
Thx Jason for the reply, I realise i was quite unclear in what i'm hoping to achieve. I wanted to make a control panel for our office so that we can provision extensions at the same time as we do users. We have a system much like the ubuntu ebox that allows use to manage users for our organization and for virtual domains - it uses postgresql as a backend. I'm not aware of freeswitch's abilities or features when it comes to databases. Can freeswitch lookup SQL tables in realtime? I would love the ability to manage dialplans, voicemail accounts, and extensions/endpoints thru a database much like mysql or postgresql The reason i was discussing XML is for this very same purpose, i though i could write helper scripts that would 'spit' out some XML configuration files thus dynamically updating Freeswitch configuration from a web frontend.. almost similar to what the freepbx.org guys have done. regards, Sam ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] XML config file parsing
Hi Sam, Take a look at mod_xml_curl. Pretty sure it'll do everything you're looking for. http://wiki.freeswitch.org/wiki/Mod_xml_curl Also, I would browse the modules and look for other nifty functionality that already exists before setting out to write something new. http://wiki.freeswitch.org/wiki/Modules Good luck! Rob On Nov 19, 2009, at 7:41 AM, Samuel Mukoti wrote: Thx Jason for the reply, I realise i was quite unclear in what i'm hoping to achieve. I wanted to make a control panel for our office so that we can provision extensions at the same time as we do users. We have a system much like the ubuntu ebox that allows use to manage users for our organization and for virtual domains - it uses postgresql as a backend. I'm not aware of freeswitch's abilities or features when it comes to databases. Can freeswitch lookup SQL tables in realtime? I would love the ability to manage dialplans, voicemail accounts, and extensions/endpoints thru a database much like mysql or postgresql The reason i was discussing XML is for this very same purpose, i though i could write helper scripts that would 'spit' out some XML configuration files thus dynamically updating Freeswitch configuration from a web frontend.. almost similar to what the freepbx.org guys have done. regards, Sam ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Need help configuring our FreeSWITCH instance
I have installed FreeSWITCH on our server, and need some help configuring our FreeSWITCH instance. All of the numbers associated with our FreeSWITCH instance are in the format: 1NPANXX (where NPA is the area code, and NXX are the last 7 digits of the phone number). I need the following configuration: Calls coming from our IP to IP gateway into our FreeSWITCH instance needs to be routed to the endpoint that is registered with FreeSWITCHCalls coming from any of the registered SIP endpoints need to be sent to the appropriate destination. The appropriate destination for any number that is not registered with FreeSWITCH is our IP to IP gateway.Our IP to IP gateway does not require any SIP registration or authentication.G.729 (but not G.729 Annex B), G.711 mu-law, and G.711 A-law need to be enabledSIP registrar enabled for registering endpoints other than our IP-IP gatewaySIP traffic needs to be accepted to and from both the IP-IP gateway and from the registered SIP endpoints. How do I get the above configured in FreeSWITCH? _ Windows 7: I wanted simpler, now it's simpler. I'm a rock star. http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Hardware echo cancellation.
Hi Brian, It just doesn't belong in user space or kernel space in the machine for true performance you should do it in hardware... I'm pretty sure the poor box would die if you tried it on 32 E1's at the same time. Disagree somewhat. The challenge that echo cancellers further from the hardware face is having some idea of the size of the buffers between the canceller and the wire; provided that this is known, or is small in comparison to the canceller's tail length, it can, in principle, go anywhere. All other things being equal, the right place for a software EC is in user space: can be done in a cross-platform way, can use FPU/MMX/SSE without guilt and voodoo, etc. And there is no reason why the same algorithm would perform differently if implemented in hardware or on the host CPU. And the OP only needed four E1s.. --Dave /b On Nov 18, 2009, at 5:39 PM, David Knell wrote: For the sort of box you're talking about (quad core++), this isn't lots; it's hardly any.. --Dave ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] TFTP Server Cisco 7540
If you are using Windows XP (or Vista for that matter), you may want to look at tftpd32. Its more compact and uses less memory than Solarwinds yet provides not only a tftp server but a dhcp and syslog server as well. In the past, I've use it to upgrade, install, and troubleshoot a variety of gear (dslams, routers, softswitches, SIP endpoints, etc.) when a dedicated server was not available. -metik Jeff Lenk wrote: Hi I run the SolarWinds TFTP server alongside FS on my small installation - works nicely! Jeff Dave Stevenson wrote: Hi, I have just about got FreeSwitch working with a Cisco 7940 Phone. After much reading, I worked out that I needed a TFTP server on the network that would supply the phone with it's SIP personality and config etc. I have been able to get the phone working and realise that the TFTP server needs to be available every time the phone loses power etc. At the moment, I have the TFTP server running on a temporary machine but it would be neater if it ran on the same machine as FreeSwitch. This will be a very small FreeSwitch installation, so, ... Is there any reason why I should not try to run FreeSwitch and the SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't think the server should put much load on the machine but wondered if there were any other reasons why this is a bad idea ? In addition, while I have the phone working - I get a status message on boot ... W310 2 Errors(s) Parsing SIPDefault.cnf Can anyone tell me how to locate the errors in this file please ? (I have posted it to the Pastebin) Regards Dave ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Call latency in conferences and echo test increases over time
Like I said, The timer by default is designed to make sure that none of the audio is lost for situations like FAX etc. There are parameters you can configure to disable the timers that I mentioned in the last email which will cause all of the audio to be jammed into your ear like twiddlebugs if you did you sleep test and brought it back. We do not use sleep for the timers we have timer objects into the code derived from a high priority thread sending conditional broadcasts to the timer objects. There is certainly a place where this can begin to break down but it has proven to provide reliable timing to thousands of concurrent channels. The auto-flush can get false positives in jittery situations is not always the best answer. What kind of CPU are you using and what kind of hardware that you suspect you are getting delayed cpu scheduling on a small number of calls? I appreciate your theory and I am willing to investigate a little for you but you must be aware we have put more than a few hours of thought into the architecture here. There may be a bigger problem with the eavesdropping which we can have a look at today because that does not sound right. On Thu, Nov 19, 2009 at 1:09 AM, Robert L Mathews li...@tigertech.comwrote: Anthony Minessale wrote: I can promise you that much of your problems will be solved with latest SVN. Okay, thanks! And in fact, I tried today's SVN, and it has fixed the problem with the conference, even without setting rtp-autoflush. Conferences now discard packets and catch up when they gets behind, even with only the default rtp-autoflush-during-bridge set. The echo test still suffers from the same problem unless rtp-autoflush is used, which I assume is simply because it's not considered a bridged call. Eavesdropping on an existing bridged call, then pressing 3 to turn it into a conference call, also requires rtp-autoflush to avoid persistent lag on the third leg. Did you answer the question about what phones? I'm going to guess Cisco based on the symptoms. It happens with all phones, as far as I can tell. I've tried at least Grandstream GXP2000, Grandstream BT102, SJPhone, Twinkle, and Express Talk (none of them Cisco). I'm fairly positive the problem is unrelated to phones; it's caused by delays in CPU scheduling of the server process. non bridge calls use a timer to make sure the audio is coming in at a steady rate to ensure bursting RTP is played at the correct rate. Stopping it and restarting 2 seconds later will cause a delay by design because you have suspended the process but not the UDP stack. Ummm well, a copy of FreeSWITCH running on any non-realtime operating system will occasionally not get scheduled for all the CPU time it wants. For example, it wouldn't be unusual for a thread to ask to sleep for 20 milliseconds but actually not wake up for 21, 25, or even 40 milliseconds because the server is busy with other things. Each time that happens, it's a smaller version of my artificial suspend test: the operating system has, of course suspended the process but not the UDP stack. It doesn't necessarily mean there's bursty network traffic or phone timing issues. Should FreeSWITCH really lag by that much for the rest of the call? 20 milliseconds here, 20 milliseconds there, and pretty soon you're talking about real seconds. I'm assuming the reason for making it catch up on bridged calls, but not non-bridged calls, is that people talking to each other can't tolerate high latency, but people listening to an IVR or something can. But is that still true if it gets seconds behind? And should the third leg of an eavesdrop-converted-to-three-way-call be considered non-bridged for this purpose? Anyway, given that current svn trunk fixes the problem by default in conferences and any other bridged call, I'm satisfied. And if anyone complains about this problem for non-bridged calls, I suppose they can enable rtp-autoflush to get the same catch-up behavior there. I took a shot at documenting these parameters in the wiki on: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#rtp-autoflush-during-bridge Thanks for the help! -- Robert L Mathews, Tiger Technologies ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
Re: [Freeswitch-users] Hardware echo cancellation.
On 11/19/2009 11:54 PM, David Knell wrote: Hi Brian, It just doesn't belong in user space or kernel space in the machine for true performance you should do it in hardware... I'm pretty sure the poor box would die if you tried it on 32 E1's at the same time. Disagree somewhat. The challenge that echo cancellers further from the hardware face is having some idea of the size of the buffers between the canceller and the wire; provided that this is known, or is small in comparison to the canceller's tail length, it can, in principle, go anywhere. All other things being equal, the right place for a software EC is in user space: can be done in a cross-platform way, can use FPU/MMX/SSE without guilt and voodoo, etc. And there is no reason why the same algorithm would perform differently if implemented in hardware or on the host CPU. And the OP only needed four E1s.. The audio path between kernel and user space is not stable with any current PC based telephony system. At some point in the day the odd chunk of data is lost here and there, whether you use asterisk, callweaver, yate or FS, with dahdi or sangoma. This is the key problem for user space echo cancellation. When the path hiccups, the EC goes crazy, and howls. So far kernel space EC has been the only way to keep the path length rock solid. There is an Intel development platform which tries to do EC with OSLEC in user space. That's the only delivered system I know that tries to do this. Its very quirky. 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] TFTP Server Cisco 7540
Metik, thanks a lot for the tip, I will certainly look at it, particularly if it does DHCP too. At the moment, I use my ADSL Router to provide DHCP to the network but I've just discovered that you can't configure options in its DHCP server to point to the TFTP server for the phone. At the moment, I have to have the phone set to a static IP address to be able to configure the TFTP server address which is not as flexible as using DHCP. I had thought about changing over to use Windows Server DHCP services but it sounds like ttpd32 would do the trick. I just need to decide whether I want all of my machines to rely on getting their IP address from another PC - it feels like having DHCP in the router is more robust. Regards Dave - Original Message - From: Metik freeswitch-users-l...@metik.com To: freeswitch-users@lists.freeswitch.org Sent: Thursday, November 19, 2009 4:01 PM Subject: Re: [Freeswitch-users] TFTP Server Cisco 7540 If you are using Windows XP (or Vista for that matter), you may want to look at tftpd32. Its more compact and uses less memory than Solarwinds yet provides not only a tftp server but a dhcp and syslog server as well. In the past, I've use it to upgrade, install, and troubleshoot a variety of gear (dslams, routers, softswitches, SIP endpoints, etc.) when a dedicated server was not available. -metik Jeff Lenk wrote: Hi I run the SolarWinds TFTP server alongside FS on my small installation - works nicely! Jeff Dave Stevenson wrote: Hi, I have just about got FreeSwitch working with a Cisco 7940 Phone. After much reading, I worked out that I needed a TFTP server on the network that would supply the phone with it's SIP personality and config etc. I have been able to get the phone working and realise that the TFTP server needs to be available every time the phone loses power etc. At the moment, I have the TFTP server running on a temporary machine but it would be neater if it ran on the same machine as FreeSwitch. This will be a very small FreeSwitch installation, so, ... Is there any reason why I should not try to run FreeSwitch and the SolarWinds Free TFTP Server on the same Windows XP Machine ? I don't think the server should put much load on the machine but wondered if there were any other reasons why this is a bad idea ? In addition, while I have the phone working - I get a status message on boot ... W310 2 Errors(s) Parsing SIPDefault.cnf Can anyone tell me how to locate the errors in this file please ? (I have posted it to the Pastebin) Regards Dave ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] TFTP Server Cisco 7540
Some Cisco phones need DHCP option 150. /b On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: Metik, thanks a lot for the tip, I will certainly look at it, particularly if it does DHCP too. At the moment, I use my ADSL Router to provide DHCP to the network but I've just discovered that you can't configure options in its DHCP server to point to the TFTP server for the phone. At the moment, I have to have the phone set to a static IP address to be able to configure the TFTP server address which is not as flexible as using DHCP. I had thought about changing over to use Windows Server DHCP services but it sounds like ttpd32 would do the trick. I just need to decide whether I want all of my machines to rely on getting their IP address from another PC - it feels like having DHCP in the router is more robust. Regards Dave ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] TFTP Server Cisco 7540
Yeah, roger that... Here is an excerpt from the page I did on the Cisco 7960G HowTo: http://wiki.freeswitch.org/wiki/Freeswitch_Cisco_7960G_Howto It's for Linux, but you'll get some good pointers on the TFTP option you're looking for. I haven't provisioned any 7540's... Good luck! Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Nov 19, 2009, at 11:55 AM, Brian West wrote: Some Cisco phones need DHCP option 150. /b On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: Metik, thanks a lot for the tip, I will certainly look at it, particularly if it does DHCP too. At the moment, I use my ADSL Router to provide DHCP to the network but I've just discovered that you can't configure options in its DHCP server to point to the TFTP server for the phone. At the moment, I have to have the phone set to a static IP address to be able to configure the TFTP server address which is not as flexible as using DHCP. I had thought about changing over to use Windows Server DHCP services but it sounds like ttpd32 would do the trick. I just need to decide whether I want all of my machines to rely on getting their IP address from another PC - it feels like having DHCP in the router is more robust. Regards Dave ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] TFTP Server Cisco 7540
I don't think a 7540 exists. /b On Nov 19, 2009, at 12:11 PM, Karl J. Vesterling wrote: I haven't provisioned any 7540's... Good luck! ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] TFTP Server Cisco 7540
Thanks Guys, I had not realised until the last couple of days that DHCP did more than just providing the IP address to the client. I have been happily just doing that for a few years now without anything other than my Router providing the DHCP function. It's only now that I have taken the plunge into IP telephony that I realise that it can do more and for Cisco phones, should provide the address of the TFTP server. My work-around at the moment is to used fixed IP addresses in the phone for it's own IP address and the TFTP server - not as neat as I would like, but it works. I will look at a better long term solution with a different DHCP server (as already mentioned earlier in this thread). Looking on the bright side, I have got the phone provisioned - though I'm still working out what all the options are, but it is working. As Brian has spotted - my reference to a 7540 was an error - I got in right in the body of the original post, but not when I edited the subject line - ps - sorry. The phone is a 7940 ! regards Dave - Original Message - From: Karl J. Vesterling k...@ken-ton.com To: freeswitch-users@lists.freeswitch.org Sent: Thursday, November 19, 2009 6:11 PM Subject: Re: [Freeswitch-users] TFTP Server Cisco 7540 Yeah, roger that... Here is an excerpt from the page I did on the Cisco 7960G HowTo: http://wiki.freeswitch.org/wiki/Freeswitch_Cisco_7960G_Howto It's for Linux, but you'll get some good pointers on the TFTP option you're looking for. I haven't provisioned any 7540's... Good luck! Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Nov 19, 2009, at 11:55 AM, Brian West wrote: Some Cisco phones need DHCP option 150. /b On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: Metik, thanks a lot for the tip, I will certainly look at it, particularly if it does DHCP too. At the moment, I use my ADSL Router to provide DHCP to the network but I've just discovered that you can't configure options in its DHCP server to point to the TFTP server for the phone. At the moment, I have to have the phone set to a static IP address to be able to configure the TFTP server address which is not as flexible as using DHCP. I had thought about changing over to use Windows Server DHCP services but it sounds like ttpd32 would do the trick. I just need to decide whether I want all of my machines to rely on getting their IP address from another PC - it feels like having DHCP in the router is more robust. Regards Dave ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] mod dptools record problem - hangup channel with invalid file path
I had almost the same problem- it was needed to record everything, even if the record path doesn't exist - it was requested to create the needed path. For this purpose I've used event_socket command api system ..., precisely, api system mkdir -p path And after this command I've started recording. So, you may the same approach. On Wed, Nov 18, 2009 at 11:26 PM, William Kendi ... william.nis...@voicetechnology.com.br wrote: Actually, I am integrating FreeSWITCH with a weird IVR Framework, and the current behaviour of the mod dptools record application breaks some rules of the weird IVR Framework that must be integrated with FreeSWITCH. In order to integrate FreeSWITCH with the weird IVR Framework, the mod dptools record application mustn't terminate the call when invalid file paths are passed, and a notification of the invalid file path through the event system of FreeSWITCH should be enough for me, like the behaviour of the mod dptools playback application when invalid file paths are passed. Thanks in advance. ** 2009/11/18 Michael Jerris m...@jerris.com Okay, I'll ask the obvious question. Why are you passing record invalid file paths and why should it not fail if you do? Mike On Nov 18, 2009, at 2:38 PM, William Kendi ... wrote: While I was testing the mod dptools record application using invalid file paths, i noted that the mod dptools record application terminated the call. I am currently looking for a way to change this behaviour. Any suggestions? Can this be done? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Call doesn't work while registration work for a VOIP provider
Hi, i'm trying to configure freeswitch with a VOIP provider, exsorsa, that uses OpenSER. Exsorsa use as own gateway, another provider, Eutelia, that it uses Cisco (or, at least, this appears in headers). Short story: If i try to setup my Eutelia account all works perfectly while if i try to setup Exsorsa account registration works fine while calling not: when fs send the ACK, as answer to a OK (sip code 200), that is sended from exsorsa as answer to an INVITE, exsorsa send back a BYE. Long story: --- I put call log on pastebin with debug and sip_trace enabled for external sip_profile and with log level on debug on fs console. Registration log, here all is ok (or at least it seems to be ok) http://pastebin.freeswitch.org/11176 Annoyng message that comes up every 30 seconds http://pastebin.freeswitch.org/11177 Call log http://pastebin.freeswitch.org/11178 As you can see from call log all works fine until fs send back the acknowledgment message (line 451 on last log). Can this depend on the annoyng message that comes up every 30 seconds? Here my external sip profile config http://pastebin.freeswitch.org/11180 while here exsorsa gateway config http://pastebin.freeswitch.org/11181 Any helps is really appreciated! I'm fought with it all the day!!! Best Regards, Daniele attachment: info.vcf___ FreeSWITCH-users mailing list FreeSWITCH-users@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 doesn't work while registration work for a VOIP provider
I'm going to guess gw+exsorsa is what they don't like. try extensions- in-contact=true on the gateway config. /b On Nov 19, 2009, at 1:29 PM, Albano Daniele Salvatore - Lavoro wrote: Hi, i'm trying to configure freeswitch with a VOIP provider, exsorsa, that uses OpenSER. Exsorsa use as own gateway, another provider, Eutelia, that it uses Cisco (or, at least, this appears in headers). Short story: If i try to setup my Eutelia account all works perfectly while if i try to setup Exsorsa account registration works fine while calling not: when fs send the ACK, as answer to a OK (sip code 200), that is sended from exsorsa as answer to an INVITE, exsorsa send back a BYE. Long story: --- I put call log on pastebin with debug and sip_trace enabled for external sip_profile and with log level on debug on fs console. Registration log, here all is ok (or at least it seems to be ok) http://pastebin.freeswitch.org/11176 Annoyng message that comes up every 30 seconds http://pastebin.freeswitch.org/11177 Call log http://pastebin.freeswitch.org/11178 As you can see from call log all works fine until fs send back the acknowledgment message (line 451 on last log). Can this depend on the annoyng message that comes up every 30 seconds? Here my external sip profile config http://pastebin.freeswitch.org/11180 while here exsorsa gateway config http://pastebin.freeswitch.org/11181 Any helps is really appreciated! I'm fought with it all the day!!! Best Regards, Daniele info.vcf___ FreeSWITCH-users mailing list FreeSWITCH-users@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] uuid_bridge kills both channels if they are executing java app
Hi there! I've got annoying FS behavior: There are 2 channels executing the same Java application (application itself is an IVR). If I try to bridge them with uuid_bridged then both channels are killed. Here is a log from FS console: uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 (sofia/internal/ 1...@192.168.147.130) State Change CS_EXECUTE - CS_HIBERNATE 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.147.130 [BREAK] 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 (sofia/internal/ 1...@master.agent.starpoundtech.net) State Change CS_EXECUTE - CS_HIBERNATE 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de freeswi...@localhost.localdomain 2009-07-09 05:58:26.674348 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@master.agent.starpoundtec 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1...@192.168.147.130 [BREAK] 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1...@master.agent.starpoundtech.net [BREAK] (FS version is 1.0.4) Any thoughts? Artem ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Want 183 w/SDP, but Get 200 w/SDP
Hello, I just pasted a log in the Pastebin with Freeswitch logging enabled. Does anyone know a way to prevent FS from connecting the call prior to the callee answering? Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Thursday, November 05, 2009 3:50 PM To: 'freeswitch-users@lists.freeswitch.org' Subject: Want 183 w/SDP, but Get 200 w/SDP I am trying to make a call through a Gateway that sends the INVITE with no SDP and ONLY wants the 200 OK w/SDP when the callee answers. For some reason, Freeswitch answers the call with 200 OK w/SDP even before the callee answers the phone. Is this to provide ringback? Can I disable that action? Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP
set enable-3pcc to proxy instead of true On Thu, Nov 19, 2009 at 2:00 PM, Jerry Richards jerry.richa...@teotech.comwrote: Hello, I just pasted a log in the Pastebin with Freeswitch logging enabled. Does anyone know a way to prevent FS from connecting the call prior to the callee answering? Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Thursday, November 05, 2009 3:50 PM To: 'freeswitch-users@lists.freeswitch.org' Subject: Want 183 w/SDP, but Get 200 w/SDP I am trying to make a call through a Gateway that sends the INVITE with no SDP and ONLY wants the 200 OK w/SDP when the callee answers. For some reason, Freeswitch answers the call with 200 OK w/SDP even before the callee answers the phone. Is this to provide ringback? Can I disable that action? Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn: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] Extension Configuration - XML File Entries for Group configuration
On Thu, Nov 19, 2009 at 3:33 AM, Dave Stevenson steve...@primrosebank.netwrote: Hi, Can someone please help me understand a little more about Group configuration ? I believe that Group Membership is configured in the \conf\directory\default.xml file I've done this and the caller groups seem to work fine. However, each extension in the \conf\directory\default directory, e.g., 111.xml also has an entry for callgroup Can someone explain what the difference in these two options is please ? The groups defined in conf/directory/default.xml correspond to the group channel or group_call API as can be found in conf/dialplan/default.xml, extensions 2000, 2001, and 2002. Go to the fs_cli and type this: group_call sa...@1.1.1.1 (where 1.1.1.1 is your FS IP addr) You'll see that it returns a nicely formatted multiple dialstring for dialing everyone in the group. These have nothing to do with the callgroup variable that is defined on each user in the default directory. That is just a variable - it isn't required and doesn't have to be used, but it's available if you want it for some reason. (For example, it will show up in XML CDRs for auth'd calls from the user.) Bottom line: if you're trying to dial multiple users (i.e. group call) then just use the group definitions in the directory and use either the group_call API (like in ext 2000) or use the group channel (like in ext 2001 and 2002). -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] Want 183 w/SDP, but Get 200 w/SDP
On Thu, Nov 19, 2009 at 12:18 PM, Anthony Minessale anthony.miness...@gmail.com wrote: set enable-3pcc to proxy instead of true FYI, the wiki entry is here: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#enable-3pcc ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] uuid_bridge kills both channels if they are executing java app
On Thu, Nov 19, 2009 at 11:46 AM, Artem Shiyanov shiya...@gmail.com wrote: Hi there! I've got annoying FS behavior: There are 2 channels executing the same Java application (application itself is an IVR). If I try to bridge them with uuid_bridged then both channels are killed. Here is a log from FS console: uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 (sofia/internal/1...@192.168.147.130) State Change CS_EXECUTE - CS_HIBERNATE 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.147.130 [BREAK] 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 (sofia/internal/1...@master.agent.starpoundtech.net) State Change CS_EXECUTE - CS_HIBERNATE 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de freeswi...@localhost.localdomain 2009-07-09 05:58:26.674348 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@master.agent.starpoundtec 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1...@192.168.147.130 [BREAK] 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1...@master.agent.starpoundtech.net [BREAK] (FS version is 1.0.4) Any thoughts? First, update to latest trunk - there are many behaviors that have been tweaked and repaired since early August when 1.0.4 came out. Try it on latest trunk and see if the behavior persists, is different, or is gone. Please report back and let us know how it all goes. -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] uuid_bridge kills both channels if they are executing java app
I don't see any hangups here, are you talking about the BREAK signals? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 19-Nov-09, at 11:46 AM, Artem Shiyanov wrote: Hi there! I've got annoying FS behavior: There are 2 channels executing the same Java application (application itself is an IVR). If I try to bridge them with uuid_bridged then both channels are killed. Here is a log from FS console: uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c- a9ae-763d607e43de 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 (sofia/internal/1...@192.168.147.130 ) State Change CS_EXECUTE - CS_HIBERNATE 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.147.130 [BREAK] 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 (sofia/internal/1...@master.agent.starpoundtech.net ) State Change CS_EXECUTE - CS_HIBERNATE 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de freeswi...@localhost.localdomain 2009-07-09 05:58:26.674348 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@master.agent.starpoundtec 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1...@192.168.147.130 [BREAK] 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1...@master.agent.starpoundtech.net [BREAK] (FS version is 1.0.4) Any thoughts? Artem ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Extension Configuration - XML File Entriesfor Group configuration
Thanks Michael, I think I've got it ! regards Dave - Original Message - From: Michael Collins To: freeswitch-users@lists.freeswitch.org Sent: Thursday, November 19, 2009 8:25 PM Subject: Re: [Freeswitch-users] Extension Configuration - XML File Entriesfor Group configuration On Thu, Nov 19, 2009 at 3:33 AM, Dave Stevenson steve...@primrosebank.net wrote: Hi, Can someone please help me understand a little more about Group configuration ? I believe that Group Membership is configured in the \conf\directory\default.xml file I've done this and the caller groups seem to work fine. However, each extension in the \conf\directory\default directory, e.g., 111.xml also has an entry for callgroup Can someone explain what the difference in these two options is please ? The groups defined in conf/directory/default.xml correspond to the group channel or group_call API as can be found in conf/dialplan/default.xml, extensions 2000, 2001, and 2002. Go to the fs_cli and type this: group_call sa...@1.1.1.1 (where 1.1.1.1 is your FS IP addr) You'll see that it returns a nicely formatted multiple dialstring for dialing everyone in the group. These have nothing to do with the callgroup variable that is defined on each user in the default directory. That is just a variable - it isn't required and doesn't have to be used, but it's available if you want it for some reason. (For example, it will show up in XML CDRs for auth'd calls from the user.) Bottom line: if you're trying to dial multiple users (i.e. group call) then just use the group definitions in the directory and use either the group_call API (like in ext 2000) or use the group channel (like in ext 2001 and 2002). -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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Another Group Question - on VoiceMail
Hi again ! I have FreeSwitch configured such that if someone dials in from the PSTN line, a group of phones ring. If nobody answers, the group extension number (100) picks up the call and voice mail kicks in. So far, so good, each of the individual phones logs a missed call and anyone in the group can call into voice mail and go to the extension 100 mailbox to check if there are any messages but the individual phones are not notified that a Voice message is waiting. Is there any way that each extension in the group can be notified that a group Voice Mail is waiting to be picked up so that each phone shows the message waiting indication ? Regards Dave ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] mod_bv16/32 removed. Added mod_bv
We have removed the two modules using the reference code from BroadVoice and added a lib with a new interface from Steve Underwood and mod_bv.c using this lib... We know their is ONE last bug to be fixed in the lib before its working so please do not open any jira's if you try to run it because it will crash right now. Thanks for your understanding and once this is fixed it'll work with aastra and x-lite on both 32bit and 64bit systems without any issues. Thanks, Brian West ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Need help configuring our FreeSWITCH instance
On Wed, Nov 18, 2009 at 6:54 PM, John Platts john_pla...@hotmail.comwrote: I have installed FreeSWITCH on our server, and need some help configuring our FreeSWITCH instance. All of the numbers associated with our FreeSWITCH instance are in the format: 1NPANXX (where NPA is the area code, and NXX are the last 7 digits of the phone number). I need the following configuration: - Calls coming from our IP to IP gateway into our FreeSWITCH instance needs to be routed to the endpoint that is registered with FreeSWITCH - Calls coming from any of the registered SIP endpoints need to be sent to the appropriate destination. The appropriate destination for any number that is not registered with FreeSWITCH is our IP to IP gateway. - Our IP to IP gateway does not require any SIP registration or authentication. - G.729 (but not G.729 Annex B), G.711 mu-law, and G.711 A-law need to be enabled - SIP registrar enabled for registering endpoints other than our IP-IP gateway - SIP traffic needs to be accepted to and from both the IP-IP gateway and from the registered SIP endpoints. How do I get the above configured in FreeSWITCH? I'd say you have two choices: roll up your sleeves and start learning or email consult...@freeswitch.org and get some paid help. All of the questions you asked are answered in the wiki (and in some cases, mailing list history) but the answers require some foundational knowledge for them to make sense. If you are not a VoIP user then I'd recommend going the paid route and getting a professional to assist you - it will be the fastest way to get up and running. -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] mod_bv16/32 removed. Added mod_bv
On Thu, Nov 19, 2009 at 12:49 PM, Brian West br...@freeswitch.org wrote: We have removed the two modules using the reference code from BroadVoice and added a lib with a new interface from Steve Underwood and mod_bv.c using this lib... We know their is ONE last bug to be fixed in the lib before its working so please do not open any jira's if you try to run it because it will crash right now. Thanks for your understanding and once this is fixed it'll work with aastra and x-lite on both 32bit and 64bit systems without any issues. Thanks, Brian West Thanks to Brian, Tony, Mike, and Steve U. for all their hard work on this. Not only did they get this implemented quickly, they found a few bugs and reported back to the Broadcom guys. :) Excellent work all the way around. -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] Call latency in conferences and echo test increases over time
Anthony Minessale wrote: Like I said, The timer by default is designed to make sure that none of the audio is lost for situations like FAX etc. Right, that makes sense. I've updated the wiki entries I made to warn about this. We do not use sleep for the timers we have timer objects into the code derived from a high priority thread sending conditional broadcasts to the timer objects. Sorry for not being clear. When I said it sleeps, I just meant the operating system isn't scheduling any FreeSWITCH threads to run for some period of time, for whatever reason. What kind of CPU are you using and what kind of hardware that you suspect you are getting delayed cpu scheduling on a small number of calls? Well, I'm using 2.4 GHz dual Xeons, but couldn't this situation happen on any hardware, if it also has non-FreeSWITCH processes consuming lots of CPU time? That's because the timer needs to make sure that rtp_common_read() is called at least once every 20 ms. If it can't be called that often, for any reason, then FreeSWITCH will fall behind the RTP stream. At that point, audio latency will certainly increase unless some of the packets are discarded. I could duplicate the latency on 1.0.4 by running many other non-FreeSWITCH processes on the same server, so that all the freeswitch threads get starved for CPU time. FreeSWITCH then can't read the RTP packets as fast as they come in, and since the 1.0.4 code didn't flush those extra packets in conferences, that caused noticeable latency. Imposing heavy server load is obviously a silly thing to do, but something similar could happen on any server that fires up lots of non-FreeSWITCH, CPU-hungry processes. (In my case it was virus scanners.) Not using a dedicated server is also silly if people care about call quality, but I was just initially using it for conferences, and I didn't care if some packets were dropped. But conference packet dropping didn't happen on 1.0.4. Instead, a noticeable lag developed, which I did care about. Since 1.0.5 *does* work the way I expect in conferences and other bridged calls (discarding packets), I'm *definitely* not complaining -- please consider this a resolved issue! I agree that it makes sense to preserve all packets for some RTP streams such as faxes and DTMF recognition, and basing that decision on whether the call is bridged makes as much sense as anything else I can think of (although perhaps that flag isn't getting set properly for the third leg of eavesdrop-converted-to-three-way calls). I've been impressed by the extremely high performance of FreeSWITCH. The conference latency I was hearing in 1.0.4 was caused by the fact that I'm stressing the server with separate, unrelated processes, which is a foolish thing to do if you care about audio quality. I was just hoping that FreeSWITCH could more gracefully deal with such foolishness in cases where people *don't* care about audio quality... and 1.0.5 does. That's perfect. Thanks again! -- Robert L Mathews, Tiger Technologies ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Hardware echo cancellation.
On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote: The audio path between kernel and user space is not stable with any current PC based telephony system. At some point in the day the odd chunk of data is lost here and there, whether you use asterisk, callweaver, yate or FS, with dahdi or sangoma. This is the key problem for user space echo cancellation. When the path hiccups, the EC goes crazy, and howls. So far kernel space EC has been the only way to keep the path length rock solid. Why do you think this is? Getting data from kernel space to user space isn't something which it's difficult to do reliably: the disk system manages it. Even if data is being lost, buffer overruns can be dealt with by using bigger buffers, or timestamping blocks of data on their way in so that missing blocks can be detected. --Dave ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Radius for registration
Hi everyone, I want to verify what the wiki says, you can use a radius server as the data source for your registrations? Lon ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Another Group Question - on VoiceMail
Is there any way that each extension in the group can be notified that a group Voice Mail is waiting to be picked up so that each phone shows the message waiting indication ? Wouldn't this be simply accomplished by setting the vicemail as box 100 for each of the users (such as ext 1011xx)? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Another Group Question - on VoiceMail
Thanks Joseph, that would be one way, but it would mean that everyone had a common mailbox for all calls, I just wanted to do it for calls coming in on the PSTN line. Maybe that's not possible though ? regards Dave - Original Message - From: Joseph L. Casale jcas...@activenetwerx.com To: freeswitch-users@lists.freeswitch.org Sent: Thursday, November 19, 2009 10:26 PM Subject: Re: [Freeswitch-users] Another Group Question - on VoiceMail Is there any way that each extension in the group can be notified that a group Voice Mail is waiting to be picked up so that each phone shows the message waiting indication ? Wouldn't this be simply accomplished by setting the vicemail as box 100 for each of the users (such as ext 1011xx)? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] TFTP Server Cisco 7540
He should be able to just use Additional Option to add option 150 (and the associated IP address to which the TFTP server is bound). Brian West wrote: Some Cisco phones need DHCP option 150. /b On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote: Metik, thanks a lot for the tip, I will certainly look at it, particularly if it does DHCP too. At the moment, I use my ADSL Router to provide DHCP to the network but I've just discovered that you can't configure options in its DHCP server to point to the TFTP server for the phone. At the moment, I have to have the phone set to a static IP address to be able to configure the TFTP server address which is not as flexible as using DHCP. I had thought about changing over to use Windows Server DHCP services but it sounds like ttpd32 would do the trick. I just need to decide whether I want all of my machines to rely on getting their IP address from another PC - it feels like having DHCP in the router is more robust. Regards Dave ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Media got stuck after attended transfer...
Hi, one of my customers is willing to contribute for t38 integration. The basic idea is to connect HylaFAX to FS: t38modem - FreeSWITCH - Media Gateway with t38 support All this without media proxy. Another idea might be to implement t38 origination/termination with a class 1 modem input/output for use with HylaFAX. Do you know how much money we need to collect for t38 support? How much time is needed for implementing this? Thanks, Klaus From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, October 16, 2009 2:10 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Media got stuck after attended transfer... On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga tculj...@gmail.commailto:tculj...@gmail.com wrote: hi, any clue when can t38 be added? Eventually. :) Of course, if we could get more to add to the bounty it might grease the wheels of innovation. http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch Of course, I was listening to my A.M radio the other day and they said that there was this new invention called the Internet that would let people send documents to each other electronically. Maybe you should look into that. Next thing you know they'll come up with telephones that people don't have to plug into the wall and can take with them in the car. ;) -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
[Freeswitch-users] RTP issues (possibly nat-related)
I have upgraded FreeSWITCH several times recently for testing purposes. Also, my router's configuration has changed slightly as I have moved from tunneled IPv6 to a new native IPv6-over-ADSL trial. However, the problem now is related to my ISP's IPv4-only SIP service, and the symptoms are as follows. 1. If I call a test number, sometimes it all works perfectly. 2. On other occasions (with no discernible pattern) the call connects but no audio is received from the remote end. When this occurs, tshark shows that rtp packets are being sent out to the correct IPv4 address of the server. I am using Stun to handle nat, as my router does not support any of the nat configuration protocols. I want to establish whether it's a router issue or a FreeSWITCH problem. The router is going to be replaced eventually with a small form-factor Linux box and an ADSL2+ card from Traverse Technologies (http://www.traverse.com.au/), but given my priorities at the moment, it won't happen until next year. I can compare SIP traces of that would help. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] RTP issues (possibly nat-related)
I think the fix for this is coming to an SVN repo near you... so give it a few and update. /b On Nov 19, 2009, at 7:15 PM, Jason White wrote: I have upgraded FreeSWITCH several times recently for testing purposes. Also, my router's configuration has changed slightly as I have moved from tunneled IPv6 to a new native IPv6-over-ADSL trial. However, the problem now is related to my ISP's IPv4-only SIP service, and the symptoms are as follows. 1. If I call a test number, sometimes it all works perfectly. 2. On other occasions (with no discernible pattern) the call connects but no audio is received from the remote end. When this occurs, tshark shows that rtp packets are being sent out to the correct IPv4 address of the server. I am using Stun to handle nat, as my router does not support any of the nat configuration protocols. I want to establish whether it's a router issue or a FreeSWITCH problem. The router is going to be replaced eventually with a small form-factor Linux box and an ADSL2+ card from Traverse Technologies (http://www.traverse.com.au/), but given my priorities at the moment, it won't happen until next year. I can compare SIP traces of that would help. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Hardware echo cancellation.
On 11/20/2009 05:15 AM, David Knell wrote: On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote: The audio path between kernel and user space is not stable with any current PC based telephony system. At some point in the day the odd chunk of data is lost here and there, whether you use asterisk, callweaver, yate or FS, with dahdi or sangoma. This is the key problem for user space echo cancellation. When the path hiccups, the EC goes crazy, and howls. So far kernel space EC has been the only way to keep the path length rock solid. Why do you think this is? Getting data from kernel space to user space isn't something which it's difficult to do reliably: the disk system manages it. Even if data is being lost, buffer overruns can be dealt with by using bigger buffers, or timestamping blocks of data on their way in so that missing blocks can be detected. Disk isn't audio. Audio is real time, and real time constraints are a harsh mistress. Big buffers are out of the question, due to latency. Some mitigation could be provided if you can detect where missing chunks occur and their exact size. Right now, the I/O schemes do not provide for that, and incorporating support would be tough. You'd need some out of band indication, like an ioctl or something, which would lead to more user space/kernel space exchanges, further increasing the problem. 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_bv16/32 removed. Added mod_bv
It now works.. update and have fun! /b On Nov 19, 2009, at 3:01 PM, Michael Collins wrote: On Thu, Nov 19, 2009 at 12:49 PM, Brian West br...@freeswitch.org wrote: We have removed the two modules using the reference code from BroadVoice and added a lib with a new interface from Steve Underwood and mod_bv.c using this lib... We know their is ONE last bug to be fixed in the lib before its working so please do not open any jira's if you try to run it because it will crash right now. Thanks for your understanding and once this is fixed it'll work with aastra and x-lite on both 32bit and 64bit systems without any issues. Thanks, Brian West Thanks to Brian, Tony, Mike, and Steve U. for all their hard work on this. Not only did they get this implemented quickly, they found a few bugs and reported back to the Broadcom guys. :) Excellent work all the way around. -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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] APT Utility
Thanks for your answers Rob and Shelby. I found more info on apt-get and ran it against all the missing dependences noted. I also ran through the sequence of commands Shelby suggested. In the end, running dpkg -checkbuilddeps I got the following in return dpkg-checkbuilddeps: Unnet builddependencies: debhelper (=5) then followed the instructions for Ubuntu to enable freeswitch nano /etc/default/freeswitch FREESWITCH_ENABLE=true And then tried invoke -rc.d freeswitch start but nothing obvious happened. I am only using Ubuntu since it came as a free DVD in the Linux Pro mag that the article about Freeswitch was in. Is there a better version of Linux to use? thanks David David V. Fansler s/v Annabelle dfans...@dv-fansler.com www.dv-fansler.com From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman Sent: Wednesday, November 18, 2009 5:53 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] APT Utility Hi David, When using Apt, you would install packages with: apt-get install package name Or search for packages with apt-cache search search term If you're not root, you'll need to stick sudo in front of those command. Honestly, you might want to find a better tutorial with explicit command-by-command instructions. Good luck! Rob On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: Greetings - I am trying to startup a freeSwitch on a P4 running Ubuntu 9.04 Jaunty. I know very little about Linux. I decided to try this after reading the article in Linux Pro Magazine. I have been following the detailed instructions in the wiki for using Ubuntu Jaunty, however I have run into an unknown - Use your favorite APT utility to get the needed packages. I am good at following direct instructions - but this statement is too vague for my minimal minimal - did I mention minimal - knowledge of Linux. Could someone please give me detailed instructions on how to use APT utility to get the needed packages - and what are the needed packages? Thanks kindly, David David V. Fansler s/v Annabelle dfans...@dv-fansler.com www.dv-fansler.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] Setting up Conference with Moderator
Cool, I will explore that option when I have some time.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Wednesday, November 18, 2009 11:02 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Hi again UK,
IVR is designed to naturally return to previous or top menus. I don't
think there's a way to change this default behavior. Maybe its time
to move to a script-based pin validation system so you have the full
control you need.
http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
Rob
On Nov 18, 2009, at 11:34 PM, Ujjval Karihaloo wrote:
I have used the following setting in ivr.conf.xml to setup
conferencing with moderator.
However, the issue I have is - the user enters 123456 and then say
if it's a moderator they enter wrong Moderator PIN 3 times then it
takes the user back to the main menu...conference_menu and asks
for main conf pin (123456) once again.
I would like the caller to be disconnected if they get into the
Moderator menu and enter wrong Moderator PIN 3 times.
menu name=conference_menu
greet-long=welcome_please_enter_conference_pin.wav
greet-short=check_and_try_again.wav
invalid-sound=passcode_invalid.wav
exit-sound=voicemail/vm-goodbye.wav
timeout=1
inter-digit-timeout=5000
max-failures=3
max-timeouts=3
digit-len=7
entry action=menu-sub digits=123456
param=conference_123456_moderator_menu / !-- conference
moderator menu --
/menu
menu name=conference_123456_moderator_menu
greet-
long
=
conference_confirmed_enter_moderator_pin_or_1_to_join_as_participant
.wav
greet-short=check_moderator_pin_or_1_to_join.wav
invalid-sound=invalid_moderator_pin.wav
exit-sound=voicemail/vm-goodbye.wav
timeout=1
inter-digit-timeout=5000
max-failures=3
max-timeouts=3
digit-len=5
entry action=menu-exec-app digits=1234 param=conference
123...@default+flags{moderator} /
entry action=menu-exec-app digits=1 param=conference
123...@default+flags{} /
/menu
/menus
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Thursday, November 05, 2009 7:52 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Hi UK,
From what I've done and read, the caller-controls (in
conference.conf.xml) can be modified to almost anything you can think
of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a
caller control just for those with the moderator flag. So unless you
want everyone able to mute/kick everyone then you can't do it.
The wiki seems to indicate this as well:
Be aware that the caller-controls are applied across the entire
conference. You cannot enter one member of the conference using
caller-
controls ABC and then enter a second member using caller-controls
XYZ.
http://wiki.freeswitch.org/wiki/Mod_conference
I think this might be a limitation of mod_conference. Perhaps one of
the pros can chime in if I'm off-base or there's some nifty way to
accomplish this.
Cheers,
Rob
On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote:
Any ideas on the below...has anyone implemented the below:
Once I have the Moderator and Participants logged on, how do I
invoke the moderator previlidges, LIk esay muting everyone/someone
or kicking someone out of the Conf and the like?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Ujjval Karihaloo
Sent: Monday, November 02, 2009 12:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Rob:
Once I have the Moderator and Participants logged on, how do I
invoke the moderator previlidges, LIk esay muting everyone/someone
or kicking someone out of the Conf and the like?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Rob Forman
Sent: Friday, October 30, 2009 9:34 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Hm, strange. I haven't seen that before. Can you pastebin your logs
at debug level?
On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
It's strange... a tcpdump tells me that there is
[Freeswitch-users] Freeswitch Video Capture and Playback
I'm using ekiga with mod_fsv, trying to record and play back video. When I dial the record extension, it seems to record something, as the video file gets bigger. Trying then to dial the extension for play back, just hangs up, with freeswitch saying: od_fsv.c:247 File version does not match! There seems to be no information on the FSV format or the mod_fsv module on the wiki. Is this at all supposed to work?. What clients and codecs were successful?. Any pointers as to what I can try?. -- Esben Stien is b...@e s a http://www. s tn m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@n n ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] uuid_bridge kills both channels if they are executing java app
Your annoying behaviour is the exact behavior you should be getting considering what you told FS to do. As soon as you call uuid_bridge you are transferring both legs of the call to bridge to each other. This means your java app must exit so the channels can connect to each other. remember that you hangup hook can be called when the channel is transferred not only when it hangs up. you have to test which is happening based on the input to your callback. On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov shiya...@gmail.com wrote: Hi there! I've got annoying FS behavior: There are 2 channels executing the same Java application (application itself is an IVR). If I try to bridge them with uuid_bridged then both channels are killed. Here is a log from FS console: uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 (sofia/internal/1...@192.168.147.130) State Change CS_EXECUTE - CS_HIBERNATE 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.147.130 [BREAK] 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 (sofia/internal/1...@master.agent.starpoundtech.net) State Change CS_EXECUTE - CS_HIBERNATE 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de freeswi...@localhost.localdomain 2009-07-09 05:58:26.674348 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@master.agent.starpoundtec 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1...@192.168.147.130 [BREAK] 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1...@master.agent.starpoundtech.net [BREAK] (FS version is 1.0.4) Any thoughts? Artem ___ FreeSWITCH-users mailing list FreeSWITCH-users@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: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] upgrading to latest SVN
Ujjval Karihaloo ujj...@simplesignal.com wrote: Getting error below..not sure whats wrong..which line number in what file does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] APT Utility
Ubuntu has pretty good package management with apt-get and should work well for a beginner. The recommended OS (though FreeSWITCH runs on a wide variety of platforms) is 64-bit CentOS. You can get it here: http://www.centos.org/ if you'd like, but at this point I think it's fine to just keep digging into whichever flavor of linux you have handy. If you have FreeSWITCH compiled and installed, have you tried just starting it from the command line? cd /usr/local/freeswitch ; ./bin/ freeswitch On Nov 19, 2009, at 8:18 PM, David V. Fansler wrote: Thanks for your answers Rob and Shelby. I found more info on apt- get and ran it against all the missing dependences noted. I also ran through the sequence of commands Shelby suggested. In the end, running dpkg –checkbuilddeps I got the following in return dpkg-checkbuilddeps: Unnet builddependencies: debhelper (=5) then followed the instructions for Ubuntu to enable freeswitch nano /etc/default/freeswitch FREESWITCH_ENABLE=”true” And then tried invoke –rc.d freeswitch start but nothing obvious happened. I am only using Ubuntu since it came as a free DVD in the Linux Pro mag that the article about Freeswitch was in. Is there a better version of Linux to use? thanks David David V. Fansler s/v Annabelle dfans...@dv-fansler.com www.dv-fansler.com From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] On Behalf Of Rob Forman Sent: Wednesday, November 18, 2009 5:53 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] APT Utility Hi David, When using Apt, you would install packages with: apt-get install package name Or search for packages with apt-cache search search term If you're not root, you'll need to stick sudo in front of those command. Honestly, you might want to find a better tutorial with explicit command-by-command instructions. Good luck! Rob On Nov 18, 2009, at 3:49 PM, David V. Fansler wrote: Greetings – I am trying to startup a freeSwitch on a P4 running Ubuntu 9.04 “Jaunty”. I know very little about Linux. I decided to try this after reading the article in Linux Pro Magazine. I have been following the detailed instructions in the wiki for using Ubuntu Jaunty, however I have run into an unknown – “Use your favorite APT utility to get the needed packages”. I am good at following direct instructions – but this statement is too vague for my minimal minimal – did I mention minimal - knowledge of Linux. Could someone please give me detailed instructions on how to use APT utility to get the needed packages – and what are the needed packages? Thanks kindly, David David V. Fansler s/v Annabelle dfans...@dv-fansler.com www.dv-fansler.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] upgrading to latest SVN
I really didn't change anything. I was running 1.0.4 and now built from SVN...I see the oddly placed entry in ivr.conf.xml sscode_invalid.wav/configuration...removed it now its error on somewhere here: configuration name=xml_rpc.conf description=XML RPC settings !-- The port where you want to run the http service (default 8080) -- param name=http-port value=8080/ !-- if all 3 of the following params exist all http traffic will require auth -- param name=auth-realm value=freeswitch/ param name=auth-user value=freeswitch/ param name=auth-pass value=works/ /settings /configuration ERROR is: [r...@ss_freeswitch freeswitch]# freeswitch 2009-11-19 20:48:54.189120 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-19 20:48:54.190970 [DEBUG] switch_event.c:553 Create event dispatch thread 0 Cannot Initialize [[error near line 2840]: unexpected closing tag /section] -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, November 19, 2009 8:30 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN Ujjval Karihaloo ujj...@simplesignal.com wrote: Getting error below..not sure whats wrong..which line number in what file does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] upgrading to latest SVN
Does svn update try to merge the config files..Need some help, I think it has added some entries in my config files that is causing tag mismatches.. Please advise how to get back my orig config? -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval Karihaloo Sent: Thursday, November 19, 2009 8:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN I really didn't change anything. I was running 1.0.4 and now built from SVN...I see the oddly placed entry in ivr.conf.xml sscode_invalid.wav/configuration...removed it now its error on somewhere here: configuration name=xml_rpc.conf description=XML RPC settings !-- The port where you want to run the http service (default 8080) -- param name=http-port value=8080/ !-- if all 3 of the following params exist all http traffic will require auth -- param name=auth-realm value=freeswitch/ param name=auth-user value=freeswitch/ param name=auth-pass value=works/ /settings /configuration ERROR is: [r...@ss_freeswitch freeswitch]# freeswitch 2009-11-19 20:48:54.189120 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-19 20:48:54.190970 [DEBUG] switch_event.c:553 Create event dispatch thread 0 Cannot Initialize [[error near line 2840]: unexpected closing tag /section] -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, November 19, 2009 8:30 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN Ujjval Karihaloo ujj...@simplesignal.com wrote: Getting error below..not sure whats wrong..which line number in what file does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] upgrading to latest SVN
Nothing will replace your configs automatically in the whole build system. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 19-Nov-09, at 9:22 PM, Ujjval Karihaloo wrote: Does svn update try to merge the config files..Need some help, I think it has added some entries in my config files that is causing tag mismatches.. Please advise how to get back my orig config? -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] On Behalf Of Ujjval Karihaloo Sent: Thursday, November 19, 2009 8:49 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN I really didn't change anything. I was running 1.0.4 and now built from SVN...I see the oddly placed entry in ivr.conf.xml sscode_invalid.wav/configuration...removed it now its error on somewhere here: configuration name=xml_rpc.conf description=XML RPC settings !-- The port where you want to run the http service (default 8080) -- param name=http-port value=8080/ !-- if all 3 of the following params exist all http traffic will require auth -- param name=auth-realm value=freeswitch/ param name=auth-user value=freeswitch/ param name=auth-pass value=works/ /settings /configuration ERROR is: [r...@ss_freeswitch freeswitch]# freeswitch 2009-11-19 20:48:54.189120 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-19 20:48:54.190970 [DEBUG] switch_event.c:553 Create event dispatch thread 0 Cannot Initialize [[error near line 2840]: unexpected closing tag / section] -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] On Behalf Of Jason White Sent: Thursday, November 19, 2009 8:30 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] upgrading to latest SVN Ujjval Karihaloo ujj...@simplesignal.com wrote: Getting error below..not sure whats wrong..which line number in what file does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] How do I know the destination profile name?
check out sofia_contact function. If you use this in combination with binding
profiles together so they are one table I think this should work right.
Mike
On Nov 18, 2009, at 12:36 AM, Eli Hayun wrote:
Brian West wrote:
Why do you need to know the destination profile like that? You get to
pick that on your own so you should already know that before hand.
/b
On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote:
Hi
We have more then one profile. To make a call I have to enter : bridge
sofia/profile/num...@ip
The problem is when I use : ${use_profile} I am getting the caller
profile, and I need the destination profile.
How do I get this information?
Thanks for your answer.
The problem is when I call to that number that the phone hook to other
server, I cannot make the call.
Is there is a variable that can tell me the destination profile?
Lets say the other profile called ph1 I have to dial
sofia/ph1/xx...@host to make the call. Is there other way to do that?
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Re: [Freeswitch-users] upgrading to latest SVN
Ujjval Karihaloo ujj...@simplesignal.com wrote: Does svn update try to merge the config files..Need some help, I think it has added some entries in my config files that is causing tag mismatches.. Building and installing FreeSWITCH won't interfere with your configuration. I would suggest using Git or another version control system to keep track of configuration files. I prefer Git. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Another Group Question - on VoiceMail
On Thu, Nov 19, Joseph L. Casale wrote: Is there any way that each extension in the group can be notified that a group Voice Mail is waiting to be picked up so that each phone shows the message waiting indication ? Wouldn't this be simply accomplished by setting the vicemail as box 100 for each of the users (such as ext 1011xx)? Check out the directory parameter MWI-Account. Along with setting the mailbox variable. The variable mailbox sets what voicemail box a person dialing the extension will be dropped in to and MWI_Account, the param, will tell the phone what mailbox to subscribe to. HTH --FC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] RTP issues (possibly nat-related)
Brian West br...@freeswitch.org wrote: I think the fix for this is coming to an SVN repo near you... so give it a few and update. Thanks Brian! I'll watch the svn logs and update when the fix lands. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] RTP issues (possibly nat-related)
Update and see if the problem is gone. /b On Nov 20, 2009, at 1:01 AM, Jason White wrote: Thanks Brian! I'll watch the svn logs and update when the fix lands. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
