Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec.
should open a jira on this? Did you guys need any more info? - original message - Subject:Re: [Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec. From: Brian West br...@freeswitch.org Date: 05/06/2009 00:35 Try SVN trunk I cna tell you're using older code! ;) /b On Jun 4, 2009, at 7:23 PM, Jim Burke wrote: Hey Brian From your comments above this appears to be the code that does the damage. I guess now the question is why?? soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called soa_static(0x9c1d4e8, soa_generate_answer): generating local description soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote description soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media Regards, Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] WikiPBX Installation
On Jun 5, 2009, at 1:36 PM, Michael Collins wrote: On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood matthew.lockw...@gmail.com wrote: That would involve me learning a totally new framework. It'll not the hardest code I'll ever write by far, so I'm okay coding it up on my own. However, I definitely need a lot of help from fabulous designers to actually make the interface pretty and useable. Plus, I'm only one person and will need a lot of feedback to create something that rocks - everybody has a different use case and I can't foresee how everybody will use it, so that kind of feedback will go into re-engineering it. If you guys are serious about this then I would like to make a few suggestions that might be obvious but for the sake of the project we'll make them explicitly obvious. First, before deciding what framework to use, it would be good to hold some discussions about what the GUI actually needs to do: What are the design goals? Will it be just for setting up extensions and the dialplan? Or will it go much farther than that? Will you be using mod_xml_curl for everything? If so, what database(s) will you support? Are you going to have extra goodies like an IVR builder? A 'visual voicemail' page? A user portal? Management interface to 'spy' on users? A CDR/call accounting system? FIFO and/or ACD queue management? MOH and sound files management? and conference management It's okay to start small and build your way out, but you need to know before you start building what the grand scheme will be. The larger the goals of the project, the narrower your choices for a framework that can do it all. The simple fact of the matter is that if you want to use a MVC web framework then you have a somewhat limited number of choices. You need a MVC WF that fits your needs, which means it needs to be at least somewhat flexible. If you want a pretty GUI then you need to decide if you want a rich Internet application (RIA) front end like AIR, or do you want something along the lines of XHTML/CSS/JS and use a platform like Dojo which gives you cross-browser widgets and tools. All of this on top of the fact that if you want volunteers to assist you will need to pick something that people either know or can learn quickly. Oh, and be prepared for people to give you unsolicited opinions about all sorts of things. :) All that being said, I say go for it. Find what works for you and see what happens. Be sure to use #freeswitch-gui. If this really takes off I'm sure that we could even start a mailing list for GUI development. Once the goals and features decided I think more ppl can join and work this out together. Enjoy! -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] Testing Freeswitch performance led to strange behavior
Anthony Minessale wrote: FS uses async rtp timers so you may want to set rtp-timer-name=none in the profile param to simulate asterisk conditions. I tried that - although I am not using rtp in my scenario - with the same results. Also keep in mind that asterisk as an atvantage in a tiny crappy 32 bit single cpu box because that was what was popular when it was designed and the chance for race conditions is minimal because there is only 1 cpu. As you scale up to a 8 core 64 bit xeon you will set a drastic difference. Yes I know that this machine is not well suited for today's test needs. But the issue occurs in every machine as long as it is pushed near (but not quite near) to its limits. I have the same odd durations using a 64 bit low end server. In this case I could achieve a better call/sec rate than that of the crappy PC but around 50-60 calls/sec the same problem showed up. I also used a Mosso Cloud Server (quad core - 64-bit) and the same thing happened at a higher rate. I will be happy to investigate this issue a bit if you'd like but i do not have any box like you describe so if I can't find anything you may have to lend us your lab. I would appreciate it if you did. After all there this might be a problem that has not surfaced yet but someday will as more and more production boxes start using FS. So it would be better to investigate it now. I don't think lending you access to my old P4 PC would help you very much :) If you have access to a normal 2-4 core system you can easily start raising the sipp parameters until it starts happening. However if you really think it is appropriate to use my test machines I'd be happy to grant access to our low-end Opteron machine (just send me a personal email). I cannot grant you access to larger systems because they are used in production. I used the embedded sipp scenarios : on the UAS side : sipp -i UAS_IP -mi UAS_IP -ci UAS_IP -mp 8000 -sn uas on the UAC side : sipp FS_IP:5060 -s 44050505-i UAC_IP -mi UAC_IP -ci UAC_IP -r 70 -d 5000 -l 500 -m 2000 -sn uac The dialplan : ?xml version=1.0 encoding=utf-8? !-- http://wiki.freeswitch.org/wiki/Dialplan_XML -- include context name=mydialplan extension name=dial1 condition field=destination_number expression=(^.*)$ !-- Dial Back -- action application=set data=absolute_codec_string=PCMU/ !-- action application=set data=proxy_media=true/ -- action application=bridge data=sofia/gateway/sipp01/$1/ /condition /extension /context /include If you need anything else from the config just notify me. In order to verify that at some point the calls start having a duration larger than the scenario's 5secs you can tcpdump on the sipp machine or turn on the cdrs logging (I know that it degrades performance, but as I said it is not a matter of when exactly it starts happening, it is a matter that it DOES start happening). On Thu, Jun 4, 2009 at 12:47 PM, r...@kinetix.gr mailto:r...@kinetix.gr r...@kinetix.gr mailto:r...@kinetix.gr wrote: Michael Collins wrote: The dialplan : ?xml version=1.0 encoding=utf-8? !-- http://wiki.freeswitch.org/wiki/Dialplan_XML -- include context name=mydialplan extension name=dial1 condition field=destination_number expression=^.*$ You forgot the parens around .* It should be expression=^(.*)$ if you plan to use $1 later in the extension... !-- Dial Back -- action application=set data=absolute_codec_string=PCMA/ action application=bridge data=sofia/gateway/sipp01/$1/ ... like here ^^^ :) -MC You are right! Although, I don't think that would change the outcome of my test :) /condition /extension /context /include ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org
Re: [Freeswitch-users] WikiPBX Installation
hello wasim, GWT will help me about Accessibility, is very very accessible tel me about how i can host GWT application thanks Wasim Baig wrote: Just to chime in, perhaps GWT might be a good framework ... -wasim ___ Freeswitch-users mailing list Freeswitch-users@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] WikiPBX Installation
Thanks for all this. What I meant to respond with before I passed out asleep was: The framework doesn't matter that much. We've all got looped on this issue. Everything that is done has to add value to the end user. The framework is far down at the bottom of the list of things that provides value, but it's not something to be ignored. The vision I have for this is something that's so simple it lowers the barriers that would otherwise stop people from using FreeSwitch. Using some relatively unheard of framework is going to most certainly complicate things. Simple = good. And plus, on a side note if we throw out a whole bunch of frameworks and acronyms and make a big deal about the actual technology that powers the GUI (not that people even care most of the time), people will start to get more confused and it'll backfire. I'll be using .NET/Mono unless I can come up with an exceptionally good reason to use something else. I'm choosing this framework over everything else because it's what I know best and I'll be writing the code base. I've got years of experience writing code in C# and developing .NET web applications so it makes more sense than learning something new that will slow the development time and result in me producing poorer code. This isn't me being mercenary, but the GUI isn't likely to cross the million codeline barrier (even with everything implemented) and this is a framework I have a lot of experience with. I'm totally fine being the lone developer for now, and there are a lot of people with similar programming skillsets as mine so it's not like there will never be anybody else that'll ever contribute code. Personally, I think it's more important to have well written code that is rapidly developed than it is to have a shiny technology that adds no value. :-) Of course, the final product will be perfectly standards compliant and 100% accessible. I know this is important. I'm going to lay the framework issue to rest now. It'll be .NET/Mono unless there is some super-compelling reason to use something else. If for some reason there is such a reason not to use .NET/Mono, the second choice is PHP. The other thing is I'm pretty much going to develop upwards of 95% of the features in one go. Nobody wants an incomplete product that lacks necessary functionality, so from v1.0 it'll be pretty much feature complete. I'm developing this for use in my business, so I need it feature complete, and that's what the community will get too - a feature complete product. Hope you're happy having a fully fledged GUI! ;-) M On Fri, Jun 5, 2009 at 1:26 AM, seven dujinf...@gmail.com wrote: On Jun 5, 2009, at 1:36 PM, Michael Collins wrote: On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood matthew.lockw...@gmail.com wrote: That would involve me learning a totally new framework. It'll not the hardest code I'll ever write by far, so I'm okay coding it up on my own. However, I definitely need a lot of help from fabulous designers to actually make the interface pretty and useable. Plus, I'm only one person and will need a lot of feedback to create something that rocks - everybody has a different use case and I can't foresee how everybody will use it, so that kind of feedback will go into re-engineering it. If you guys are serious about this then I would like to make a few suggestions that might be obvious but for the sake of the project we'll make them explicitly obvious. First, before deciding what framework to use, it would be good to hold some discussions about what the GUI actually needs to do: What are the design goals? Will it be just for setting up extensions and the dialplan? Or will it go much farther than that? Will you be using mod_xml_curl for everything? If so, what database(s) will you support? Are you going to have extra goodies like an IVR builder? A 'visual voicemail' page? A user portal? Management interface to 'spy' on users? A CDR/call accounting system? FIFO and/or ACD queue management? MOH and sound files management? and conference management It's okay to start small and build your way out, but you need to know before you start building what the grand scheme will be. The larger the goals of the project, the narrower your choices for a framework that can do it all. The simple fact of the matter is that if you want to use a MVC web framework then you have a somewhat limited number of choices. You need a MVC WF that fits your needs, which means it needs to be at least somewhat flexible. If you want a pretty GUI then you need to decide if you want a rich Internet application (RIA) front end like AIR, or do you want something along the lines of XHTML/CSS/JS and use a platform like Dojo which gives you cross-browser widgets and tools. All of this on top of the fact that if you want volunteers to assist you will need to pick something that people either know or can learn quickly. Oh, and be prepared for people to give you unsolicited
Re: [Freeswitch-users] WikiPBX Installation
This is a really ironic post, Seven. :-) I agree with all your points. A while ago I started the TCAPI project to build a front-end for FreeSWITCH. I very quickly got inundated with debates about framework and language. These debates were initially appreciated but at some point we needed to decide move on. The real work to be done was, as you point out, in design of the application business logic, interface and actually coding it up and putting it together. So we decided to go a bit radio silent and and focus on a few developers who were willing to build out the foundational pieces of the MVC architecture, and to let you create FreeSWITCH config files and general database and software modules with a set of standardized, simple to use libraries/APIs. Once we are done with that, the intention was to release it to those who wanted to help build the pieces related to modules in FreeSWITCH. That project is about 6 weeks from release into beta, give or take a few weeks (hey, it's software dev! heh who's ever on time?). So anyone who is on here reading this and might be interested in contributing code to an already very active FreeSWITCH GUI development project please feel free to contact me - we are now accepting serious developer inquiries. The project is in PHP and uses two pretty nifty frameworks (we, as you point out, couldn't find exactly what we were looking for, so we merged two libraries that fit the bill very nicely). It is database agnostic and is designed to work on Windows or Linux so don't let that be a barrier to participation. This will be an open source project for all, btw. I will be presenting on it at the upcoming ClueCon, warts and all, so you should go register and then you can participate in the demo/tutorial! :-) - Darren _ From: seven [mailto:dujinf...@gmail.com] Sent: Friday, June 05, 2009 1:26 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] WikiPBX Installation On Jun 5, 2009, at 1:36 PM, Michael Collins wrote: On Thu, Jun 4, 2009 at 6:52 PM, Matthew Lockwood matthew.lockw...@gmail.com wrote: That would involve me learning a totally new framework. It'll not the hardest code I'll ever write by far, so I'm okay coding it up on my own. However, I definitely need a lot of help from fabulous designers to actually make the interface pretty and useable. Plus, I'm only one person and will need a lot of feedback to create something that rocks - everybody has a different use case and I can't foresee how everybody will use it, so that kind of feedback will go into re-engineering it. If you guys are serious about this then I would like to make a few suggestions that might be obvious but for the sake of the project we'll make them explicitly obvious. First, before deciding what framework to use, it would be good to hold some discussions about what the GUI actually needs to do: What are the design goals? Will it be just for setting up extensions and the dialplan? Or will it go much farther than that? Will you be using mod_xml_curl for everything? If so, what database(s) will you support? Are you going to have extra goodies like an IVR builder? A 'visual voicemail' page? A user portal? Management interface to 'spy' on users? A CDR/call accounting system? FIFO and/or ACD queue management? MOH and sound files management? and conference management It's okay to start small and build your way out, but you need to know before you start building what the grand scheme will be. The larger the goals of the project, the narrower your choices for a framework that can do it all. The simple fact of the matter is that if you want to use a MVC web framework then you have a somewhat limited number of choices. You need a MVC WF that fits your needs, which means it needs to be at least somewhat flexible. If you want a pretty GUI then you need to decide if you want a rich Internet application (RIA) front end like AIR, or do you want something along the lines of XHTML/CSS/JS and use a platform like Dojo which gives you cross-browser widgets and tools. All of this on top of the fact that if you want volunteers to assist you will need to pick something that people either know or can learn quickly. Oh, and be prepared for people to give you unsolicited opinions about all sorts of things. :) All that being said, I say go for it. Find what works for you and see what happens. Be sure to use #freeswitch-gui. If this really takes off I'm sure that we could even start a mailing list for GUI development. Once the goals and features decided I think more ppl can join and work this out together. Enjoy! -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] Missing lines copying data from console to vi
Thanks, Jim, that advice really helped in more ways than I asked for. Lars -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jim Burke Sent: Thursday, June 04, 2009 6:52 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Missing lines copying data from console to vi After logging into my linux box using putty. I then change directory the the ~/freeswitch/bin directory and run ./fs_cli You can, but you don't need to start FS from the putty terminal. We run ours as a background process. On Fri, Jun 5, 2009 at 11:38 AM, Lars Zeb larc...@yahoo.com wrote: Does this mean that I must start fs from the putty terminal, or can I attach to an already running instance via putty? -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jim Burke Sent: Thursday, June 04, 2009 4:19 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Missing lines copying data from console to vi Not that this helps your question directly. Using a putty terminal from windows allows the data to be copied correctly from console sessions. On Fri, Jun 5, 2009 at 2:59 AM, Lars Zeb larc...@yahoo.com wrote: I want to copy the results of a siptrace captured on the fs console to a file. The console is running on a Gnome terminal. I highlight the text I want to copy in the fs console, open a vi session in insert mode, and paste the text. However, the text is not pasted as I copied it - it is missing characters/lines. I know I am doing something wrong. Is there another way to save siptraces to a file? Redirection doesn't work. sofia profile internal siptrace on is the command I use. Thanks Lars ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net ___ Freeswitch-users mailing list Freeswitch-users@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 -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net ___ Freeswitch-users mailing list Freeswitch-users@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] Freeswitch creating more then two sessions for one call
Hi FS-Users, I am having an strange problem of sessions in freeswitch. For every call Freeswitch is creating more than two sessions (in and out legs). For example, I have seen 260+ sessions for just 30+ calls whereas there should not be more than 60 sessions for just 30 calls. I have seen the same problem with 1.0.4pre8 and trunk version. I am using default values from switch.conf.xml for max_sessions (1000) and sessions_per_second(30). Freeswitch create many sessions only when there is good load on the system. It works fine with the steady load of 50-100 calls. However, if I give it 200+ calls at once then it breaks. Any suggestion? Shoaib ___ Freeswitch-users mailing list Freeswitch-users@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] Calls drop immediately when terminator forc es G.729 Codec.
DId you update to trunk? /b On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: should open a jira on this? Did you guys need any more info? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] GnuGK vs FreeSWITCH
Hi, I'm using GnuGK H323 gatekeeper. It has good performance and many features, but doesn't supports SIP. I see many Asterisk vs FreeSwitch comparisons, but how about GnuGK vs FreeSwitch? The setup for which I'm asking is in a range of 200 concurrent calls. The points that I'm really interested for comparison are: 1) Proxy of RTP feature and it's stability 2) NAT support 3) Direct SQL AAA support (without the need of using RADIUS server) 4) Performance as an endpoint registrar 5) Rerouting to a second carrier on failed call Also, is the H323 library of FreeSWITCH based on h323plus/openh323? Marcus ___ Freeswitch-users mailing list Freeswitch-users@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] Reuse ODBC connection in javascript
Hi, I want to reuse database connection (mysql) in one of my javascript which is executed on each call. Following is how I am creating ODBC connection. Line1) var db = new ODBC(DSN, DB_USER, DB_PASS); Line2) db.connect(); My first question is, where does it create a database connection on line1 or on line2? Secondly, how can I reuse this connection? so that it is not created for each call and I just use a previously created codedb/code object in my routing script. My Objective is: 1) Create ODBC connection on freeswitch startup (could be a database connection pool) 2) Reuse the connection on each call 3) Close database connection on freeswitch shutdown (or on some other event) Any help would be highly appreciated. Shoaib ___ Freeswitch-users mailing list Freeswitch-users@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] Using setGlobalVar and getGlobalVar
On Jun 5, 2009, at 4:10 PM, Shoaib Khanzada wrote: Hi, How can I use setGlobalVar and getGlobalVar in my javascript to store a ODBC connection? I want to set an ODBC database connection object globally so that I can access it from anywhere. This connection will be used for read- only so no threading issues. no, those are for strings only. Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] GnuGK vs FreeSWITCH
Can you explain your friendly advice? Some parts can be compared On Fri, Jun 5, 2009 at 10:11 PM, EdPimentledpime...@gmail.com wrote: Is this a joke or do want to save the question for April Fools? -E On Fri, Jun 5, 2009 at 2:14 PM, Marcus Frenkel marcus.fren...@gmail.com wrote: Hi, I'm using GnuGK H323 gatekeeper. It has good performance and many features, but doesn't supports SIP. I see many Asterisk vs FreeSwitch comparisons, but how about GnuGK vs FreeSwitch? The setup for which I'm asking is in a range of 200 concurrent calls. The points that I'm really interested for comparison are: 1) Proxy of RTP feature and it's stability 2) NAT support 3) Direct SQL AAA support (without the need of using RADIUS server) 4) Performance as an endpoint registrar 5) Rerouting to a second carrier on failed call Also, is the H323 library of FreeSWITCH based on h323plus/openh323? Marcus ___ Freeswitch-users mailing list Freeswitch-users@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] Calls drop immediately when terminator forc es G.729 Codec.
Yup sure did, same result :( - original message - Subject:Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. From: Brian West br...@freeswitch.org Date: 05/06/2009 14:10 DId you update to trunk? /b On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: should open a jira on this? Did you guys need any more info? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Calls drop immediately when terminator forc es G.729 Codec.
it is using G729a which is not the correct RFC value that goes with payload 18 so it's moving it to 96 as if it's some other codec. the only thing you can do is get them to stop using invalid data in their sdp or hack it to replace G729a with G729 before it's too late. On Fri, Jun 5, 2009 at 6:25 PM, Jim Burke j...@evolutiontel.net wrote: Yup sure did, same result :( - original message - Subject:Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. From: Brian West br...@freeswitch.org Date: 05/06/2009 14:10 DId you update to trunk? /b On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: should open a jira on this? Did you guys need any more info? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Freeswitch creating more then two sessions for one call
Could it be that you are getting several INVITE messages per call due to 100 Trying message is not being sent until after the route is selected from your java routing script? You might be able to send a 100 trying from your dialplan or script. Keep in mind that this message stops timers on the originating side. - original message - Subject:Re: [Freeswitch-users] Freeswitch creating more then two sessions for one call From: Michael Jerris m...@jerris.com Date: 05/06/2009 20:27 On Jun 5, 2009, at 4:11 PM, Shoaib Khanzada wrote: Thanks for the reply…. I was not running it for testing purpose because I’ve completed all testing successfully. We are running freeswitch in a voip carrier grade environment. It works perfectly alright in off peak hours when we have low calls ratio around 100-150 concurrent calls. However, in peak hours this goes beyond 200 calls and on that point freeswitch start creating many sessions for each calls. I have seen the sessions using “status” command and calls using “show calls count”. We are using javascript to select the route from the mysql database for each call. Could it be because script is taking longer than expected amount of time to retrieve a route? and freeswitch continuously keep creating sessions for incoming calls. That’s why I see low no of connected calls (if ‘show calls count’ only display the connected calls) whereas sessions are continuously being created by freeswitch as it is receiving many calls. If above text confuses you, nevermind just answer the following questions. 1) Does ‘show calls count’ display the connected calls only? Only bridged calls (2 sessions) 2) When freeswitch create session instances? Before bridge or after bridge? Or one before bridge and one after bridge? It creates a session when it gets an incomming call and creates one for each outgoing call, unrelated to bridging. Thanks, Shoaib How are you doing the bridge in your script? Are you setting a var then dropping out of the js to do the bridge? Can you post your js file? Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@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] Reducing record_session load
we experience some latency in the recording files even with PCMU-PCMU session to a stereo WAV file. i want to reduce the CPU load hoping to reduce this problem. would it help if do the ff? 1. save it in PCMU file. i can use sox at the end of the shift. 2. record in mono. does it help? 3. will record_session work w/ proxy_media=true? tks for your help. -nandy ___ Freeswitch-users mailing list Freeswitch-users@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] Calls drop immediately when terminator forc es G.729 Codec.
Yes, the terminator is Linksys so will change it and test. Noticed there is a list of mime types associated with FS and G729a was not listed, does this have anything to do with the root cause? - original message - Subject:Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. From: Mathieu Rene mrene_li...@avgs.ca Date: 05/06/2009 23:56 Linksys still uses G729a in their sdp, but you can change it in the admin panel (if thats what you have) Math On 5-Jun-09, at 7:53 PM, Anthony Minessale wrote: it is using G729a which is not the correct RFC value that goes with payload 18 so it's moving it to 96 as if it's some other codec. the only thing you can do is get them to stop using invalid data in their sdp or hack it to replace G729a with G729 before it's too late. On Fri, Jun 5, 2009 at 6:25 PM, Jim Burke j...@evolutiontel.net wrote: Yup sure did, same result :( - original message - Subject:Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. From: Brian West br...@freeswitch.org Date: 05/06/2009 14:10 DId you update to trunk? /b On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: should open a jira on this? Did you guys need any more info? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn: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 ___ Freeswitch-users mailing list Freeswitch-users@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] Reducing record_session load
On Jun 5, 2009, at 9:24 PM, Nandy Dagondon wrote: we experience some latency in the recording files even with PCMU- PCMU session to a stereo WAV file. i want to reduce the CPU load hoping to reduce this problem. would it help if do the ff? 1. save it in PCMU file. i can use sox at the end of the shift. You shouldn't be experiencing this at all... how many are you doing at once? 2. record in mono. does it help? No. 3. will record_session work w/ proxy_media=true? Nope. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec.
G729a is 100% INVALID in the sdp on codec 18. /b On Jun 5, 2009, at 9:25 PM, Jim Burke wrote: Noticed there is a list of mime types associated with FS and G729a was not listed, does this have anything to do with the root cause? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Reducing record_session load
You shouldn't be having problems... what version are you using? /b On Jun 5, 2009, at 9:58 PM, Nandy Dagondon wrote: there 10 client seats so at max. 10 simultaneous calls. however, the number of clients may be increased. -nandy Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Freeswitch creating more then two sessions for one call
We already send a 100 before the call even hits the dialplan. Mike On Jun 5, 2009, at 8:00 PM, Jim Burke j...@evolutiontel.net wrote: Could it be that you are getting several INVITE messages per call due to 100 Trying message is not being sent until after the route is selected from your java routing script? You might be able to send a 100 trying from your dialplan or script. Keep in mind that this message stops timers on the originating side. - original message - Subject:Re: [Freeswitch-users] Freeswitch creating more then two sessions for one call From:Michael Jerris m...@jerris.com Date:05/06/2009 20:27 On Jun 5, 2009, at 4:11 PM, Shoaib Khanzada wrote: Thanks for the reply…. I was not running it for testing purpose because I’ve completed all testing successfully. We are running freeswitch in a voip carrier grade environment. It works perfectly alright in off peak hours when we have low calls ratio around 100-150 concurrent calls. However, in peak hours this goes beyond 200 calls and on that point freeswitch start creating many sessions for each calls. I have seen the sessions using “status” command and calls using “show calls count”. We are using javascript to select the route from the mysql database for each call. Could it be because script is taking longer than expected amount of time to retrieve a route? and freeswitch continuously keep creating sessions for incoming calls. That’s why I see low no of connected calls (if ‘show calls count’ only display the connected calls) whereas sessions are continuously being created by freeswitch as it is receiving many calls. If above text confuses you, nevermind just answer the following questions. 1) Does ‘show calls count’ display the connected calls only? Only bridged calls (2 sessions) 2) When freeswitch create session instances? Before bridge or after bridge? Or one before bridge and one after bridge? It creates a session when it gets an incomming call and creates one for each outgoing call, unrelated to bridging. Thanks, Shoaib How are you doing the bridge in your script? Are you setting a var then dropping out of the js to do the bridge? Can you post your js file? Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@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] How to reject a call without answering
Hi, Going through the socket api, how can i reject a call without having to answer it first? I tried sending a hangup command with cause set either to NO_ANSWER or NORMAL_CLEARING. In both cases, Freeswitch does create another socket to deliver the very same call. More precisely, when a call comes in i send a connect command. Then after some few seconds, i then send the following hangup command: SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f call-command: hangup hangup-cause: NO_ANSWER Thanks for any feedback. Klaus. -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 ___ Freeswitch-users mailing list Freeswitch-users@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 reject a call without answering
Try the respond app. /b On Jun 5, 2009, at 10:34 PM, Klaus Teller wrote: Hi, Going through the socket api, how can i reject a call without having to answer it first? I tried sending a hangup command with cause set either to NO_ANSWER or NORMAL_CLEARING. In both cases, Freeswitch does create another socket to deliver the very same call. More precisely, when a call comes in i send a connect command. Then after some few seconds, i then send the following hangup command: SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f call-command: hangup hangup-cause: NO_ANSWER Thanks for any feedback. Klaus. -- Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to reject a call without answering
It doesn't seem to work. I tried the following: api respond 9015430e-82cf-418c-bf4c-f3ac6e85caf2 503 SendMsg 9015430e-82cf-418c-bf4c-f3ac6e85caf2 call-command: execute execute-app-name: respond execute-app-arg: 503 Is one of these what you meant? Klaus. Original-Nachricht Datum: Fri, 5 Jun 2009 22:45:29 -0500 Von: Brian West br...@freeswitch.org An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] How to reject a call without answering Try the respond app. /b On Jun 5, 2009, at 10:34 PM, Klaus Teller wrote: Hi, Going through the socket api, how can i reject a call without having to answer it first? I tried sending a hangup command with cause set either to NO_ANSWER or NORMAL_CLEARING. In both cases, Freeswitch does create another socket to deliver the very same call. More precisely, when a call comes in i send a connect command. Then after some few seconds, i then send the following hangup command: SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f call-command: hangup hangup-cause: NO_ANSWER Thanks for any feedback. Klaus. -- Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -- GMX FreeDSL mit DSL 6.000 Flatrate und Telefonanschluss nur 17,95 Euro/mtl.! http://dslspecial.gmx.de/freedsl-aktionspreis/?ac=OM.AD.PD003K11308T4569a ___ Freeswitch-users mailing list Freeswitch-users@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] (no subject)
Ttrfrtttgteruoywtklou Regards,juuyuuu Mitul Limbani, Founder CEO, iuokljkknnvvfcxzasqwwhjhyljljjifkkkljjyjjjkkjllgjjggllyjkljkokjkjjjujkmktdswwdsflyjhhbhh mlkkkjjjhhhjykvytyyp Enterux Solutions Pvt Ltd,bu. B. P The Enterprise Linux Company(r), http://www.enterux.com/i Pio ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
