From callum.guy at x-on.co.uk Tue Nov 1 12:11:46 2011 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 1 Nov 2011 09:11:46 +0000 Subject: [Freeswitch-dev] joining LegA and LegB and to have the UUID_LegA and UUID_LegB accessible In-Reply-To: <75208aef16a616.4eae7bb6@videotron.ca> References: <75208aef16a616.4eae7bb6@videotron.ca> Message-ID: Hi Victor, Can you please confirm what action you were hoping to send to the bridged calls? An example of the command that you are having issues with would be great. Thanks, Callum ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 31 October 2011 14:43, wrote: > Hello, > > I tried to join two legs (A,B) using the "api uuid_bridge " + legA_uuid + > " " + legB_uuid in ESL program. It seems to work, but I could not have > access to legA nor to legB anymore after the bridge. > > Q: Is there a way to join the two legs that allows me to have access to > the two legs after the bridge? > > Thanks in advance, > > Victor > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111101/1450df73/attachment.html From peter at enovatiq.com Tue Nov 1 23:27:10 2011 From: peter at enovatiq.com (P. Broennimann) Date: Tue, 1 Nov 2011 21:27:10 +0100 Subject: [Freeswitch-dev] Newbie Q Message-ID: Hi there I am new to FreeSwitch and I have a question... I'd like to record a call/session outputing same length .wav files (all .wav files to be exactly 30 seconds long). I browsed through the FreeSwitch sources from GIT and I found the module 'mod_snapshot'. Would this module be the best start/base for the functionality I am looking for resp. are there other modules/scripts that I should better check first? -> I tried to build the module in MSVC Express 2010 but I get errors -> Anyone was recently successful building it? Thx in advance, P. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111101/ac35800f/attachment.html From msc at freeswitch.org Wed Nov 2 01:07:36 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Nov 2011 15:07:36 -0700 Subject: [Freeswitch-dev] Newbie Q In-Reply-To: References: Message-ID: Welcome to FreeSWITCH! First off, let me just mention that this question is probably better suited to our -users list. (This is the -dev list which has a smaller audience and is reserved for more intense discussions.) If you don't get a satisfactory answer here then re-post on freeswitch-users list. Now, as far as your question goes... you have different options available to you but it really depends on how you are generating your calls. Also, do you want the call to last more than 30 seconds and have only the recording be 30 seconds? Do you want recording to start on early media or on answer? As an example of how to do a simple 30-second recording here's one that when you call it, it will start recording, play music for 30 sec, then end the recording and disconnect: Dial 732664 (i.e. "REC MOH" :) and just wait. You'll hear music and then at 30 seconds it will stop recording. Happy FreeSWITCHing! -MC On Tue, Nov 1, 2011 at 1:27 PM, P. Broennimann wrote: > Hi there > > I am new to FreeSwitch and I have a question... > > I'd like to record a call/session outputing same length .wav files (all > .wav files to be exactly 30 seconds long). > > I browsed through the FreeSwitch sources from GIT and I found the module > 'mod_snapshot'. Would this module be the best start/base for the > functionality I am looking for resp. are there other modules/scripts that I > should better check first? > > -> I tried to build the module in MSVC Express 2010 but I get errors -> > Anyone was recently successful building it? > > Thx in advance, > P. > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111101/e50daeb4/attachment.html From peter at enovatiq.com Wed Nov 2 01:50:15 2011 From: peter at enovatiq.com (P. Broennimann) Date: Tue, 1 Nov 2011 23:50:15 +0100 Subject: [Freeswitch-dev] Newbie Q In-Reply-To: References: Message-ID: Hi Michael Thx for answering. Sorry next time I'll post on the other list... Basically I am looking for a solution to split "on-the-fly" recordings on answer. The result for each call should be a consecutive/sequential set of wav files of each X seconds (saved on-the-fly in a specific directory and using a specific filename stucture)... I don't want to reinvent the wheel so that's why I asked if something similar already exists. Performance is quiet important here (small latency). P.S: I bought the FreeSwitch book and I am making progress learning :) Thx & cheers, P. 2011/11/1 Michael Collins > Welcome to FreeSWITCH! > > First off, let me just mention that this question is probably better > suited to our -users list. (This is the -dev list which has a smaller > audience and is reserved for more intense discussions.) If you don't get a > satisfactory answer here then re-post on freeswitch-users list. > > Now, as far as your question goes... you have different options available > to you but it really depends on how you are generating your calls. Also, do > you want the call to last more than 30 seconds and have only the recording > be 30 seconds? Do you want recording to start on early media or on answer? > > As an example of how to do a simple 30-second recording here's one that > when you call it, it will start recording, play music for 30 sec, then end > the recording and disconnect: > > > > > data="$${recordings_dir}/${uuid}.wav"/> > > > > > > Dial 732664 (i.e. "REC MOH" :) and just wait. You'll hear music and then > at 30 seconds it will stop recording. > > Happy FreeSWITCHing! > > -MC > > On Tue, Nov 1, 2011 at 1:27 PM, P. Broennimann wrote: > >> Hi there >> >> I am new to FreeSwitch and I have a question... >> >> I'd like to record a call/session outputing same length .wav files (all >> .wav files to be exactly 30 seconds long). >> >> I browsed through the FreeSwitch sources from GIT and I found the module >> 'mod_snapshot'. Would this module be the best start/base for the >> functionality I am looking for resp. are there other modules/scripts that I >> should better check first? >> >> -> I tried to build the module in MSVC Express 2010 but I get errors -> >> Anyone was recently successful building it? >> >> Thx in advance, >> P. >> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111101/a6dea8f6/attachment-0001.html From egable+freeswitch at gmail.com Wed Nov 2 03:48:49 2011 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Tue, 1 Nov 2011 20:48:49 -0400 Subject: [Freeswitch-dev] Newbie Q In-Reply-To: References: Message-ID: Maybe just record the entire call and then split it up afterwards using a wav-file post-processing program. Maybe sox or ffmpeg or something similar might work? Alternatively, you could use the API in FreeSWITCH to start and stop and start and stop the recording every 30 seconds, but that seems like overkill and lots of extra work compared to just using post-processing software. Plus, you'll miss bits of the recording between when you stop one segment and start the next. Is there some reason you need to be doing this on the fly versus post-processing? Alternatively, you could write a module in C for FreeSWITCH which taps the audio and you could buffer it internally in your module and then save out 30-second segments. This would let you get all of the audio with micro-second precision splits so you wouldn't loose any audio between segments, but it's an awful lot of work to go through compared to post-processing. On Tue, Nov 1, 2011 at 6:50 PM, P. Broennimann wrote: > Hi Michael > > Thx for answering. Sorry next time I'll post on the other list... > > Basically I am looking for a solution to split "on-the-fly" recordings on > answer. The result for each call should be a consecutive/sequential set of > wav files of each X seconds (saved on-the-fly in a specific directory and > using a specific filename stucture)... > > I don't want to reinvent the wheel so that's why I asked if something > similar already exists. Performance is quiet important here (small latency). > > P.S: I bought the FreeSwitch book and I am making progress learning :) > > Thx & cheers, > P. > > > 2011/11/1 Michael Collins > >> Welcome to FreeSWITCH! >> >> First off, let me just mention that this question is probably better >> suited to our -users list. (This is the -dev list which has a smaller >> audience and is reserved for more intense discussions.) If you don't get a >> satisfactory answer here then re-post on freeswitch-users list. >> >> Now, as far as your question goes... you have different options available >> to you but it really depends on how you are generating your calls. Also, do >> you want the call to last more than 30 seconds and have only the recording >> be 30 seconds? Do you want recording to start on early media or on answer? >> >> As an example of how to do a simple 30-second recording here's one that >> when you call it, it will start recording, play music for 30 sec, then end >> the recording and disconnect: >> >> >> >> >> > data="$${recordings_dir}/${uuid}.wav"/> >> >> >> >> >> >> Dial 732664 (i.e. "REC MOH" :) and just wait. You'll hear music and then >> at 30 seconds it will stop recording. >> >> Happy FreeSWITCHing! >> >> -MC >> >> On Tue, Nov 1, 2011 at 1:27 PM, P. Broennimann wrote: >> >>> Hi there >>> >>> I am new to FreeSwitch and I have a question... >>> >>> I'd like to record a call/session outputing same length .wav files (all >>> .wav files to be exactly 30 seconds long). >>> >>> I browsed through the FreeSwitch sources from GIT and I found the module >>> 'mod_snapshot'. Would this module be the best start/base for the >>> functionality I am looking for resp. are there other modules/scripts that I >>> should better check first? >>> >>> -> I tried to build the module in MSVC Express 2010 but I get errors -> >>> Anyone was recently successful building it? >>> >>> Thx in advance, >>> P. >>> >>> >>> _______________________________________________ >>> Join us at ClueCon 2011, Aug 9-11, Chicago >>> http://www.cluecon.com 877-7-4ACLUE >>> >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111101/1997138e/attachment.html From xin at ind.rwth-aachen.de Wed Nov 2 17:42:22 2011 From: xin at ind.rwth-aachen.de (Han Xin) Date: Wed, 02 Nov 2011 15:42:22 +0100 Subject: [Freeswitch-dev] How to display callee_id_name on the caller's phone Message-ID: <10191907bb5b9103bb41ad1b7c8f0597@gw.ind.rwth-aachen.de> Hi all, There is a bridged call, the caller_id_name of leg-A can be displayed correctly on both FS console and callee's phone, but the callee_id_name was always displayed as "Outbound Call"... Some variables during the call are dumped as follows: uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num 8e4a992c-d82f-4919-aee4-6ac0190784ce,inbound,2011-11-02 14:29:15,1320240555,sofia/internal/1000 at asterisk,CS_EXECUTE,Bob 1000 (FS),1000,137.226.198.252,1001,bridge,user/1001 at 137.226.198.143,XML,default,G722,16000,64000,G722,16000,64000,,asterisk,1000 @asterisk,,ACTIVE,Outbound Call,1001,SEND,8e4a992c-d82f-4919-aee4-6ac0190784ce,Outbound Call,1001 be08b9b5-ccd8-44e0-bbd8-4ca93841f930,outbound,2011-11-02 14:29:15,1320240555,sofia/internal/sip:1001 at 137.226.198.252 :5060,CS_EXCHANGE_MEDIA,Bob 1000,1000,137.226.198.252,1001,,,XML,default,G722,16000,64000,G722,16000,64000,,asterisk,1001 at 137.226.198.143,,ACTIVE,Outbound Call,1001,SEND,8e4a992c-d82f-4919-aee4-6ac0190784ce,Bob 1000 (FS),1000 And the configuration of user "Alice 1001"(in file 1001.xml) is like this: (Bob 1000 is also the same kind of config) why the "effective_callee_id_name" does not work? Is there something I have not done in order to display the real callee_id_name? Thanks in advance. Best Regards, Han -- From peter at enovatiq.com Wed Nov 2 17:56:58 2011 From: peter at enovatiq.com (P. Broennimann) Date: Wed, 2 Nov 2011 15:56:58 +0100 Subject: [Freeswitch-dev] Newbie Q In-Reply-To: References: Message-ID: Thx Eliot 4 ur answer Unfortunately post-processing in not an option -> This needs to happen on-the-fly. I am not scared of work and I have time so I will dig deep and try to figure out how to write/modify FreeSwitch C modules... Thanks & cheers, P. 2011/11/2 Eliot Gable > Maybe just record the entire call and then split it up afterwards using a > wav-file post-processing program. Maybe sox or ffmpeg or something similar > might work? Alternatively, you could use the API in FreeSWITCH to start and > stop and start and stop the recording every 30 seconds, but that seems like > overkill and lots of extra work compared to just using post-processing > software. Plus, you'll miss bits of the recording between when you stop one > segment and start the next. Is there some reason you need to be doing this > on the fly versus post-processing? Alternatively, you could write a module > in C for FreeSWITCH which taps the audio and you could buffer it internally > in your module and then save out 30-second segments. This would let you get > all of the audio with micro-second precision splits so you wouldn't loose > any audio between segments, but it's an awful lot of work to go through > compared to post-processing. > > > On Tue, Nov 1, 2011 at 6:50 PM, P. Broennimann wrote: > >> Hi Michael >> >> Thx for answering. Sorry next time I'll post on the other list... >> >> Basically I am looking for a solution to split "on-the-fly" recordings on >> answer. The result for each call should be a consecutive/sequential set of >> wav files of each X seconds (saved on-the-fly in a specific directory and >> using a specific filename stucture)... >> >> I don't want to reinvent the wheel so that's why I asked if something >> similar already exists. Performance is quiet important here (small latency). >> >> P.S: I bought the FreeSwitch book and I am making progress learning :) >> >> Thx & cheers, >> P. >> >> >> 2011/11/1 Michael Collins >> >>> Welcome to FreeSWITCH! >>> >>> First off, let me just mention that this question is probably better >>> suited to our -users list. (This is the -dev list which has a smaller >>> audience and is reserved for more intense discussions.) If you don't get a >>> satisfactory answer here then re-post on freeswitch-users list. >>> >>> Now, as far as your question goes... you have different options >>> available to you but it really depends on how you are generating your >>> calls. Also, do you want the call to last more than 30 seconds and have >>> only the recording be 30 seconds? Do you want recording to start on early >>> media or on answer? >>> >>> As an example of how to do a simple 30-second recording here's one that >>> when you call it, it will start recording, play music for 30 sec, then end >>> the recording and disconnect: >>> >>> >>> >>> >>> >> data="$${recordings_dir}/${uuid}.wav"/> >>> >>> >>> >>> >>> >>> Dial 732664 (i.e. "REC MOH" :) and just wait. You'll hear music and then >>> at 30 seconds it will stop recording. >>> >>> Happy FreeSWITCHing! >>> >>> -MC >>> >>> On Tue, Nov 1, 2011 at 1:27 PM, P. Broennimann wrote: >>> >>>> Hi there >>>> >>>> I am new to FreeSwitch and I have a question... >>>> >>>> I'd like to record a call/session outputing same length .wav files (all >>>> .wav files to be exactly 30 seconds long). >>>> >>>> I browsed through the FreeSwitch sources from GIT and I found the >>>> module 'mod_snapshot'. Would this module be the best start/base for the >>>> functionality I am looking for resp. are there other modules/scripts that I >>>> should better check first? >>>> >>>> -> I tried to build the module in MSVC Express 2010 but I get errors -> >>>> Anyone was recently successful building it? >>>> >>>> Thx in advance, >>>> P. >>>> >>>> >>>> _______________________________________________ >>>> Join us at ClueCon 2011, Aug 9-11, Chicago >>>> http://www.cluecon.com 877-7-4ACLUE >>>> >>>> FreeSWITCH-dev mailing list >>>> FreeSWITCH-dev at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Join us at ClueCon 2011, Aug 9-11, Chicago >>> http://www.cluecon.com 877-7-4ACLUE >>> >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > > -- > Eliot Gable > > "We do not inherit the Earth from our ancestors: we borrow it from our > children." ~David Brower > > "I decided the words were too conservative for me. We're not borrowing > from our children, we're stealing from them--and it's not even considered > to be a crime." ~David Brower > > "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; > not live to eat.) ~Marcus Tullius Cicero > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111102/fd262705/attachment-0001.html From msc at freeswitch.org Wed Nov 2 18:09:31 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Nov 2011 08:09:31 -0700 Subject: [Freeswitch-dev] Newbie Q In-Reply-To: References: Message-ID: Before you go the route of hacking modules be sure to examine the extensive list of dialplan tools and FS cmd line APIs. Personally, I would experiment with the sched_api and uuid_broadcast API's. You could create a generic dialplan extension and then have the sched_api perform a uuid_broadcast that does an execute_extension on the uuid of the call leg. The (generic) extension you execute can stop the current recording, start up a new recording, and schedule another uuid_broadcast for 30 seconds later. It should be a fun exercise! :) -MC On Wed, Nov 2, 2011 at 7:56 AM, P. Broennimann wrote: > Thx Eliot 4 ur answer > > Unfortunately post-processing in not an option -> This needs to happen > on-the-fly. > > I am not scared of work and I have time so I will dig deep > and try to figure out how to write/modify FreeSwitch C modules... > > Thanks & cheers, > P. > > > > > 2011/11/2 Eliot Gable > >> Maybe just record the entire call and then split it up afterwards using a >> wav-file post-processing program. Maybe sox or ffmpeg or something similar >> might work? Alternatively, you could use the API in FreeSWITCH to start and >> stop and start and stop the recording every 30 seconds, but that seems like >> overkill and lots of extra work compared to just using post-processing >> software. Plus, you'll miss bits of the recording between when you stop one >> segment and start the next. Is there some reason you need to be doing this >> on the fly versus post-processing? Alternatively, you could write a module >> in C for FreeSWITCH which taps the audio and you could buffer it internally >> in your module and then save out 30-second segments. This would let you get >> all of the audio with micro-second precision splits so you wouldn't loose >> any audio between segments, but it's an awful lot of work to go through >> compared to post-processing. >> >> >> On Tue, Nov 1, 2011 at 6:50 PM, P. Broennimann wrote: >> >>> Hi Michael >>> >>> Thx for answering. Sorry next time I'll post on the other list... >>> >>> Basically I am looking for a solution to split "on-the-fly" recordings >>> on answer. The result for each call should be a consecutive/sequential set >>> of wav files of each X seconds (saved on-the-fly in a specific directory >>> and using a specific filename stucture)... >>> >>> I don't want to reinvent the wheel so that's why I asked if something >>> similar already exists. Performance is quiet important here (small latency). >>> >>> P.S: I bought the FreeSwitch book and I am making progress learning :) >>> >>> Thx & cheers, >>> P. >>> >>> >>> 2011/11/1 Michael Collins >>> >>>> Welcome to FreeSWITCH! >>>> >>>> First off, let me just mention that this question is probably better >>>> suited to our -users list. (This is the -dev list which has a smaller >>>> audience and is reserved for more intense discussions.) If you don't get a >>>> satisfactory answer here then re-post on freeswitch-users list. >>>> >>>> Now, as far as your question goes... you have different options >>>> available to you but it really depends on how you are generating your >>>> calls. Also, do you want the call to last more than 30 seconds and have >>>> only the recording be 30 seconds? Do you want recording to start on early >>>> media or on answer? >>>> >>>> As an example of how to do a simple 30-second recording here's one that >>>> when you call it, it will start recording, play music for 30 sec, then end >>>> the recording and disconnect: >>>> >>>> >>>> >>>> >>>> >>> data="$${recordings_dir}/${uuid}.wav"/> >>>> >>>> >>>> >>>> >>>> >>>> Dial 732664 (i.e. "REC MOH" :) and just wait. You'll hear music and >>>> then at 30 seconds it will stop recording. >>>> >>>> Happy FreeSWITCHing! >>>> >>>> -MC >>>> >>>> On Tue, Nov 1, 2011 at 1:27 PM, P. Broennimann wrote: >>>> >>>>> Hi there >>>>> >>>>> I am new to FreeSwitch and I have a question... >>>>> >>>>> I'd like to record a call/session outputing same length .wav files >>>>> (all .wav files to be exactly 30 seconds long). >>>>> >>>>> I browsed through the FreeSwitch sources from GIT and I found the >>>>> module 'mod_snapshot'. Would this module be the best start/base for the >>>>> functionality I am looking for resp. are there other modules/scripts that I >>>>> should better check first? >>>>> >>>>> -> I tried to build the module in MSVC Express 2010 but I get errors >>>>> -> Anyone was recently successful building it? >>>>> >>>>> Thx in advance, >>>>> P. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Join us at ClueCon 2011, Aug 9-11, Chicago >>>>> http://www.cluecon.com 877-7-4ACLUE >>>>> >>>>> FreeSWITCH-dev mailing list >>>>> FreeSWITCH-dev at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Join us at ClueCon 2011, Aug 9-11, Chicago >>>> http://www.cluecon.com 877-7-4ACLUE >>>> >>>> FreeSWITCH-dev mailing list >>>> FreeSWITCH-dev at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Join us at ClueCon 2011, Aug 9-11, Chicago >>> http://www.cluecon.com 877-7-4ACLUE >>> >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Eliot Gable >> >> "We do not inherit the Earth from our ancestors: we borrow it from our >> children." ~David Brower >> >> "I decided the words were too conservative for me. We're not borrowing >> from our children, we're stealing from them--and it's not even considered >> to be a crime." ~David Brower >> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; >> not live to eat.) ~Marcus Tullius Cicero >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111102/bf8ccaac/attachment.html From msc at freeswitch.org Wed Nov 2 18:21:45 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Nov 2011 08:21:45 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today - New Dialplan Feature, SIP 101 Message-ID: Hello all! In our conference call today we are going to discuss a cool new dialplan feature (better logical OR processing) and then Ken Rice and I will be discussing some basic SIP knowledge to help get everyone up to speed when helping newbies. Talk to you all soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111102/3600de93/attachment.html From anthony.minessale at gmail.com Wed Nov 2 19:07:09 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Nov 2011 11:07:09 -0500 Subject: [Freeswitch-dev] How to display callee_id_name on the caller's phone In-Reply-To: <10191907bb5b9103bb41ad1b7c8f0597@gw.ind.rwth-aachen.de> References: <10191907bb5b9103bb41ad1b7c8f0597@gw.ind.rwth-aachen.de> Message-ID: add origination_callee_id_name to the dial string On Wed, Nov 2, 2011 at 9:42 AM, Han Xin wrote: > Hi all, > There is a bridged call, the caller_id_name of leg-A can be displayed > correctly on both FS console and callee's phone, but the callee_id_name was > always displayed as "Outbound Call"... > > Some variables during the call are dumped as follows: > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num > > 8e4a992c-d82f-4919-aee4-6ac0190784ce,inbound,2011-11-02 > 14:29:15,1320240555,sofia/internal/1000 at asterisk,CS_EXECUTE,Bob 1000 > (FS),1000,137.226.198.252,1001,bridge,user/1001 at 137.226.198.143,XML,default,G722,16000,64000,G722,16000,64000,,asterisk,1000 > @asterisk,,ACTIVE,Outbound > Call,1001,SEND,8e4a992c-d82f-4919-aee4-6ac0190784ce,Outbound Call,1001 > > be08b9b5-ccd8-44e0-bbd8-4ca93841f930,outbound,2011-11-02 > 14:29:15,1320240555,sofia/internal/sip:1001 at 137.226.198.252:5060,CS_EXCHANGE_MEDIA,Bob > 1000,1000,137.226.198.252,1001,,,XML,default,G722,16000,64000,G722,16000,64000,,asterisk, > 1001 at 137.226.198.143,,ACTIVE,Outbound > Call,1001,SEND,8e4a992c-d82f-4919-aee4-6ac0190784ce,Bob 1000 (FS),1000 > > And the configuration of user "Alice 1001"(in file 1001.xml) is like this: > (Bob 1000 is also the same kind of config) > > > > > > > > > > > > > > > > > > > > > > why the "effective_callee_id_name" does not work? Is there something I > have not done in order to display the real callee_id_name? > > Thanks in advance. > > Best Regards, > Han > -- > > > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111102/4ec02349/attachment-0001.html From khovayko at gmail.com Thu Nov 3 00:16:57 2011 From: khovayko at gmail.com (Oleg Khovayko) Date: Wed, 02 Nov 2011 17:16:57 -0400 Subject: [Freeswitch-dev] Nortel 1535 drops connection to FS In-Reply-To: References: <10191907bb5b9103bb41ad1b7c8f0597@gw.ind.rwth-aachen.de> Message-ID: <4EB1B349.9010204@gmail.com> Hi, I am using FS for couple years, it was works OK. And, approx 1 year ago, I tested with FreeSWITCH phone Nortel 1535. It worked perfect. Thereafter, I couple times upgraded FS, and right now, I run version from GIT, dated 11-10-29. OS: FreeBSD 7.2, 32 bits. I again connected Nortel 1535, and this time see, connection suddenly broken during call to any "robot": FS #5000, FS #4000, CallCentric #17771234567 Every time, before disconnect, I see chain of log messages like: [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] and HANGUP thereafter. See following samples from the log: Another phones, like VM1188T or PAP2, still works OK. About Nortel: I tested 3 phones, with two firmwares (2.50 & 2.76) -- problem same on both. Can you suggest me, how to fix this issue? Thanks, Oleg CallCentric: 7771234567 2011-11-02 16:39:20.901219 [DEBUG] sofia_reg.c:1984 Changing expire time to 103 by request of proxy sip:callcentric.com 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 16:39:31.130669 [DEBUG] sofia.c:5283 Channel sofia/internal/1000 at olegh.ath.cx entering state [terminating][0] 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2804 (sofia/internal/1000 at olegh.ath.cx) Callstate Change ACTIVE -> HANGUP 2011-11-02 16:39:31.130669 [NOTICE] sofia.c:6039 Hangup sofia/internal/1000 at olegh.ath.cx [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2820 Send signal sofia/internal/1000 at olegh.ath.cx [KILL] 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] FS: 5000 2011-11-02 17:03:08.207852 [DEBUG] switch_ivr_play_say.c:1672 done playing file /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-to_do_a_freeswitch_echo_test.wav 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[ivr/ivr-please.wav] (ru:ru) 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 17:03:09.280490 [DEBUG] sofia.c:5283 Channel sofia/internal/1000 at olegh.ath.cx entering state [terminating][0] 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2804 (sofia/internal/1000 at olegh.ath.cx) Callstate Change ACTIVE -> HANGUP 2011-11-02 17:03:09.280490 [NOTICE] sofia.c:6039 Hangup sofia/internal/1000 at olegh.ath.cx [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2820 Send signal sofia/internal/1000 at olegh.ath.cx [KILL] 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 17:03:09.280490 [DEBUG] switch_ivr_play_say.c:1672 done playing file /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-please.wav 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:348 waiting for 4/4 digits t/o 2000 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:395 digits '' 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:585 IVR menu 'demo_ivr' no input detected 2011-11-02 17:03:10.587476 [DEBUG] switch_ivr_menu.c:599 exit-sound 'voicemail/vm-goodbye.wav' 2011-11-02 17:03:10.587476 [DEBUG] switch_core_session.c:2269 sofia/internal/1000 at olegh.ath.cx skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) FS: voicemail 2011-11-02 16:51:50.217856 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [ru] 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1290554584] (ru:ru) 2011-11-02 16:51:50.307721 [INFO] mod_say_ru.c:721 ru_say!!! 1290554584! say_opt.gender=0 say_opt.cases=0 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 16:51:50.990445 [DEBUG] sofia.c:5283 Channel sofia/internal/1000 at olegh.ath.cx entering state [terminating][0] 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2804 (sofia/internal/1000 at olegh.ath.cx) Callstate Change ACTIVE -> HANGUP 2011-11-02 16:51:50.990445 [NOTICE] sofia.c:6039 Hangup sofia/internal/1000 at olegh.ath.cx [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2820 Send signal sofia/internal/1000 at olegh.ath.cx [KILL] 2011-11-02 16:51:50.990445 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] From anthony.minessale at gmail.com Thu Nov 3 00:24:06 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Nov 2011 16:24:06 -0500 Subject: [Freeswitch-dev] Nortel 1535 drops connection to FS In-Reply-To: <4EB1B349.9010204@gmail.com> References: <10191907bb5b9103bb41ad1b7c8f0597@gw.ind.rwth-aachen.de> <4EB1B349.9010204@gmail.com> Message-ID: snippets of logs rarely are helpful, try complete debug logs including sip trace On Wed, Nov 2, 2011 at 4:16 PM, Oleg Khovayko wrote: > Hi, > > I am using FS for couple years, it was works OK. > And, approx 1 year ago, I tested with FreeSWITCH phone Nortel 1535. > It worked perfect. > > Thereafter, I couple times upgraded FS, and right now, I run version > from GIT, dated 11-10-29. OS: FreeBSD 7.2, 32 bits. > > I again connected Nortel 1535, and this time see, connection suddenly > broken during call to any "robot": > FS #5000, FS #4000, CallCentric #17771234567 > > Every time, before disconnect, I see chain of log messages like: > > [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > and HANGUP thereafter. See following samples from the log: > > Another phones, like VM1188T or PAP2, still works OK. > About Nortel: I tested 3 phones, with two firmwares (2.50 & 2.76) -- > problem same on both. > > Can you suggest me, how to fix this issue? > > Thanks, > > Oleg > > > > CallCentric: 7771234567 > > 2011-11-02 16:39:20.901219 [DEBUG] sofia_reg.c:1984 Changing expire time > to 103 by request of proxy sip:callcentric.com > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.130669 [DEBUG] sofia.c:5283 Channel > sofia/internal/1000 at olegh.ath.cx entering state [terminating][0] > 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2804 > (sofia/internal/1000 at olegh.ath.cx) Callstate Change ACTIVE -> HANGUP > 2011-11-02 16:39:31.130669 [NOTICE] sofia.c:6039 Hangup > sofia/internal/1000 at olegh.ath.cx [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2820 Send signal > sofia/internal/1000 at olegh.ath.cx [KILL] > 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/1000 at olegh.ath.cx [BREAK] > > > FS: 5000 > > 2011-11-02 17:03:08.207852 [DEBUG] switch_ivr_play_say.c:1672 done > playing file > > /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-to_do_a_freeswitch_echo_test.wav > 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[ivr/ivr-please.wav] (ru:ru) > 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:1302 Codec > Activated L16 at 8000hz 1 channels 20ms > 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.280490 [DEBUG] sofia.c:5283 Channel > sofia/internal/1000 at olegh.ath.cx entering state [terminating][0] > 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2804 > (sofia/internal/1000 at olegh.ath.cx) Callstate Change ACTIVE -> HANGUP > 2011-11-02 17:03:09.280490 [NOTICE] sofia.c:6039 Hangup > sofia/internal/1000 at olegh.ath.cx [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2820 Send signal > sofia/internal/1000 at olegh.ath.cx [KILL] > 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.280490 [DEBUG] switch_ivr_play_say.c:1672 done > playing file /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-please.wav > 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:348 waiting for 4/4 > digits t/o 2000 > 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:395 digits '' > 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:585 IVR menu > 'demo_ivr' no input detected > 2011-11-02 17:03:10.587476 [DEBUG] switch_ivr_menu.c:599 exit-sound > 'voicemail/vm-goodbye.wav' > 2011-11-02 17:03:10.587476 [DEBUG] switch_core_session.c:2269 > sofia/internal/1000 at olegh.ath.cx skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > > FS: voicemail > > 2011-11-02 16:51:50.217856 [DEBUG] switch_ivr_play_say.c:67 No language > specified - Using [ru] > 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:244 Handle > say:[1290554584] (ru:ru) > 2011-11-02 16:51:50.307721 [INFO] mod_say_ru.c:721 ru_say!!! > 1290554584! say_opt.gender=0 say_opt.cases=0 > 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:1302 Codec > Activated L16 at 8000hz 1 channels 20ms > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.990445 [DEBUG] sofia.c:5283 Channel > sofia/internal/1000 at olegh.ath.cx entering state [terminating][0] > 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2804 > (sofia/internal/1000 at olegh.ath.cx) Callstate Change ACTIVE -> HANGUP > 2011-11-02 16:51:50.990445 [NOTICE] sofia.c:6039 Hangup > sofia/internal/1000 at olegh.ath.cx [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2820 Send signal > sofia/internal/1000 at olegh.ath.cx [KILL] > 2011-11-02 16:51:50.990445 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/1000 at olegh.ath.cx [BREAK] > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111102/66fa281e/attachment.html From khovayko at gmail.com Thu Nov 3 03:23:18 2011 From: khovayko at gmail.com (Oleg Khovayko) Date: Wed, 02 Nov 2011 20:23:18 -0400 Subject: [Freeswitch-dev] Nortel 1535 drops connection to FS In-Reply-To: References: <10191907bb5b9103bb41ad1b7c8f0597@gw.ind.rwth-aachen.de> <4EB1B349.9010204@gmail.com> Message-ID: <4EB1DEF6.40008@gmail.com> Excuse me, I can not enforce FS to write into logfile all debug info. I trued to run according http://wiki.freeswitch.org/wiki/Reporting_Bugs: TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch but anyway, siptrace does not writing to log, only to fs_cli terminal output. Anyway, I caught SIPtrace messages for Nortel, this is piece of terminal output. I see, FS sends signal BYE, because ACK timeout: 2011-11-02 18:32:07.929071 [DEBUG] switch_ivr_play_say.c:1672 done playing file /usr/local/freeswitch/sounds/ru/RU/elena/digits/1.wav recv 330 bytes from udp/[192.168.1.17]:5061 at 22:32:07.995778: ------------------------------------------------------------------------ NOTIFY sip:192.168.1.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.17:5061;branch=z9hG4bK-c51ee6c4 From: 1004;tag=12dc6681765b973do1 To: Call-ID: af69bdb1-3c02a9cd at 192.168.1.17 CSeq: 274133 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/PAP2-3.1.22(LS) Content-Length: 0 ------------------------------------------------------------------------ send 669 bytes to udp/[192.168.1.17]:5061 at 22:32:08.001770: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.17:5061;branch=z9hG4bK-c51ee6c4 From: 1004;tag=12dc6681765b973do1 To:;tag=c0N5gH4XH8DQm Call-ID: af69bdb1-3c02a9cd at 192.168.1.17 CSeq: 274133 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f847059 2011-10-28 15-58-17 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ 2011-11-02 18:32:08.195841 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[ivr/ivr-to_do_a_freeswitch_echo_test.wav] (ru:ru) 2011-11-02 18:32:08.195841 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms send 1154 bytes to udp/[192.168.1.158]:5060 at 22:32:09.084079: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bK-6e-1ae15-7e234642 From: "1000";tag=2f5c50-9e01a8c0-13c4-45026-6d-ad711f2-6d To:;tag=1vZmX7aQH6r9m Call-ID: 3039a8-9e01a8c0-13c4-45026-6d-1a07e714-6d CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f847059 2011-10-28 15-58-17 -0500 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 242 Remote-Party-ID: "5000";party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1320248473 1320248474 IN IP4 192.168.1.5 s=FreeSWITCH c=IN IP4 192.168.1.5 t=0 0 m=audio 24624 RTP/AVP 0 99 a=rtpmap:0 PCMU/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2011-11-02 18:32:09.554915 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] send 685 bytes to udp/[192.168.1.158]:5060 at 22:32:09.573772: ------------------------------------------------------------------------ BYE sip:1000 at 192.168.1.158:5060 SIP/2.0 Via: SIP/2.0/UDP 173.79.237.112;rport;branch=z9hG4bKmBgZ4SN8yDSag Max-Forwards: 70 From:;tag=1vZmX7aQH6r9m To: "1000";tag=2f5c50-9e01a8c0-13c4-45026-6d-ad711f2-6d Call-ID: 3039a8-9e01a8c0-13c4-45026-6d-1a07e714-6d CSeq: 19800500 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f847059 2011-10-28 15-58-17 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: SIP;cause=408;text="ACK Timeout" Content-Length: 0 ------------------------------------------------------------------------ 2011-11-02 18:32:09.554915 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] recv 479 bytes from udp/[192.168.1.158]:5060 at 22:32:09.586581: ------------------------------------------------------------------------ SIP/2.0 200 OK From:;tag=1vZmX7aQH6r9m To: "1000";tag=2f5c50-9e01a8c0-13c4-45026-6d-ad711f2-6d Call-ID: 3039a8-9e01a8c0-13c4-45026-6d-1a07e714-6d CSeq: 19800500 BYE Via: SIP/2.0/UDP 173.79.237.112;received=192.168.1.5;rport=5060;branch=z9hG4bKmBgZ4SN8yDSag Supported: replaces Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: Nortel IP Phone 1535 (0.2.50.0905) Content-Length: 0 ------------------------------------------------------------------------ 2011-11-02 18:32:09.582519 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1000 at olegh.ath.cx [BREAK] 2011-11-02 18:32:09.582519 [DEBUG] sofia.c:5283 Channel sofia/internal/1000 at olegh.ath.cx entering state [terminating][0] 2011-11-02 18:32:09.582519 [DEBUG] switch_channel.c:2804 (sofia/internal/1000 at olegh.ath.cx) Callstate Change ACTIVE -> HANGUP 2011-11-02 18:32:09.582519 [NOTICE] sofia.c:6039 Hangup sofia/internal/1000 at olegh.ath.cx [CS_EXECUTE] [NORMAL_UNSPECIFIED] Anthony Minessale wrote: > snippets of logs rarely are helpful, try complete debug logs including > sip trace > > > On Wed, Nov 2, 2011 at 4:16 PM, Oleg Khovayko > wrote: > > Hi, > > I am using FS for couple years, it was works OK. > And, approx 1 year ago, I tested with FreeSWITCH phone Nortel 1535. > It worked perfect. > > Thereafter, I couple times upgraded FS, and right now, I run version > from GIT, dated 11-10-29. OS: FreeBSD 7.2, 32 bits. > > I again connected Nortel 1535, and this time see, connection suddenly > broken during call to any "robot": > FS #5000, FS #4000, CallCentric #17771234567 > > Every time, before disconnect, I see chain of log messages like: > > [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > and HANGUP thereafter. See following samples from the log: > > Another phones, like VM1188T or PAP2, still works OK. > About Nortel: I tested 3 phones, with two firmwares (2.50 & 2.76) -- > problem same on both. > > Can you suggest me, how to fix this issue? > > Thanks, > > Oleg > > > > CallCentric: 7771234567 > > 2011-11-02 16:39:20.901219 [DEBUG] sofia_reg.c:1984 Changing > expire time > to 103 by request of proxy sip:callcentric.com > > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.130669 [DEBUG] sofia.c:5283 Channel > sofia/internal/1000 at olegh.ath.cx > entering state [terminating][0] > 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2804 > (sofia/internal/1000 at olegh.ath.cx ) > Callstate Change ACTIVE -> HANGUP > 2011-11-02 16:39:31.130669 [NOTICE] sofia.c:6039 Hangup > sofia/internal/1000 at olegh.ath.cx > [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2820 Send signal > sofia/internal/1000 at olegh.ath.cx [KILL] > 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/1000 at olegh.ath.cx > [BREAK] > > > FS: 5000 > > 2011-11-02 17:03:08.207852 [DEBUG] switch_ivr_play_say.c:1672 done > playing file > /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-to_do_a_freeswitch_echo_test.wav > 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[ivr/ivr-please.wav] (ru:ru) > 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:1302 Codec > Activated L16 at 8000hz 1 channels 20ms > 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.280490 [DEBUG] sofia.c:5283 Channel > sofia/internal/1000 at olegh.ath.cx > entering state [terminating][0] > 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2804 > (sofia/internal/1000 at olegh.ath.cx ) > Callstate Change ACTIVE -> HANGUP > 2011-11-02 17:03:09.280490 [NOTICE] sofia.c:6039 Hangup > sofia/internal/1000 at olegh.ath.cx > [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2820 Send signal > sofia/internal/1000 at olegh.ath.cx [KILL] > 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/1000 at olegh.ath.cx > [BREAK] > 2011-11-02 17:03:09.280490 [DEBUG] switch_ivr_play_say.c:1672 done > playing file > /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-please.wav > 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:348 waiting > for 4/4 > digits t/o 2000 > 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:395 digits '' > 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:585 IVR menu > 'demo_ivr' no input detected > 2011-11-02 17:03:10.587476 [DEBUG] switch_ivr_menu.c:599 exit-sound > 'voicemail/vm-goodbye.wav' > 2011-11-02 17:03:10.587476 [DEBUG] switch_core_session.c:2269 > sofia/internal/1000 at olegh.ath.cx skip > receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > > FS: voicemail > > 2011-11-02 16:51:50.217856 [DEBUG] switch_ivr_play_say.c:67 No > language > specified - Using [ru] > 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:244 Handle > say:[1290554584] (ru:ru) > 2011-11-02 16:51:50.307721 [INFO] mod_say_ru.c:721 ru_say!!! > 1290554584! say_opt.gender=0 say_opt.cases=0 > 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:1302 Codec > Activated L16 at 8000hz 1 channels 20ms > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.990445 [DEBUG] sofia.c:5283 Channel > sofia/internal/1000 at olegh.ath.cx > entering state [terminating][0] > 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2804 > (sofia/internal/1000 at olegh.ath.cx ) > Callstate Change ACTIVE -> HANGUP > 2011-11-02 16:51:50.990445 [NOTICE] sofia.c:6039 Hangup > sofia/internal/1000 at olegh.ath.cx > [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2820 Send signal > sofia/internal/1000 at olegh.ath.cx [KILL] > 2011-11-02 16:51:50.990445 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/1000 at olegh.ath.cx > [BREAK] > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111102/ee19aaa8/attachment-0001.html From xin at ind.rwth-aachen.de Thu Nov 3 13:48:24 2011 From: xin at ind.rwth-aachen.de (Han Xin) Date: Thu, 03 Nov 2011 11:48:24 +0100 Subject: [Freeswitch-dev] How to display callee_id_name on the caller'sphone In-Reply-To: Message-ID: Thanks Anthony! It is solved by adding to the dial plan to realize the display of callee's name in FS console. The name displayed in the caller's phone is still the callee's number, but it does not matter. Thans again! ----------------original message----------------- From: "Anthony Minessale" anthony.minessale at gmail.com To: freeswitch-dev at lists.freeswitch.org Date: Wed, 02 Nov 2011 11:07:09 -0500 ------------------------------------------------- > add origination_callee_id_name to the dial string > > > > On Wed, Nov 2, 2011 at 9:42 AM, Han Xin xin at ind.rwth-aachen.de wrote: > >> Hi all, >> There is a bridged call, the caller_id_name of leg-A can be displayed >> correctly on both FS console and callee's phone, but the callee_id_name was >> always displayed as "Outbound Call"... >> >> Some variables during the call are dumped as follows: >> >> >> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,i >> p_addr,dest,application,application_data,dialplan,context,read_cod >> ec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,s >> ecure,hostname,presence_id,presence_data,callstate,callee_name,cal >> lee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_nu >> m >> >> 8e4a992c-d82f-4919-aee4-6ac0190784ce,inbound,2011-11-02 >> 14:29:15,1320240555,sofia/internal/1000 at asterisk,CS_EXECUTE,Bob >> 1000 >> >> (FS),1000,137.226.198.252,1001,bridge,user/1001 at 137.226.198.143,XM >> L,default,G722,16000,64000,G722,16000,64000,,asterisk,1000 >> @asterisk,,ACTIVE,Outbound >> Call,1001,SEND,8e4a992c-d82f-4919-aee4-6ac0190784ce,Outbound >> Call,1001 >> >> be08b9b5-ccd8-44e0-bbd8-4ca93841f930,outbound,2011-11-02 >> >> 14:29:15,1320240555,sofia/internal/sip:1001 at 137.226.198.252 :5060,C >> S_EXCHANGE_MEDIA,Bob >> >> 1000,1000,137.226.198.252,1001,,,XML,default,G722,16000,64000,G722 >> ,16000,64000,,asterisk, >> 1001 at 137.226.198.143,,ACTIVE,Outbound >> Call,1001,SEND,8e4a992c-d82f-4919-aee4-6ac0190784ce,Bob 1000 >> (FS),1000 >> >> And the configuration of user "Alice 1001"(in file 1001.xml) is like this: >> (Bob 1000 is also the same kind of config) >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> why the "effective_callee_id_name" does not work? Is there something I >> have not done in order to display the real callee_id_name? >> >> Thanks in advance. >> >> Best Regards, >> Han >> -- >> >> >> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch >> -dev >> 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > __________________________________________________ > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-d > ev > http://www.freeswitch.org > -- From anthony.minessale at gmail.com Thu Nov 3 20:16:32 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Nov 2011 12:16:32 -0500 Subject: [Freeswitch-dev] How to display callee_id_name on the caller'sphone In-Reply-To: References: Message-ID: you can also set origination_caller_id_number if you wish On Thu, Nov 3, 2011 at 5:48 AM, Han Xin wrote: > Thanks Anthony! It is solved by adding > > data="{origination_callee_id_name=${user_data(${dialed_extension}@${domain_name} > var effective_caller_id_name)}}sofia/internal/foo at bar.com "/> > > to the dial plan to realize the display of callee's name in FS console. > The name displayed in the caller's phone is still the callee's number, but > it does not matter. Thans again! > > ----------------original message----------------- > From: "Anthony Minessale" anthony.minessale at gmail.com > To: freeswitch-dev at lists.freeswitch.org > Date: Wed, 02 Nov 2011 11:07:09 -0500 > ------------------------------------------------- > > > > add origination_callee_id_name to the dial string > > > > > > > > On Wed, Nov 2, 2011 at 9:42 AM, Han Xin xin at ind.rwth-aachen.de wrote: > > > >> Hi all, > >> There is a bridged call, the caller_id_name of leg-A can be displayed > >> correctly on both FS console and callee's phone, but the callee_id_name > was > >> always displayed as "Outbound Call"... > >> > >> Some variables during the call are dumped as follows: > >> > >> > >> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,i > >> p_addr,dest,application,application_data,dialplan,context,read_cod > >> ec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,s > >> ecure,hostname,presence_id,presence_data,callstate,callee_name,cal > >> lee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_nu > >> m > >> > >> 8e4a992c-d82f-4919-aee4-6ac0190784ce,inbound,2011-11-02 > >> 14:29:15,1320240555,sofia/internal/1000 at asterisk,CS_EXECUTE,Bob > >> 1000 > >> > >> (FS),1000,137.226.198.252,1001,bridge,user/1001 at 137.226.198.143,XM > >> L,default,G722,16000,64000,G722,16000,64000,,asterisk,1000 > >> @asterisk,,ACTIVE,Outbound > >> Call,1001,SEND,8e4a992c-d82f-4919-aee4-6ac0190784ce,Outbound > >> Call,1001 > >> > >> be08b9b5-ccd8-44e0-bbd8-4ca93841f930,outbound,2011-11-02 > >> > >> 14:29:15,1320240555,sofia/internal/sip:1001 at 137.226.198.252 :5060,C > >> S_EXCHANGE_MEDIA,Bob > >> > >> 1000,1000,137.226.198.252,1001,,,XML,default,G722,16000,64000,G722 > >> ,16000,64000,,asterisk, > >> 1001 at 137.226.198.143,,ACTIVE,Outbound > >> Call,1001,SEND,8e4a992c-d82f-4919-aee4-6ac0190784ce,Bob 1000 > >> (FS),1000 > >> > >> And the configuration of user "Alice 1001"(in file 1001.xml) is like > this: > >> (Bob 1000 is also the same kind of config) > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> why the "effective_callee_id_name" does not work? Is there something I > >> have not done in order to display the real callee_id_name? > >> > >> Thanks in advance. > >> > >> Best Regards, > >> Han > >> -- > >> > >> > >> > >> > >> _______________________________________________ > >> Join us at ClueCon 2011, Aug 9-11, Chicago > >> http://www.cluecon.com 877-7-4ACLUE > >> > >> FreeSWITCH-dev mailing list > >> FreeSWITCH-dev at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch > >> -dev > >> 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > __________________________________________________ > > _______________________________________________ > > Join us at ClueCon 2011, Aug 9-11, Chicago > > http://www.cluecon.com 877-7-4ACLUE > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-d > > ev > > http://www.freeswitch.org > > > > -- > > > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111103/2cd63eb5/attachment.html From khovayko at gmail.com Thu Nov 3 22:38:47 2011 From: khovayko at gmail.com (Oleg Khovayko) Date: Thu, 03 Nov 2011 15:38:47 -0400 Subject: [Freeswitch-dev] Nortel 1535 drops connection to FS In-Reply-To: References: <10191907bb5b9103bb41ad1b7c8f0597@gw.ind.rwth-aachen.de> <4EB1B349.9010204@gmail.com> Message-ID: <4EB2EDC7.3080400@gmail.com> Anthony, Thank you very much for assistance. I produced log-file, contains siptrace, you can get it: http://olegh.ath.cx/freeswitch.log-1535-1 In thins test, I called from numbed 1000 to number 5000. Oleg Anthony Minessale wrote: > snippets of logs rarely are helpful, try complete debug logs including > sip trace > > > On Wed, Nov 2, 2011 at 4:16 PM, Oleg Khovayko > wrote: > > Hi, > > I am using FS for couple years, it was works OK. > And, approx 1 year ago, I tested with FreeSWITCH phone Nortel 1535. > It worked perfect. > > Thereafter, I couple times upgraded FS, and right now, I run version > from GIT, dated 11-10-29. OS: FreeBSD 7.2, 32 bits. > > I again connected Nortel 1535, and this time see, connection suddenly > broken during call to any "robot": > FS #5000, FS #4000, CallCentric #17771234567 > > Every time, before disconnect, I see chain of log messages like: > > [DEBUG] switch_core_session.c:872 Send signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > and HANGUP thereafter. See following samples from the log: > > Another phones, like VM1188T or PAP2, still works OK. > About Nortel: I tested 3 phones, with two firmwares (2.50 & 2.76) -- > problem same on both. > > Can you suggest me, how to fix this issue? > > Thanks, > > Oleg > > > > CallCentric: 7771234567 > > 2011-11-02 16:39:20.901219 [DEBUG] sofia_reg.c:1984 Changing > expire time > to 103 by request of proxy sip:callcentric.com > > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:39:31.130669 [DEBUG] sofia.c:5283 Channel > sofia/internal/1000 at olegh.ath.cx > entering state [terminating][0] > 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2804 > (sofia/internal/1000 at olegh.ath.cx ) > Callstate Change ACTIVE -> HANGUP > 2011-11-02 16:39:31.130669 [NOTICE] sofia.c:6039 Hangup > sofia/internal/1000 at olegh.ath.cx > [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2820 Send signal > sofia/internal/1000 at olegh.ath.cx [KILL] > 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/1000 at olegh.ath.cx > [BREAK] > > > FS: 5000 > > 2011-11-02 17:03:08.207852 [DEBUG] switch_ivr_play_say.c:1672 done > playing file > /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-to_do_a_freeswitch_echo_test.wav > 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[ivr/ivr-please.wav] (ru:ru) > 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:1302 Codec > Activated L16 at 8000hz 1 channels 20ms > 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 17:03:09.280490 [DEBUG] sofia.c:5283 Channel > sofia/internal/1000 at olegh.ath.cx > entering state [terminating][0] > 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2804 > (sofia/internal/1000 at olegh.ath.cx ) > Callstate Change ACTIVE -> HANGUP > 2011-11-02 17:03:09.280490 [NOTICE] sofia.c:6039 Hangup > sofia/internal/1000 at olegh.ath.cx > [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2820 Send signal > sofia/internal/1000 at olegh.ath.cx [KILL] > 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/1000 at olegh.ath.cx > [BREAK] > 2011-11-02 17:03:09.280490 [DEBUG] switch_ivr_play_say.c:1672 done > playing file > /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-please.wav > 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:348 waiting > for 4/4 > digits t/o 2000 > 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:395 digits '' > 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:585 IVR menu > 'demo_ivr' no input detected > 2011-11-02 17:03:10.587476 [DEBUG] switch_ivr_menu.c:599 exit-sound > 'voicemail/vm-goodbye.wav' > 2011-11-02 17:03:10.587476 [DEBUG] switch_core_session.c:2269 > sofia/internal/1000 at olegh.ath.cx skip > receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > > FS: voicemail > > 2011-11-02 16:51:50.217856 [DEBUG] switch_ivr_play_say.c:67 No > language > specified - Using [ru] > 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:244 Handle > say:[1290554584] (ru:ru) > 2011-11-02 16:51:50.307721 [INFO] mod_say_ru.c:721 ru_say!!! > 1290554584! say_opt.gender=0 say_opt.cases=0 > 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:1302 Codec > Activated L16 at 8000hz 1 channels 20ms > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 Send > signal > sofia/internal/1000 at olegh.ath.cx [BREAK] > 2011-11-02 16:51:50.990445 [DEBUG] sofia.c:5283 Channel > sofia/internal/1000 at olegh.ath.cx > entering state [terminating][0] > 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2804 > (sofia/internal/1000 at olegh.ath.cx ) > Callstate Change ACTIVE -> HANGUP > 2011-11-02 16:51:50.990445 [NOTICE] sofia.c:6039 Hangup > sofia/internal/1000 at olegh.ath.cx > [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2820 Send signal > sofia/internal/1000 at olegh.ath.cx [KILL] > 2011-11-02 16:51:50.990445 [DEBUG] switch_core_session.c:1177 Send > signal sofia/internal/1000 at olegh.ath.cx > [BREAK] > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111103/7e5158f4/attachment-0001.html From rml at tollfreeforwarding.com Thu Nov 3 23:38:45 2011 From: rml at tollfreeforwarding.com (RaviRaj Mulasa) Date: Thu, 3 Nov 2011 20:38:45 +0000 Subject: [Freeswitch-dev] Send Event/User Channel Variable from FreeSWITCH to RTMP client Message-ID: <8B94625BC339264DBA61E314BE9EC2CF230E95BE@EXCH125.IFN.com> Hi Please let us know how to send an Event/ User Channel Variable from FreeSWITCH to a RTMP client. Thanks RaviRaj. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111103/d7cd3f46/attachment.html From khovayko at gmail.com Fri Nov 4 06:23:20 2011 From: khovayko at gmail.com (Oleg Khovayko) Date: Thu, 03 Nov 2011 23:23:20 -0400 Subject: [Freeswitch-dev] Nortel 1535 drops connection to FS In-Reply-To: References: <10191907bb5b9103bb41ad1b7c8f0597@gw.ind.rwth-aachen.de> <4EB1B349.9010204@gmail.com> <4EB2EDC7.3080400@gmail.com> Message-ID: <4EB35AA8.8010900@gmail.com> Anthony, I found problem source. I think, it will be valuable information for our community. Also, maybe will need to fix FS according this info. 1. About "FS does not receive ACK" - you was right. Really, FS was not received ACKs, and breaks connection. 2. Currently, FS runs behind NAT router, with option "-nonat" and static routing. Also, there is firewall on the FS server, which allows connection to port 5060 only from some selected networks. Server has static IP=192.168.1.5, WAN IP=173.79.237.11, Phone connected to LAN, IP=192.168.1.158 (DHCP). 3. When I (and Stan G.) investigated this problem, we run tcpdump _inside_ the phone, and found -- phone sends ACKs to WAN ROUTERs port (173.79.237.11), when must send to FS server, 192.168.1.5 {{{ tcpdump -A -vvv port 5060 Via: SIP/2 20:36:27.790809 IP (tos 0xa0, ttl 64, id 28499, offset 0, flags [none], length: 699) lgvp.5060 > Wireless_Broadband_Router.5060: UDP, length: 671 E...oS.. at ..8.....O.p......%GACK sip:5000 at 173.79.237.11 20:36:27.809815 IP (tos 0xa0, ttl 64, id 28500, offset 0, flags [none], length: 699) lgvp.5060 > Wireless_Broadband_Router.5060: UDP, length: 671 E...oT.. at ..7.....O.p......%GACK sip:5000 at 173.79.237.11 20:36:27.815188 IP (tos 0xa0, ttl 64, id 28501, offset 0, flags [none], length: 699) lgvp.5060 > Wireless_Broadband_Router.5060: UDP, length: 671 E...oU.. at ..6.....O.p......%GACK sip:5000 at 173.79.237.11 20:36:29.179099 IP (tos 0x0, ttl 64, id 43922, offset 0, flags [none], length: 1174) 192.168.1.5.5060 > lgvp.5060: UDP, length: 1146 E....... at .F.................SIP/2.0 200 OK }} When router sends UDP packets to FS box, they receiving from Router's address 173.79.237.11, and rejected by IP firewall. I allowed my own WAN subnet 173.79.0.0/16 with firewall, and thereafter phone works OK. Following - sample of the FS log, when connection works OK, with ACKs. http://olegh.ath.cx/freeswitch.log-1535-2 But, question: How can I enforce phone works directly with FS, without loopback in the router? I assume, phone sends request to WAN address instead of direct FS, because FS sends to LAN client contact field, contains WAN address in the SIP 200 message: Contact: Of course, client sends replies back to this WAN address... How to fix this issue, for FS send to local clients only something like Contact: Thanks, Oleg Anthony Minessale wrote: > you are not getting an ACK back from the phone after the 200OK > investigate the sip traffic and look for something like SIP ALG or > other things that may cause the ACK to not make it back to FS. > > > 2011/11/3 Oleg Khovayko > > > Anthony, > > > Thank you very much for assistance. I produced log-file, contains > siptrace, you can get it: > > http://olegh.ath.cx/freeswitch.log-1535-1 > > In thins test, I called from numbed 1000 to number 5000. > > > Oleg > > > > > > > > > Anthony Minessale wrote: >> snippets of logs rarely are helpful, try complete debug logs >> including sip trace >> >> >> On Wed, Nov 2, 2011 at 4:16 PM, Oleg Khovayko > > wrote: >> >> Hi, >> >> I am using FS for couple years, it was works OK. >> And, approx 1 year ago, I tested with FreeSWITCH phone Nortel >> 1535. >> It worked perfect. >> >> Thereafter, I couple times upgraded FS, and right now, I run >> version >> from GIT, dated 11-10-29. OS: FreeBSD 7.2, 32 bits. >> >> I again connected Nortel 1535, and this time see, connection >> suddenly >> broken during call to any "robot": >> FS #5000, FS #4000, CallCentric #17771234567 >> >> Every time, before disconnect, I see chain of log messages like: >> >> [DEBUG] switch_core_session.c:872 Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> and HANGUP thereafter. See following samples from the log: >> >> Another phones, like VM1188T or PAP2, still works OK. >> About Nortel: I tested 3 phones, with two firmwares (2.50 & >> 2.76) -- >> problem same on both. >> >> Can you suggest me, how to fix this issue? >> >> Thanks, >> >> Oleg >> >> >> >> CallCentric: 7771234567 >> >> 2011-11-02 16:39:20.901219 [DEBUG] sofia_reg.c:1984 Changing >> expire time >> to 103 by request of proxy sip:callcentric.com >> >> 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 16:39:31.130669 [DEBUG] sofia.c:5283 Channel >> sofia/internal/1000 at olegh.ath.cx >> entering state [terminating][0] >> 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2804 >> (sofia/internal/1000 at olegh.ath.cx ) >> Callstate Change ACTIVE -> HANGUP >> 2011-11-02 16:39:31.130669 [NOTICE] sofia.c:6039 Hangup >> sofia/internal/1000 at olegh.ath.cx >> [CS_EXECUTE] [NORMAL_UNSPECIFIED] >> 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2820 Send >> signal >> sofia/internal/1000 at olegh.ath.cx >> [KILL] >> 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:1177 >> Send >> signal sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> >> >> FS: 5000 >> >> 2011-11-02 17:03:08.207852 [DEBUG] switch_ivr_play_say.c:1672 >> done >> playing file >> /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-to_do_a_freeswitch_echo_test.wav >> 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:244 >> Handle >> play-file:[ivr/ivr-please.wav] (ru:ru) >> 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:1302 >> Codec >> Activated L16 at 8000hz 1 channels 20ms >> 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 17:03:09.280490 [DEBUG] sofia.c:5283 Channel >> sofia/internal/1000 at olegh.ath.cx >> entering state [terminating][0] >> 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2804 >> (sofia/internal/1000 at olegh.ath.cx ) >> Callstate Change ACTIVE -> HANGUP >> 2011-11-02 17:03:09.280490 [NOTICE] sofia.c:6039 Hangup >> sofia/internal/1000 at olegh.ath.cx >> [CS_EXECUTE] [NORMAL_UNSPECIFIED] >> 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2820 Send >> signal >> sofia/internal/1000 at olegh.ath.cx >> [KILL] >> 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:1177 >> Send >> signal sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 17:03:09.280490 [DEBUG] switch_ivr_play_say.c:1672 >> done >> playing file >> /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-please.wav >> 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:348 >> waiting for 4/4 >> digits t/o 2000 >> 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:395 >> digits '' >> 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:585 IVR menu >> 'demo_ivr' no input detected >> 2011-11-02 17:03:10.587476 [DEBUG] switch_ivr_menu.c:599 >> exit-sound >> 'voicemail/vm-goodbye.wav' >> 2011-11-02 17:03:10.587476 [DEBUG] switch_core_session.c:2269 >> sofia/internal/1000 at olegh.ath.cx >> skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> >> >> FS: voicemail >> >> 2011-11-02 16:51:50.217856 [DEBUG] switch_ivr_play_say.c:67 >> No language >> specified - Using [ru] >> 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:244 >> Handle >> say:[1290554584] (ru:ru) >> 2011-11-02 16:51:50.307721 [INFO] mod_say_ru.c:721 ru_say!!! >> 1290554584! say_opt.gender=0 say_opt.cases=0 >> 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:1302 >> Codec >> Activated L16 at 8000hz 1 channels 20ms >> 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> 2011-11-02 16:51:50.990445 [DEBUG] sofia.c:5283 Channel >> sofia/internal/1000 at olegh.ath.cx >> entering state [terminating][0] >> 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2804 >> (sofia/internal/1000 at olegh.ath.cx ) >> Callstate Change ACTIVE -> HANGUP >> 2011-11-02 16:51:50.990445 [NOTICE] sofia.c:6039 Hangup >> sofia/internal/1000 at olegh.ath.cx >> [CS_EXECUTE] [NORMAL_UNSPECIFIED] >> 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2820 Send >> signal >> sofia/internal/1000 at olegh.ath.cx >> [KILL] >> 2011-11-02 16:51:50.990445 [DEBUG] switch_core_session.c:1177 >> Send >> signal sofia/internal/1000 at olegh.ath.cx >> [BREAK] >> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> 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_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111103/54db8cab/attachment-0001.html From khovayko at gmail.com Fri Nov 4 07:09:53 2011 From: khovayko at gmail.com (Oleg Khovayko) Date: Fri, 04 Nov 2011 00:09:53 -0400 Subject: [Freeswitch-dev] Nortel 1535 drops connection to FS In-Reply-To: References: <10191907bb5b9103bb41ad1b7c8f0597@gw.ind.rwth-aachen.de> <4EB1B349.9010204@gmail.com> Message-ID: <4EB36591.2000601@gmail.com> Stan suggested me to create additional SIP-profile, especial for LAN users. I think, this will solve the problem. Thanks to everyone, Oleg From tomp at tomp.co.uk Sat Nov 5 19:16:15 2011 From: tomp at tomp.co.uk (Tom Parrott) Date: Sat, 05 Nov 2011 16:16:15 +0000 Subject: [Freeswitch-dev] Cepstral locks up Message-ID: <4EB5614F.5090109@tomp.co.uk> Hi, I am using mod_cepstral to perform call announce features. On one of our servers the API call to run a TTS blocked, and stopped the call from proceeding until a 20 second delay had occurred. Additionally, when I ran /opt/swift/bin/swift -o test.wav "Hello World" The command would also block for 20 seconds, and not generate any audio. I ran reload mod_cepstra, to no avail. Eventually I restarted Freeswitch and it fixed the problem. Do you know what could cause this? I have a license purchase from Cepstral. Thanks Tom From freeswitch-list at puzzled.xs4all.nl Sun Nov 6 15:43:31 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Sun, 06 Nov 2011 13:43:31 +0100 Subject: [Freeswitch-dev] Cepstral locks up In-Reply-To: <4EB5614F.5090109@tomp.co.uk> References: <4EB5614F.5090109@tomp.co.uk> Message-ID: <4EB680F3.7070902@puzzled.xs4all.nl> On 11/05/2011 05:16 PM, Tom Parrott wrote: > Hi, > > I am using mod_cepstral to perform call announce features. > > On one of our servers the API call to run a TTS blocked, and stopped the > call from proceeding until a 20 second delay had occurred. > > Additionally, when I ran /opt/swift/bin/swift -o test.wav "Hello World" > > The command would also block for 20 seconds, and not generate any audio. > > I ran reload mod_cepstra, to no avail. Is this a x86_64 box? If so, iirc Brian K. West once mentioned that the 64bit version of Cepstral has/had issues. Bkw: of you read this, can you please share your findings? Regards, Patrick From anthony.minessale at gmail.com Sun Nov 6 19:24:06 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 6 Nov 2011 10:24:06 -0600 Subject: [Freeswitch-dev] Cepstral locks up In-Reply-To: <4EB680F3.7070902@puzzled.xs4all.nl> References: <4EB5614F.5090109@tomp.co.uk> <4EB680F3.7070902@puzzled.xs4all.nl> Message-ID: It only works with v4 the rest have deadlocks in the cepstral itself.... On Nov 6, 2011 6:47 AM, "Patrick Lists" wrote: > On 11/05/2011 05:16 PM, Tom Parrott wrote: > > Hi, > > > > I am using mod_cepstral to perform call announce features. > > > > On one of our servers the API call to run a TTS blocked, and stopped the > > call from proceeding until a 20 second delay had occurred. > > > > Additionally, when I ran /opt/swift/bin/swift -o test.wav "Hello World" > > > > The command would also block for 20 seconds, and not generate any audio. > > > > I ran reload mod_cepstra, to no avail. > > Is this a x86_64 box? If so, iirc Brian K. West once mentioned that the > 64bit version of Cepstral has/had issues. > > Bkw: of you read this, can you please share your findings? > > Regards, > Patrick > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111106/16c35589/attachment.html From freeswitch-list at puzzled.xs4all.nl Mon Nov 7 04:18:26 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Mon, 07 Nov 2011 02:18:26 +0100 Subject: [Freeswitch-dev] Cepstral locks up In-Reply-To: References: <4EB5614F.5090109@tomp.co.uk> <4EB680F3.7070902@puzzled.xs4all.nl> Message-ID: <4EB731E2.8030508@puzzled.xs4all.nl> On 11/06/2011 05:24 PM, Anthony Minessale wrote: > It only works with v4 the rest have deadlocks in the cepstral itself.... Thanks Anthony. Tom: I'm not very familiar with Cepstral but perhaps you could use UniMRCP between FreeSWITCH and Cepstral? More info at http://www.unimrcp.org/ Regards, Patrick From brian at freeswitch.org Mon Nov 7 04:32:16 2011 From: brian at freeswitch.org (Brian West) Date: Sun, 6 Nov 2011 19:32:16 -0600 Subject: [Freeswitch-dev] Cepstral locks up In-Reply-To: <4EB731E2.8030508@puzzled.xs4all.nl> References: <4EB5614F.5090109@tomp.co.uk> <4EB680F3.7070902@puzzled.xs4all.nl> <4EB731E2.8030508@puzzled.xs4all.nl> Message-ID: <0B30B426-E7A2-47A7-B53F-1F8C8C63905E@freeswitch.org> You'll have the same problem you'll just move the location of the problem outside of FreeSWITCH! :P /b On Nov 6, 2011, at 7:18 PM, Patrick Lists wrote: > On 11/06/2011 05:24 PM, Anthony Minessale wrote: >> It only works with v4 the rest have deadlocks in the cepstral itself.... > > Thanks Anthony. > > Tom: I'm not very familiar with Cepstral but perhaps you could use > UniMRCP between FreeSWITCH and Cepstral? > > More info at http://www.unimrcp.org/ > > Regards, > Patrick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111106/17eab017/attachment.html From msc at freeswitch.org Wed Nov 9 19:37:13 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Nov 2011 08:37:13 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: Hello FreeSWITCHers! Here's today's agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_09 We have a few miscellaneous items to discuss and I have some updates. Also, I'm hoping to continue the discussion on SIP 101 that we started last week. I think it would be a good time to go over the Contact: header and what it's for. That will set the ground work for what the NDLB-connectile-dysfunction parameter does. Talk to you soon, -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111109/1407ae35/attachment.html From trap at 2dayhost.com Fri Nov 4 00:39:46 2011 From: trap at 2dayhost.com (Max Machula) Date: Thu, 03 Nov 2011 21:39:46 +0000 Subject: [Freeswitch-dev] FreeSwitch date and time conditions Message-ID: <4EB30A22.1030800@2dayhost.com> Hello, Hope someone can help me. I have a strange problem with date and time on my FreeSwitch. I have dialplan: ... skip ... Server time: # date Thu Nov 3 21:34:22 GMT 2011 However FreeSwitch can not confirm the year 2011. Log file: Dialplan: sofia/internal/07711567890 at xx.xx.xx.xx parsing [public->WW26] continue=true Dialplan: sofia/internal/07711567890 at xx.xx.xx.xxRegex (PASS) [WW26] destination_number(26) =~ /26/ break=on-false Dialplan: sofia/internal/07711567890 at xx.xx.xx.xxRegex (FAIL) [WW26] year() =~ /2011/ break=on-false I get the same problem if I change field year to wday which set to 1-6 for example now. Any ideas would be much appreciated. Thanks, Max -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111103/65117b8e/attachment-0001.html From dujinfang at gmail.com Sat Nov 5 15:40:31 2011 From: dujinfang at gmail.com (Seven Du) Date: Sat, 5 Nov 2011 20:40:31 +0800 Subject: [Freeswitch-dev] subscribe to a gateway Message-ID: <4D8A5CE32F484AC3823353C948EF8E00@gmail.com> Hi, Can freeswitch as a client subscribe to a gateway(maybe another freeswitch) to get notifications? If not, is there any plan to do it or highlight on how hard to implement and where should I start from code? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111105/b8c06e16/attachment-0001.html From rob.kirkby at gmail.com Tue Nov 8 14:20:32 2011 From: rob.kirkby at gmail.com (Robert Kirkby) Date: Tue, 8 Nov 2011 11:20:32 +0000 Subject: [Freeswitch-dev] Asterisk / FreeSWITCH Developer required.. In-Reply-To: References: Message-ID: Hi We are looking for a developer / expert for an ongoing project. The project would include development as well as consultancy on support and hosting options. We are looking for a telephone solution with the features of a conference bridge, a simple outbound dialer with a large emphasis on call recording. The candidate will need a good knowledge of hosting Asterisk / FreeSWITCH , specifically in an environment where call quality is important, so knowledge of data centres with access to ISDN lines, not just VOIP. A full specification of the project will be provided after initial contact. Please email me on: rob.kirkby at gmail.com Thanks Rob Kirkby -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111108/1f19d23b/attachment-0001.html From daleiliu at gmail.com Wed Nov 9 11:21:47 2011 From: daleiliu at gmail.com (Dalei Liu) Date: Wed, 9 Nov 2011 00:21:47 -0800 Subject: [Freeswitch-dev] version numbers Message-ID: Hi, I've been using freeswitch for a while and really like the stability and sip compatibility. While the code is excellent, I think the version management could be better. >From the git repository, the last tag is 1.0.6 dated a year and half ago, and there was a 1.0.7 announcement but no tag anywhere. As described in the installation guide git repository is strongly recommended but the problem is, it's a changing target. And using SHA1 is really not helpful to the users. Is it possible to define a formal version numbering system and regular release procedure? For example, if we think the current master is stable enough has the features we need and no major issue, tag it as 1.1-rc1, release it and let the users to test it under all different conditions. And then tag 1.1-rc2, 1.1-rc3, etc until all the the minor issues are either fixed or moved to next minor release v1.2. At that moment, tag and release 1.1 and create a branch b1.1 on master. Later, only the bug fixes go into the branch and it can be tagged as v1.1.1, then 1.1.2, etc. All the new features or enhancements will sit in master and be included in the next minor release v1.2, which will follow the same procedure. In this way, the users will have a fixed target to test and report issues and the developers will have a way to reproduce the same problem. And it will help the users and developers from outside understand the development progress. I believe it will also help integrate freeswitch into mainstream distros, like debian and fedora. My dream is one day, I could just type apt-get install freeswitch from the fresh debian installation. If you think it worths the time, I could contribute some time on it. Best regards, Dalei From Matthew.Margolis at patlive.com Thu Nov 10 20:52:51 2011 From: Matthew.Margolis at patlive.com (Matthew Margolis) Date: Thu, 10 Nov 2011 12:52:51 -0500 Subject: [Freeswitch-dev] mod_say with a date Message-ID: When I try to use mod_say to speak a date, it never gets the 19XX's correct. It simply says it as a number such as "One thousand, nine hundred, ninety nine" as opposed to "Nineteen ninety nine." Am I doing something wrong? Or is this a bug? I am using a lua script: session:execute("say", "en-Callie current_date pronounced " .. [unix-timestamp]) Matthew Margolis Programmer PATLive T 850.422.2527 ext 74 E matthew.margolis at patlive.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111110/21654de9/attachment.html From anthony.minessale at gmail.com Fri Nov 11 01:46:26 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Nov 2011 16:46:26 -0600 Subject: [Freeswitch-dev] Asterisk / FreeSWITCH Developer required.. In-Reply-To: References: Message-ID: contact consulting at freeswitch.org for commercial consulting. On Tue, Nov 8, 2011 at 5:20 AM, Robert Kirkby wrote: > Hi > > > We are looking for a developer / expert for an ongoing project. > > > The project would include development as well as consultancy on support > and hosting options. > > > We are looking for a telephone solution with the features of a conference > bridge, a simple outbound dialer with a large emphasis on call recording. > > > The candidate will need a good knowledge of hosting Asterisk / FreeSWITCH > , specifically in an environment where call quality is important, so > knowledge of data centres with access to ISDN lines, not just VOIP. > > > A full specification of the project will be provided after initial contact. > > > Please email me on: rob.kirkby at gmail.com > > > Thanks > > > Rob Kirkby > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111110/f6056e09/attachment.html From mercutio.viz at gmail.com Fri Nov 11 01:58:12 2011 From: mercutio.viz at gmail.com (Michael Collins) Date: Thu, 10 Nov 2011 14:58:12 -0800 Subject: [Freeswitch-dev] ignore me Message-ID: test test From krice at freeswitch.org Fri Nov 11 02:06:20 2011 From: krice at freeswitch.org (Ken Rice) Date: Thu, 10 Nov 2011 17:06:20 -0600 Subject: [Freeswitch-dev] Testing 1 2 3 Message-ID: M Collins says ?TAP TAP TAP, Is this thing on??? K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111110/b2c85b5e/attachment.html From michal.bielicki at seventhsignal.de Fri Nov 11 02:11:30 2011 From: michal.bielicki at seventhsignal.de (Michal Bielicki) Date: Fri, 11 Nov 2011 00:11:30 +0100 Subject: [Freeswitch-dev] ignore me In-Reply-To: References: Message-ID: I always do :P Am 10.11.2011 um 23:58 schrieb Michael Collins: > test test > > _______________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org or http://www.freeswitchsoltions.com > > FreeSWITCH-powered IP PBX: The CudaTel > > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org Michal Bielicki Gesch?ftsf?hrer / CEO Seventh Signal Ltd. & Co. KG Weigandufer 45, B?ro 105, D-12059 Berlin Voice: +49 30 60988730 Amtsgericht Charlottenburg HRA 44413 B Ust.-ID: DE266981999 Gesch?ftsf?hrer: Michal Bielicki Pers?nlich Haftende Gesellschafterin: Seventh Signal Ltd, 69 Great Hampton St. Birmingham, B18 6EW, GB, Company Nr.: 06889439 WWW.: http://www.seventhsignal.de ---- From james at freedomnet.co.nz Fri Nov 11 02:13:15 2011 From: james at freedomnet.co.nz (James Jones) Date: Thu, 10 Nov 2011 18:13:15 -0500 Subject: [Freeswitch-dev] Testing 1 2 3 In-Reply-To: References: Message-ID: <4EBC5A8B.4000809@freedomnet.co.nz> We can frigging hear ya!!!! On 11/10/11 6:06 PM, Ken Rice wrote: > M Collins says "TAP TAP TAP, Is this thing on??" > > K > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111110/77c85d34/attachment-0001.html From anthony.minessale at gmail.com Fri Nov 11 02:25:53 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Nov 2011 17:25:53 -0600 Subject: [Freeswitch-dev] version numbers In-Reply-To: References: Message-ID: You are very right, we get no help in this department so we are slow about fixing it. 1.0.7 is the nick name for the development cycle. We have plans to start a new 1.2 branch soon and we will need stable 1.0 branch managers. We prefer not to produce burned releases because they get stale rather fast based on our development pace. On Wed, Nov 9, 2011 at 2:21 AM, Dalei Liu wrote: > Hi, > I've been using freeswitch for a while and really like the stability > and sip compatibility. While the code is excellent, I think the > version management could be better. > > >From the git repository, the last tag is 1.0.6 dated a year and half > ago, and there was a 1.0.7 announcement but no tag anywhere. As > described in the installation guide git repository is strongly > recommended but the problem is, it's a changing target. And using > SHA1 is really not helpful to the users. > > Is it possible to define a formal version numbering system and regular > release procedure? For example, if we think the current master is > stable enough has the features we need and no major issue, tag it as > 1.1-rc1, release it and let the users to test it under all different > conditions. And then tag 1.1-rc2, 1.1-rc3, etc until all the the > minor issues are either fixed or moved to next minor release v1.2. At > that moment, tag and release 1.1 and create a branch b1.1 on master. > Later, only the bug fixes go into the branch and it can be tagged as > v1.1.1, then 1.1.2, etc. All the new features or enhancements will > sit in master and be included in the next minor release v1.2, which > will follow the same procedure. > > In this way, the users will have a fixed target to test and report > issues and the developers will have a way to reproduce the same > problem. And it will help the users and developers from outside > understand the development progress. I believe it will also help > integrate freeswitch into mainstream distros, like debian and fedora. > My dream is one day, I could just type apt-get install freeswitch from > the fresh debian installation. > > If you think it worths the time, I could contribute some time on it. > > Best regards, > Dalei > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111110/d4a159aa/attachment.html From tomp at tomp.co.uk Fri Nov 11 03:00:47 2011 From: tomp at tomp.co.uk (Tom Parrott) Date: Fri, 11 Nov 2011 00:00:47 +0000 Subject: [Freeswitch-dev] Cepstral locks up In-Reply-To: <0B30B426-E7A2-47A7-B53F-1F8C8C63905E@freeswitch.org> References: <4EB5614F.5090109@tomp.co.uk> <4EB680F3.7070902@puzzled.xs4all.nl> <4EB731E2.8030508@puzzled.xs4all.nl> <0B30B426-E7A2-47A7-B53F-1F8C8C63905E@freeswitch.org> Message-ID: <4EBC65AF.5040403@tomp.co.uk> Hi, Thanks for all your replies, its great to know I am not suffering this issue alone. I have actually gotten around the issue, and improved our IVR response times by taking inspiration from mod_tts_commandline and writing a LUA script that can be activated as a phrase. It checks to see if the supplied text to speak is available as a cached wav file, and if not uses the cepstral swift command line to generate a new wav file in using an md5 hash of the text to speak. This solves both problems, namely the issue with mod_cepstral and the concurrency problems, and the response times of using any TTS engine to generate sounds. As long as most of your sounds are common amongst users (which ours are) then this simple LUA script ensures that the output is cached for future user. Let me know if you would like to see the script. Cheers Tom On 01/-10/-28163 08:59 PM, Brian West wrote: > You'll have the same problem you'll just move the location of the > problem outside of FreeSWITCH! :P > > /b > > On Nov 6, 2011, at 7:18 PM, Patrick Lists wrote: > >> On 11/06/2011 05:24 PM, Anthony Minessale wrote: >>> It only works with v4 the rest have deadlocks in the cepstral itself.... >> >> Thanks Anthony. >> >> Tom: I'm not very familiar with Cepstral but perhaps you could use >> UniMRCP between FreeSWITCH and Cepstral? >> >> More info athttp://www.unimrcp.org/ >> >> Regards, >> Patrick > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111111/774b6e38/attachment.html From freeswitch-list at puzzled.xs4all.nl Fri Nov 11 03:39:00 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Fri, 11 Nov 2011 01:39:00 +0100 Subject: [Freeswitch-dev] Cepstral locks up In-Reply-To: <4EBC65AF.5040403@tomp.co.uk> References: <4EB5614F.5090109@tomp.co.uk> <4EB680F3.7070902@puzzled.xs4all.nl> <4EB731E2.8030508@puzzled.xs4all.nl> <0B30B426-E7A2-47A7-B53F-1F8C8C63905E@freeswitch.org> <4EBC65AF.5040403@tomp.co.uk> Message-ID: <4EBC6EA4.5040406@puzzled.xs4all.nl> On 11/11/2011 01:00 AM, Tom Parrott wrote: [snip] > Let me know if you would like to see the script. Good to see you found a solution. If you don't mind sharing your script with the Community perhaps you could ask one of the developers to put it somewhere in contrib/ or give you access to your own branch in contrib/. Regards, Patrick From daleiliu at gmail.com Fri Nov 11 08:25:23 2011 From: daleiliu at gmail.com (Dalei Liu) Date: Thu, 10 Nov 2011 21:25:23 -0800 Subject: [Freeswitch-dev] version numbers In-Reply-To: References: Message-ID: Thanks for the reply. Here are some thoughts might be useful: 1. Create/update a wiki page about the definition of the version system; 2. Create wiki pages for the goals of each planned version, including stable and dev version, like 1.2, 1.3; 3. Tag the current master to be 1.0.8-rc1 or 1.1-rc1, and set an initial release date for the stable release, e.g Nov 30; 4. Create a branch only if we have to merge master with a major feature not on the goal list of the stable version; The goal for each version can be changed and the release date can be delayed. More RC versions can be released before the formal one. But the most important thing is, the users will have a version to test, to discuss and to fix. About the next release, I feel v1.1 is better than v1.0.8 because it saves a digit for all the following versions (v1.1.3 vs v1.0.8.3, etc). Personally, I'm not a big fan to the odd/even number style. I think all the features in master should be tested and stable enough for integration (otherwise it'd better stay in a branch as branching is very cheap in git). It just needs several beta and RC versions to make it stable. Using a whole minor number for testing is a waste. Again, if you think it worths the time, I would like to start contributing on wiki pages. From anthony.minessale at gmail.com Fri Nov 11 19:20:40 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Nov 2011 10:20:40 -0600 Subject: [Freeswitch-dev] version numbers In-Reply-To: References: Message-ID: The most important thing is to choose a way that works that disturbs me the least. I am open to work it out and you are welcome to participate but my basic principles need to stand. Beliefs I want to stick with: 1) The first number is the project version number, i.e. 1 since its the first complete implementation of a particular strategy. I would never see that version reach 2 unless I did a complete rewrite or changed my attack angle. apache 1.x is forking philosophy and 2.x is threading. 2) The second number is the major release number, each time it goes up its a new milestone. FreeSWITCH 1.0 was dubbed "Phoenix" because it was a new era being born. FreeSWITCH 1.2 will be "Dragon" since I am sticking with mythical beings as it seems. Since FreeSWITCH 1.2 might make drastic changes that other people do not want to endure, I want to permanently branch 1.x as a stable branch where small bugs can still be fixed and possibly pushed up. 3) I want to release branches of GIT rather than just tarballs because we need to be able to deploy quickly. I'm ok with rolling a tarball but people tend to get stuck on them and then we have to fight to get people to test new versions and we get all of our bug reports after its too late rather than before we mark it stable. 4) I do a huge portion of the coding on the project all by myself. So I don't have tons of resources to devote to anything too complicated. I have had volunteers in the past but nobody really seems to fully commit or stick around long enough so I need to make sure I do not let people implement their ideas then disappear and leave us stuck with it. You might want to hang out on IRC irc.freenode.net #freeswitch #freeswitch-dev or visit our weekly conf call sip:888 at conference.freeswitch.org every Wednesday 12-Npm CST On Thu, Nov 10, 2011 at 11:25 PM, Dalei Liu wrote: > Thanks for the reply. Here are some thoughts might be useful: > > 1. Create/update a wiki page about the definition of the version system; > 2. Create wiki pages for the goals of each planned version, including > stable and dev version, like 1.2, 1.3; > 3. Tag the current master to be 1.0.8-rc1 or 1.1-rc1, and set an > initial release date for the stable release, e.g Nov 30; > 4. Create a branch only if we have to merge master with a major > feature not on the goal list of the stable version; > > The goal for each version can be changed and the release date can be > delayed. More RC versions can be released before the formal one. But > the most important thing is, the users will have a version to test, to > discuss and to fix. > > About the next release, I feel v1.1 is better than v1.0.8 because it > saves a digit for all the following versions (v1.1.3 vs v1.0.8.3, > etc). > > Personally, I'm not a big fan to the odd/even number style. I think > all the features in master should be tested and stable enough for > integration (otherwise it'd better stay in a branch as branching is > very cheap in git). It just needs several beta and RC versions to > make it stable. Using a whole minor number for testing is a waste. > > Again, if you think it worths the time, I would like to start > contributing on wiki pages. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111111/817c26ec/attachment-0001.html From tomp at tomp.co.uk Fri Nov 11 20:10:23 2011 From: tomp at tomp.co.uk (Tom Parrott) Date: Fri, 11 Nov 2011 17:10:23 -0000 Subject: [Freeswitch-dev] Cepstral locks up In-Reply-To: References: Message-ID: Hi, I am happy to, although I'm sure it can be improved as this is my first lua script. Although its in production now and working like a charm. :) First, I created a phrase macro for requesting TTS, and then call the LUA script from there: Then this is the tts_cache.lua script: -- This script generates a wav file of the sentence passed in to it. -- It uses the Cepstral swift command to perform text-to-speech conversion. -- If the wav file already exists for this sentence, then it is not generated. -- Author: Thomas Parrott api = freeswitch.API(); msg = argv[1]; msgMd5 = api:execute( "md5", msg ); filename = '/var/lib/tts_cache/' .. msgMd5 .. '.wav'; cmd = '/opt/swift/bin/swift'; -- Set a channel variable so that we know which file to play back. session:setVariable( 'tts_file', filename ); -- Check whether the file already exists. file, errMsg = io.open( filename, "r" ) if not file then api:execute( 'system', cmd .. ' -o "' .. filename .. '" "' .. msg .. '"' ); end If I can have a branch into contrib I'll post it up there. Thanks Tom > On 11/11/2011 01:00 AM, Tom Parrott wrote: > [snip] >> Let me know if you would like to see the script. > > Good to see you found a solution. If you don't mind sharing your script > with the Community perhaps you could ask one of the developers to put it > somewhere in contrib/ or give you access to your own branch in contrib/. > > Regards, > Patrick > > > From msc at freeswitch.org Fri Nov 11 20:16:10 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Nov 2011 09:16:10 -0800 Subject: [Freeswitch-dev] Cepstral locks up In-Reply-To: References: Message-ID: Just throw this up on the wiki on the mod_cepstral page. -MC On Fri, Nov 11, 2011 at 9:10 AM, Tom Parrott wrote: > Hi, > > I am happy to, although I'm sure it can be improved as this is my first > lua script. Although its in production now and working like a charm. :) > > First, I created a phrase macro for requesting TTS, and then call the LUA > script from there: > > > > > > > > > > > > Then this is the tts_cache.lua script: > > -- This script generates a wav file of the sentence passed in to it. > -- It uses the Cepstral swift command to perform text-to-speech conversion. > -- If the wav file already exists for this sentence, then it is not > generated. > -- Author: Thomas Parrott > api = freeswitch.API(); > msg = argv[1]; > msgMd5 = api:execute( "md5", msg ); > filename = '/var/lib/tts_cache/' .. msgMd5 .. '.wav'; > cmd = '/opt/swift/bin/swift'; > > -- Set a channel variable so that we know which file to play back. > session:setVariable( 'tts_file', filename ); > > -- Check whether the file already exists. > file, errMsg = io.open( filename, "r" ) > if not file then > api:execute( 'system', cmd .. ' -o "' .. filename .. '" "' .. msg .. '"' ); > end > > If I can have a branch into contrib I'll post it up there. > > Thanks > Tom > > > On 11/11/2011 01:00 AM, Tom Parrott wrote: > > [snip] > >> Let me know if you would like to see the script. > > > > Good to see you found a solution. If you don't mind sharing your script > > with the Community perhaps you could ask one of the developers to put it > > somewhere in contrib/ or give you access to your own branch in contrib/. > > > > Regards, > > Patrick > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111111/da91d227/attachment.html From daleiliu at gmail.com Fri Nov 11 21:38:29 2011 From: daleiliu at gmail.com (Dalei Liu) Date: Fri, 11 Nov 2011 10:38:29 -0800 Subject: [Freeswitch-dev] version numbers In-Reply-To: References: Message-ID: Thanks for the inside thoughts. About 1) and 2), these are things I was looking for and I hope they may be put into wiki, so all the users and developers will understand it and stick on it. Some additional thoughts: a) What's the definition of the versions with odd number, like 1.1 and 1.3? Reserved for dev or same as even number? b) About the major releases, my personal opinion is, please try not to put all new fancy features into one version and release as early as possible once a feature is ,completed and tested. If a huge change affects all the modules and need significant time to develop and test, it might need a new project version number (major number). c) Stable releases with same major/minor number do not have to be strictly limited to bug fixes only. They may contain small improvements as far as the new version is a drop-in replacement to the old one, say, not breaking any existing installation. In this way, it will save some minor numbers for really big features. I believe in release early release often. To me, if I don't see a new version released in two years, most likely the project is dead. About 3), I'm a little confused about "release branches rather than tarballs" part. For me, each release has a version number and refers to a static point of the repository. A branch is just for developers to organize patches and can not be released to any one. I think if a software has easy installation and excellent compatibility, a user will have no problem to upgrade it. Anyway, if there is a bug, he has no way but to patch it. I think the best way to reach an audience is to build package binaries, deb, rpm and exe. About 4), I completely understand the concern and actually had similar feeling in some projects. The way I did/does is, I put everything to wiki if it doesn't fit a code comment. Everything is build-able by scripts and built automatically by cron. When there is a new comer, I ask him to review all the documents before he asks any questions, and add the answers to new questions to the documents. For anything new that affects other developers, there should be a document base upon the agreement, so it becomes an idea of the project instead of an idea of somebody. From anthony.minessale at gmail.com Sat Nov 12 01:46:07 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Nov 2011 16:46:07 -0600 Subject: [Freeswitch-dev] version numbers In-Reply-To: References: Message-ID: We probably will not be talking about the odd ones very much, I think we just do even for vanity sake. Yes releasing branches is our own radical idea that we want to implement. This basically means a stable branch. 1.0 branch will slow down to a crawl and people who are paranoid latch onto that and the community maintains it. 1.2 branch (HEAD) will continue business as usual. some day we then release 1.4 and 1.2 becomes a branch and 1.0 slows down even more. Then when they are older it's easier to make tarballs out of them. On Fri, Nov 11, 2011 at 12:38 PM, Dalei Liu wrote: > Thanks for the inside thoughts. > > About 1) and 2), these are things I was looking for and I hope they > may be put into wiki, so all the users and developers will understand > it and stick on it. > > Some additional thoughts: > a) What's the definition of the versions with odd number, like 1.1 and > 1.3? Reserved for dev or same as even number? > > b) About the major releases, my personal opinion is, please try not to > put all new fancy features into one version and release as early as > possible once a feature is ,completed and tested. If a huge change > affects all the modules and need significant time to develop and test, > it might need a new project version number (major number). > > c) Stable releases with same major/minor number do not have to be > strictly limited to bug fixes only. They may contain small > improvements as far as the new version is a drop-in replacement to the > old one, say, not breaking any existing installation. In this way, it > will save some minor numbers for really big features. > > I believe in release early release often. To me, if I don't see a new > version released in two years, most likely the project is dead. > > About 3), I'm a little confused about "release branches rather than > tarballs" part. For me, each release has a version number and refers > to a static point of the repository. A branch is just for developers > to organize patches and can not be released to any one. I think if a > software has easy installation and excellent compatibility, a user > will have no problem to upgrade it. Anyway, if there is a bug, he has > no way but to patch it. I think the best way to reach an audience is > to build package binaries, deb, rpm and exe. > > About 4), I completely understand the concern and actually had similar > feeling in some projects. The way I did/does is, I put everything to > wiki if it doesn't fit a code comment. Everything is build-able by > scripts and built automatically by cron. When there is a new comer, I > ask him to review all the documents before he asks any questions, and > add the answers to new questions to the documents. For anything new > that affects other developers, there should be a document base upon > the agreement, so it becomes an idea of the project instead of an idea > of somebody. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111111/59a9c0e8/attachment.html From daleiliu at gmail.com Sat Nov 12 09:45:07 2011 From: daleiliu at gmail.com (Dalei Liu) Date: Fri, 11 Nov 2011 22:45:07 -0800 Subject: [Freeswitch-dev] version numbers In-Reply-To: References: Message-ID: Sound great. As everything is planned, do you have a rough estimate of the code need to be done before freeze v1.0? And what's feature list of project dragon? Any target release date for both projects? From tomp at tomp.co.uk Sat Nov 12 17:30:58 2011 From: tomp at tomp.co.uk (Tom Parrott) Date: Sat, 12 Nov 2011 14:30:58 +0000 Subject: [Freeswitch-dev] version numbers In-Reply-To: References: Message-ID: <4EBE8322.7040802@tomp.co.uk> Anthony, I would be interested to know how you and the other Freeswitch developers go about deploying new 'versions' of the code. Traditionally Linux uses various flavours of package management, such as RPM and APT that takes an upstream source tarball, combines it with patches and any other vendor supplies changes and then makes a package that can be installed on multiple servers. At Infinity Tracking we have been using Freeswitch in production for over 6 months now and it has been extremely reliable. We run several servers in a cluster, so it is important that we can manage code deployments effectively. We use RPM and Puppet to ensure consistent package versions. With every other piece of software in the stack we use; MySQL, PHP, Apache, Nginx, Memcached, Redis to name a few they are released as tarballs or at least versioned tags in git/svn. This enables us to pull down either a tarball or extract a specific tag version and from there build a package with a version number. With Freeswitch we have to 'git pull' the latest head, and then increment our build number, e.g. 1.0.7-12, however our version number of the package is internal to us, and so is of no use to other people for reporting bugs against. I understand your concerns about the rate of development and how you don't want to get people stuck on older versions, and we don't expect a tagged version to instantly deployable without internal testing ourselves, just like git head wouldn't be. I was wondering is there any possibility of having something like a 'nightly' build version that is tagged in git that people could use as a common reference point? Thanks Tom From anthony.minessale at gmail.com Sat Nov 12 20:52:26 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 12 Nov 2011 11:52:26 -0600 Subject: [Freeswitch-dev] version numbers In-Reply-To: References: Message-ID: On Sat, Nov 12, 2011 at 12:45 AM, Dalei Liu wrote: > Sound great. As everything is planned, do you have a rough estimate > of the code need to be done before freeze v1.0? And what's feature > list of project dragon? Any target release date for both projects? > > We don't really plan too far ahead so far. It's probably a good idea to get into at some point but we haven't thus far. The one major architecture idea we have been considering for a while is a new kind of controller module, something that could plug into the concept of using signalling on a controller box to command the media resources on another but that's been an idea for years now ;) Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111112/d90edf86/attachment.html From anthony.minessale at gmail.com Sat Nov 12 20:55:24 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 12 Nov 2011 11:55:24 -0600 Subject: [Freeswitch-dev] version numbers In-Reply-To: <4EBE8322.7040802@tomp.co.uk> References: <4EBE8322.7040802@tomp.co.uk> Message-ID: On Sat, Nov 12, 2011 at 8:30 AM, Tom Parrott wrote: > Anthony, > > I would be interested to know how you and the other Freeswitch > developers go about deploying new 'versions' of the code. > > > Right now we make a new version every time we commit and that's about it. We currently have a policy to never intentionally leave our git HEAD in an unusable state. This is hard to do naturally so that's why we have the idea to create 2 branches where one is only updated with important changes etc. None if this is written in stone, I am just waiting for it all to fall into place the most sensible way so we can keep things moving. I will say a new release is eminent, I just need to find the resources to pull everything together. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111112/2fbf109c/attachment.html From sz.krisz at freemail.hu Sat Nov 12 18:56:24 2011 From: sz.krisz at freemail.hu (=?ISO-8859-2?Q?Szentesi_Kriszti=E1n?=) Date: Sat, 12 Nov 2011 16:56:24 +0100 (CET) Subject: [Freeswitch-dev] G.729AB License In-Reply-To: Message-ID: Hello, any news with the windows version of G.729 transcoder? Any support needed for that? Thanks, Krisztian -----Original Message----- From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Friday, June 18, 2010 8:14 PM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] G.729AB License I am going to try to work on that a bit this weekend. Drop me a line directly next week, and we'll see if we can test this out. Mike On Jun 17, 2010, at 4:53 PM, Bob Coleman wrote: > Hi, > > Any news on the windows version? Progress? > _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111112/2c77acbf/attachment.html From bhrugumehta at gmail.com Mon Nov 14 09:18:12 2011 From: bhrugumehta at gmail.com (Bhrugu Mehta) Date: Mon, 14 Nov 2011 11:48:12 +0530 Subject: [Freeswitch-dev] autodialer Message-ID: HI, all Is there any way to create autodialer in freeswitch. I am new to freeswitch. Thanks and regards, -- Bhrugu Mehta Sr. S/W Engineer VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) India -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111114/f931d10b/attachment.html From msc at freeswitch.org Mon Nov 14 20:10:49 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Nov 2011 09:10:49 -0800 Subject: [Freeswitch-dev] autodialer In-Reply-To: References: Message-ID: On Sun, Nov 13, 2011 at 10:18 PM, Bhrugu Mehta wrote: > HI, all > > Is there any way to create autodialer in freeswitch. I am new to > freeswitch. > Yes, there is. ;) -MC > > Thanks and regards, > > -- > Bhrugu Mehta > Sr. S/W Engineer > VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) > India > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111114/62a82db5/attachment.html From fdelawarde at wirelessmundi.com Mon Nov 14 20:30:27 2011 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Mon, 14 Nov 2011 18:30:27 +0100 Subject: [Freeswitch-dev] How can I detect the cause of a potential memory leak? Message-ID: <1321291827.20161.66.camel@luna.madrid.commsmundi.com> Hello, I'm experiencing what I think is a memory leak in a production system with a recent git (2-Nov), with FS consuming >5GB and increasing after a few days with never more than 10 calls at once. While the users are not experiencing problems so far, I'm a bit worried. So I have a few questions: - How can I be sure it's a leak and not some memory pool thing that FS would free when the system needs (system has 10GB total)? - If I unload modules one by one, is the memory used by this module freed immediately? - What would be a "good" way to try and narrow down the cause? Is valgrind a good tool for that? Of course I'll try with GIT HEAD and try my best to find the cause before considering to add a new Jira issue, otherwise it's quite useless. Thanks, Fran?ois. From anthony.minessale at gmail.com Mon Nov 14 20:41:07 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Nov 2011 11:41:07 -0600 Subject: [Freeswitch-dev] How can I detect the cause of a potential memory leak? In-Reply-To: <1321291827.20161.66.camel@luna.madrid.commsmundi.com> References: <1321291827.20161.66.camel@luna.madrid.commsmundi.com> Message-ID: valgrind is your best bet: valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes /usr/local/freeswitch/bin/freeswitch -vg run this on normal traffic for a while and get the log file. if you unload mods it would not help with a leak but it would with a swelling pool. On Mon, Nov 14, 2011 at 11:30 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > Hello, > > I'm experiencing what I think is a memory leak in a production system > with a recent git (2-Nov), with FS consuming >5GB and increasing after a > few days with never more than 10 calls at once. > > While the users are not experiencing problems so far, I'm a bit worried. > So I have a few questions: > > - How can I be sure it's a leak and not some memory pool thing that FS > would free when the system needs (system has 10GB total)? > > - If I unload modules one by one, is the memory used by this module > freed immediately? > > - What would be a "good" way to try and narrow down the cause? Is > valgrind a good tool for that? > > Of course I'll try with GIT HEAD and try my best to find the cause > before considering to add a new Jira issue, otherwise it's quite > useless. > > > Thanks, > Fran?ois. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111114/4afc595d/attachment-0001.html From bhrugumehta at gmail.com Tue Nov 15 08:13:19 2011 From: bhrugumehta at gmail.com (Bhrugu Mehta) Date: Tue, 15 Nov 2011 10:43:19 +0530 Subject: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 65, Issue 12 In-Reply-To: References: Message-ID: ok thnks for reply for autodialer. but how to create this in freeswitch. Any starting point of this matter. suggest me.. Regards, 2011/11/14 > Send FreeSWITCH-dev mailing list submissions to > freeswitch-dev at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > or, via email, send a message with subject or body 'help' to > freeswitch-dev-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-dev-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-dev digest..." > > Today's Topics: > > 1. Re: version numbers (Tom Parrott) > 2. Re: version numbers (Anthony Minessale) > 3. Re: version numbers (Anthony Minessale) > 4. Re: G.729AB License (Szentesi Kriszti?n) > 5. autodialer (Bhrugu Mehta) > 6. Re: autodialer (Michael Collins) > 7. How can I detect the cause of a potential memory leak? > (Fran?ois Delawarde) > 8. Re: How can I detect the cause of a potential memory leak? > (Anthony Minessale) > > > ---------- Forwarded message ---------- > From: Tom Parrott > To: freeswitch-dev at lists.freeswitch.org > Date: Sat, 12 Nov 2011 14:30:58 +0000 > Subject: Re: [Freeswitch-dev] version numbers > Anthony, > > I would be interested to know how you and the other Freeswitch developers > go about deploying new 'versions' of the code. > > Traditionally Linux uses various flavours of package management, such as > RPM and APT that takes an upstream source tarball, combines it with patches > and any other vendor supplies changes and then makes a package that can be > installed on multiple servers. > > At Infinity Tracking we have been using Freeswitch in production for over > 6 months now and it has been extremely reliable. > > We run several servers in a cluster, so it is important that we can manage > code deployments effectively. > > We use RPM and Puppet to ensure consistent package versions. > > With every other piece of software in the stack we use; MySQL, PHP, > Apache, Nginx, Memcached, Redis to name a few they are released as tarballs > or at least versioned tags in git/svn. > > This enables us to pull down either a tarball or extract a specific tag > version and from there build a package with a version number. > > With Freeswitch we have to 'git pull' the latest head, and then increment > our build number, e.g. 1.0.7-12, however our version number of the package > is internal to us, and so is of no use to other people for reporting bugs > against. > > I understand your concerns about the rate of development and how you don't > want to get people stuck on older versions, and we don't expect a tagged > version to instantly deployable without internal testing ourselves, just > like git head wouldn't be. > > I was wondering is there any possibility of having something like a > 'nightly' build version that is tagged in git that people could use as a > common reference point? > > Thanks > Tom > > > > > > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: freeswitch-dev at lists.freeswitch.org > Date: Sat, 12 Nov 2011 11:52:26 -0600 > Subject: Re: [Freeswitch-dev] version numbers > > > On Sat, Nov 12, 2011 at 12:45 AM, Dalei Liu wrote: > >> Sound great. As everything is planned, do you have a rough estimate >> of the code need to be done before freeze v1.0? And what's feature >> list of project dragon? Any target release date for both projects? >> >> > We don't really plan too far ahead so far. It's probably a good idea to > get into at some point but we haven't thus far. > The one major architecture idea we have been considering for a while is a > new kind of controller module, something that could plug into the concept > of using signalling on a controller box to command the media resources on > another but that's been an idea for years now ;) > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: freeswitch-dev at lists.freeswitch.org > Date: Sat, 12 Nov 2011 11:55:24 -0600 > Subject: Re: [Freeswitch-dev] version numbers > > > On Sat, Nov 12, 2011 at 8:30 AM, Tom Parrott wrote: > >> Anthony, >> >> I would be interested to know how you and the other Freeswitch >> developers go about deploying new 'versions' of the code. >> >> >> > > Right now we make a new version every time we commit and that's about it. > We currently have a policy to never intentionally leave our git HEAD in an > unusable state. > > This is hard to do naturally so that's why we have the idea to create 2 > branches where one is only updated with important changes etc. > > None if this is written in stone, I am just waiting for it all to fall > into place the most sensible way so we can keep things moving. > > I will say a new release is eminent, I just need to find the resources to > pull everything together. > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > ---------- Forwarded message ---------- > From: "Szentesi Kriszti?n" > To: freeswitch-dev at lists.freeswitch.org > Date: Sat, 12 Nov 2011 16:56:24 +0100 (CET) > Subject: Re: [Freeswitch-dev] G.729AB License > Hello, > > any news with the windows version of G.729 transcoder? Any support needed > for that? > > Thanks, > Krisztian > > -----Original Message----- > From: freeswitch-dev-bounces at lists.freeswitch.org > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael > Jerris > Sent: Friday, June 18, 2010 8:14 PM > To: freeswitch-dev at lists.freeswitch.org > Subject: Re: [Freeswitch-dev] G.729AB License > > I am going to try to work on that a bit this weekend. Drop me a line > directly next week, and we'll see if we can test this out. > > Mike > > On Jun 17, 2010, at 4:53 PM, Bob Coleman wrote: > > > Hi, > > > > Any news on the windows version? Progress? > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > ---------- Forwarded message ---------- > From: Bhrugu Mehta > To: freeswitch-dev at lists.freeswitch.org > Date: Mon, 14 Nov 2011 11:48:12 +0530 > Subject: [Freeswitch-dev] autodialer > HI, all > > Is there any way to create autodialer in freeswitch. I am new to > freeswitch. > > Thanks and regards, > > -- > Bhrugu Mehta > Sr. S/W Engineer > VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) > India > > > > ---------- Forwarded message ---------- > From: Michael Collins > To: freeswitch-dev at lists.freeswitch.org > Date: Mon, 14 Nov 2011 09:10:49 -0800 > Subject: Re: [Freeswitch-dev] autodialer > > > On Sun, Nov 13, 2011 at 10:18 PM, Bhrugu Mehta wrote: > >> HI, all >> >> Is there any way to create autodialer in freeswitch. I am new to >> freeswitch. >> > Yes, there is. ;) > -MC > > >> >> Thanks and regards, >> >> -- >> Bhrugu Mehta >> Sr. S/W Engineer >> VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) >> India >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: Fran?ois Delawarde > To: freeswitch-dev at lists.freeswitch.org > Date: Mon, 14 Nov 2011 18:30:27 +0100 > Subject: [Freeswitch-dev] How can I detect the cause of a potential memory > leak? > Hello, > > I'm experiencing what I think is a memory leak in a production system > with a recent git (2-Nov), with FS consuming >5GB and increasing after a > few days with never more than 10 calls at once. > > While the users are not experiencing problems so far, I'm a bit worried. > So I have a few questions: > > - How can I be sure it's a leak and not some memory pool thing that FS > would free when the system needs (system has 10GB total)? > > - If I unload modules one by one, is the memory used by this module > freed immediately? > > - What would be a "good" way to try and narrow down the cause? Is > valgrind a good tool for that? > > Of course I'll try with GIT HEAD and try my best to find the cause > before considering to add a new Jira issue, otherwise it's quite > useless. > > > Thanks, > Fran?ois. > > > > > > ---------- Forwarded message ---------- > From: Anthony Minessale > To: freeswitch-dev at lists.freeswitch.org > Date: Mon, 14 Nov 2011 11:41:07 -0600 > Subject: Re: [Freeswitch-dev] How can I detect the cause of a potential > memory leak? > > valgrind is your best bet: > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full > --leak-resolution=high --show-reachable=yes > /usr/local/freeswitch/bin/freeswitch -vg > > > run this on normal traffic for a while and get the log file. > > > if you unload mods it would not help with a leak but it would with a > swelling pool. > > > > On Mon, Nov 14, 2011 at 11:30 AM, Fran?ois Delawarde < > fdelawarde at wirelessmundi.com> wrote: > >> Hello, >> >> I'm experiencing what I think is a memory leak in a production system >> with a recent git (2-Nov), with FS consuming >5GB and increasing after a >> few days with never more than 10 calls at once. >> >> While the users are not experiencing problems so far, I'm a bit worried. >> So I have a few questions: >> >> - How can I be sure it's a leak and not some memory pool thing that FS >> would free when the system needs (system has 10GB total)? >> >> - If I unload modules one by one, is the memory used by this module >> freed immediately? >> >> - What would be a "good" way to try and narrow down the cause? Is >> valgrind a good tool for that? >> >> Of course I'll try with GIT HEAD and try my best to find the cause >> before considering to add a new Jira issue, otherwise it's quite >> useless. >> >> >> Thanks, >> Fran?ois. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Bhrugu Mehta Sr. S/W Engineer VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) India -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111115/83760148/attachment-0001.html From mitul at enterux.com Tue Nov 15 08:26:54 2011 From: mitul at enterux.com (Mitul Limbani) Date: Tue, 15 Nov 2011 10:56:54 +0530 Subject: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 65, Issue 12 In-Reply-To: References: Message-ID: Bhrugu, Its very similar to how you would write an autodialer in ast. Open a Manager Control and Originate calls. Same, open a FS Event Socket : http://wiki.freeswitch.org/wiki/Event_Socket I am not sure if you did any of your homework read the above page to know what you may need to write code to originate a call and control it. Regards, Mitul Limbani Enterux Solutions, www.enterux.com On Tue, Nov 15, 2011 at 10:43 AM, Bhrugu Mehta wrote: > ok thnks for reply for autodialer. > > but how to create this in freeswitch. Any starting point of this matter. > suggest me.. > > Regards, > > 2011/11/14 > >> Send FreeSWITCH-dev mailing list submissions to >> freeswitch-dev at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> or, via email, send a message with subject or body 'help' to >> freeswitch-dev-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-dev-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-dev digest..." >> >> Today's Topics: >> >> 1. Re: version numbers (Tom Parrott) >> 2. Re: version numbers (Anthony Minessale) >> 3. Re: version numbers (Anthony Minessale) >> 4. Re: G.729AB License (Szentesi Kriszti?n) >> 5. autodialer (Bhrugu Mehta) >> 6. Re: autodialer (Michael Collins) >> 7. How can I detect the cause of a potential memory leak? >> (Fran?ois Delawarde) >> 8. Re: How can I detect the cause of a potential memory leak? >> (Anthony Minessale) >> >> >> ---------- Forwarded message ---------- >> From: Tom Parrott >> To: freeswitch-dev at lists.freeswitch.org >> Date: Sat, 12 Nov 2011 14:30:58 +0000 >> Subject: Re: [Freeswitch-dev] version numbers >> Anthony, >> >> I would be interested to know how you and the other Freeswitch developers >> go about deploying new 'versions' of the code. >> >> Traditionally Linux uses various flavours of package management, such as >> RPM and APT that takes an upstream source tarball, combines it with patches >> and any other vendor supplies changes and then makes a package that can be >> installed on multiple servers. >> >> At Infinity Tracking we have been using Freeswitch in production for over >> 6 months now and it has been extremely reliable. >> >> We run several servers in a cluster, so it is important that we can >> manage code deployments effectively. >> >> We use RPM and Puppet to ensure consistent package versions. >> >> With every other piece of software in the stack we use; MySQL, PHP, >> Apache, Nginx, Memcached, Redis to name a few they are released as tarballs >> or at least versioned tags in git/svn. >> >> This enables us to pull down either a tarball or extract a specific tag >> version and from there build a package with a version number. >> >> With Freeswitch we have to 'git pull' the latest head, and then increment >> our build number, e.g. 1.0.7-12, however our version number of the package >> is internal to us, and so is of no use to other people for reporting bugs >> against. >> >> I understand your concerns about the rate of development and how you >> don't want to get people stuck on older versions, and we don't expect a >> tagged version to instantly deployable without internal testing ourselves, >> just like git head wouldn't be. >> >> I was wondering is there any possibility of having something like a >> 'nightly' build version that is tagged in git that people could use as a >> common reference point? >> >> Thanks >> Tom >> >> >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale >> To: freeswitch-dev at lists.freeswitch.org >> Date: Sat, 12 Nov 2011 11:52:26 -0600 >> Subject: Re: [Freeswitch-dev] version numbers >> >> >> On Sat, Nov 12, 2011 at 12:45 AM, Dalei Liu wrote: >> >>> Sound great. As everything is planned, do you have a rough estimate >>> of the code need to be done before freeze v1.0? And what's feature >>> list of project dragon? Any target release date for both projects? >>> >>> >> We don't really plan too far ahead so far. It's probably a good idea to >> get into at some point but we haven't thus far. >> The one major architecture idea we have been considering for a while is a >> new kind of controller module, something that could plug into the concept >> of using signalling on a controller box to command the media resources on >> another but that's been an idea for years now ;) >> >> >> >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale >> To: freeswitch-dev at lists.freeswitch.org >> Date: Sat, 12 Nov 2011 11:55:24 -0600 >> Subject: Re: [Freeswitch-dev] version numbers >> >> >> On Sat, Nov 12, 2011 at 8:30 AM, Tom Parrott wrote: >> >>> Anthony, >>> >>> I would be interested to know how you and the other Freeswitch >>> developers go about deploying new 'versions' of the code. >>> >>> >>> >> >> Right now we make a new version every time we commit and that's about it. >> We currently have a policy to never intentionally leave our git HEAD in >> an unusable state. >> >> This is hard to do naturally so that's why we have the idea to create 2 >> branches where one is only updated with important changes etc. >> >> None if this is written in stone, I am just waiting for it all to fall >> into place the most sensible way so we can keep things moving. >> >> I will say a new release is eminent, I just need to find the resources to >> pull everything together. >> >> >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> ---------- Forwarded message ---------- >> From: "Szentesi Kriszti?n" >> To: freeswitch-dev at lists.freeswitch.org >> Date: Sat, 12 Nov 2011 16:56:24 +0100 (CET) >> Subject: Re: [Freeswitch-dev] G.729AB License >> Hello, >> >> any news with the windows version of G.729 transcoder? Any support needed >> for that? >> >> Thanks, >> Krisztian >> >> -----Original Message----- >> From: freeswitch-dev-bounces at lists.freeswitch.org >> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of >> Michael Jerris >> Sent: Friday, June 18, 2010 8:14 PM >> To: freeswitch-dev at lists.freeswitch.org >> Subject: Re: [Freeswitch-dev] G.729AB License >> >> I am going to try to work on that a bit this weekend. Drop me a line >> directly next week, and we'll see if we can test this out. >> >> Mike >> >> On Jun 17, 2010, at 4:53 PM, Bob Coleman wrote: >> >> > Hi, >> > >> > Any news on the windows version? Progress? >> > >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> >> ---------- Forwarded message ---------- >> From: Bhrugu Mehta >> To: freeswitch-dev at lists.freeswitch.org >> Date: Mon, 14 Nov 2011 11:48:12 +0530 >> Subject: [Freeswitch-dev] autodialer >> HI, all >> >> Is there any way to create autodialer in freeswitch. I am new to >> freeswitch. >> >> Thanks and regards, >> >> -- >> Bhrugu Mehta >> Sr. S/W Engineer >> VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) >> India >> >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins >> To: freeswitch-dev at lists.freeswitch.org >> Date: Mon, 14 Nov 2011 09:10:49 -0800 >> Subject: Re: [Freeswitch-dev] autodialer >> >> >> On Sun, Nov 13, 2011 at 10:18 PM, Bhrugu Mehta wrote: >> >>> HI, all >>> >>> Is there any way to create autodialer in freeswitch. I am new to >>> freeswitch. >>> >> Yes, there is. ;) >> -MC >> >> >>> >>> Thanks and regards, >>> >>> -- >>> Bhrugu Mehta >>> Sr. S/W Engineer >>> VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) >>> India >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >>> >> >> >> ---------- Forwarded message ---------- >> From: Fran?ois Delawarde >> To: freeswitch-dev at lists.freeswitch.org >> Date: Mon, 14 Nov 2011 18:30:27 +0100 >> Subject: [Freeswitch-dev] How can I detect the cause of a potential >> memory leak? >> Hello, >> >> I'm experiencing what I think is a memory leak in a production system >> with a recent git (2-Nov), with FS consuming >5GB and increasing after a >> few days with never more than 10 calls at once. >> >> While the users are not experiencing problems so far, I'm a bit worried. >> So I have a few questions: >> >> - How can I be sure it's a leak and not some memory pool thing that FS >> would free when the system needs (system has 10GB total)? >> >> - If I unload modules one by one, is the memory used by this module >> freed immediately? >> >> - What would be a "good" way to try and narrow down the cause? Is >> valgrind a good tool for that? >> >> Of course I'll try with GIT HEAD and try my best to find the cause >> before considering to add a new Jira issue, otherwise it's quite >> useless. >> >> >> Thanks, >> Fran?ois. >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale >> To: freeswitch-dev at lists.freeswitch.org >> Date: Mon, 14 Nov 2011 11:41:07 -0600 >> Subject: Re: [Freeswitch-dev] How can I detect the cause of a potential >> memory leak? >> >> valgrind is your best bet: >> >> valgrind --tool=memcheck --log-file=vg.log --leak-check=full >> --leak-resolution=high --show-reachable=yes >> /usr/local/freeswitch/bin/freeswitch -vg >> >> >> run this on normal traffic for a while and get the log file. >> >> >> if you unload mods it would not help with a leak but it would with a >> swelling pool. >> >> >> >> On Mon, Nov 14, 2011 at 11:30 AM, Fran?ois Delawarde < >> fdelawarde at wirelessmundi.com> wrote: >> >>> Hello, >>> >>> I'm experiencing what I think is a memory leak in a production system >>> with a recent git (2-Nov), with FS consuming >5GB and increasing after a >>> few days with never more than 10 calls at once. >>> >>> While the users are not experiencing problems so far, I'm a bit worried. >>> So I have a few questions: >>> >>> - How can I be sure it's a leak and not some memory pool thing that FS >>> would free when the system needs (system has 10GB total)? >>> >>> - If I unload modules one by one, is the memory used by this module >>> freed immediately? >>> >>> - What would be a "good" way to try and narrow down the cause? Is >>> valgrind a good tool for that? >>> >>> Of course I'll try with GIT HEAD and try my best to find the cause >>> before considering to add a new Jira issue, otherwise it's quite >>> useless. >>> >>> >>> Thanks, >>> Fran?ois. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > > -- > Bhrugu Mehta > Sr. S/W Engineer > VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) > India > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111115/8959e09b/attachment-0001.html From fdelawarde at wirelessmundi.com Tue Nov 15 12:25:16 2011 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 15 Nov 2011 10:25:16 +0100 Subject: [Freeswitch-dev] How can I detect the cause of a potential memory leak? In-Reply-To: References: <1321291827.20161.66.camel@luna.madrid.commsmundi.com> Message-ID: <1321349116.20161.101.camel@luna.madrid.commsmundi.com> I did a small valgrind test with git HEAD: 1. turn on, make 5 calls or so 2. leave FS on for the night (without calls) 3. shut it down cleanly in the morning During execution, some "Warning: invalid file descriptor -1 in syscall close()" (4-5 for each call + some extras). After shutting down, something strikes me in the vg.log file: "definitely lost: 173,407,885 bytes in 84,680 blocks" Is this normal? valgrind log: http://pastebin.freeswitch.org/17779 Thanks, Fran?ois. On Mon, 2011-11-14 at 11:41 -0600, Anthony Minessale wrote: > valgrind is your best bet: > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full > --leak-resolution=high > --show-reachable=yes /usr/local/freeswitch/bin/freeswitch -vg > > > run this on normal traffic for a while and get the log file. > > > if you unload mods it would not help with a leak but it would with a > swelling pool. > > > > On Mon, Nov 14, 2011 at 11:30 AM, Fran?ois Delawarde > wrote: > Hello, > > I'm experiencing what I think is a memory leak in a production > system > with a recent git (2-Nov), with FS consuming >5GB and > increasing after a > few days with never more than 10 calls at once. > > While the users are not experiencing problems so far, I'm a > bit worried. > So I have a few questions: > > - How can I be sure it's a leak and not some memory pool thing > that FS > would free when the system needs (system has 10GB total)? > > - If I unload modules one by one, is the memory used by this > module > freed immediately? > > - What would be a "good" way to try and narrow down the cause? > Is > valgrind a good tool for that? > > Of course I'll try with GIT HEAD and try my best to find the > cause > before considering to add a new Jira issue, otherwise it's > quite > useless. > > > Thanks, > Fran?ois. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org From anthony.minessale at gmail.com Tue Nov 15 19:02:46 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Nov 2011 10:02:46 -0600 Subject: [Freeswitch-dev] How can I detect the cause of a potential memory leak? In-Reply-To: <1321349116.20161.101.camel@luna.madrid.commsmundi.com> References: <1321291827.20161.66.camel@luna.madrid.commsmundi.com> <1321349116.20161.101.camel@luna.madrid.commsmundi.com> Message-ID: That report showed me 2 leaks. 1) in mod_cdr_sqlite where it was expanding a variable template and not checking if it needed to free the results. 2) in mod_dptools in the user channel code. Wait, there's more. Fixing the leak in the user channel then led me to realize there was a regression in the change that introduced that leak that was breaking the code that copies all the variables from the {} into the far end channel. anyway, try out GIT head. On Tue, Nov 15, 2011 at 3:25 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > I did a small valgrind test with git HEAD: > 1. turn on, make 5 calls or so > 2. leave FS on for the night (without calls) > 3. shut it down cleanly in the morning > > During execution, some "Warning: invalid file descriptor -1 in syscall > close()" (4-5 for each call + some extras). > > After shutting down, something strikes me in the vg.log file: > "definitely lost: 173,407,885 bytes in 84,680 blocks" > > Is this normal? > > valgrind log: http://pastebin.freeswitch.org/17779 > > Thanks, > Fran?ois. > > > On Mon, 2011-11-14 at 11:41 -0600, Anthony Minessale wrote: > > valgrind is your best bet: > > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full > > --leak-resolution=high > > --show-reachable=yes /usr/local/freeswitch/bin/freeswitch -vg > > > > > > run this on normal traffic for a while and get the log file. > > > > > > if you unload mods it would not help with a leak but it would with a > > swelling pool. > > > > > > > > On Mon, Nov 14, 2011 at 11:30 AM, Fran?ois Delawarde > > wrote: > > Hello, > > > > I'm experiencing what I think is a memory leak in a production > > system > > with a recent git (2-Nov), with FS consuming >5GB and > > increasing after a > > few days with never more than 10 calls at once. > > > > While the users are not experiencing problems so far, I'm a > > bit worried. > > So I have a few questions: > > > > - How can I be sure it's a leak and not some memory pool thing > > that FS > > would free when the system needs (system has 10GB total)? > > > > - If I unload modules one by one, is the memory used by this > > module > > freed immediately? > > > > - What would be a "good" way to try and narrow down the cause? > > Is > > valgrind a good tool for that? > > > > Of course I'll try with GIT HEAD and try my best to find the > > cause > > before considering to add a new Jira issue, otherwise it's > > quite > > useless. > > > > > > Thanks, > > Fran?ois. > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-dev > > 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111115/c1d9f308/attachment.html From msc at freeswitch.org Tue Nov 15 19:44:29 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Nov 2011 08:44:29 -0800 Subject: [Freeswitch-dev] mod_say with a date In-Reply-To: References: Message-ID: I've confirmed this behavior. Can you please open a jira if you have not already done so? Thanks, MC On Thu, Nov 10, 2011 at 9:52 AM, Matthew Margolis < Matthew.Margolis at patlive.com> wrote: > When I try to use mod_say to speak a date, it never gets the 19XX?s > correct. It simply says it as a number such as ?One thousand, nine hundred, > ninety nine? as opposed to ?Nineteen ninety nine.? Am I doing something > wrong? Or is this a bug? I am using a lua script:**** > > ** ** > > session:execute("say", "en-Callie current_date pronounced " .. > [unix-timestamp])**** > > ** ** > > *Matthew Margolis* > > Programmer > PATLive**** > > T 850.422.2527 ext 74**** > > E matthew.margolis at patlive.com**** > > ** ** > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111115/042d5490/attachment.html From fdelawarde at wirelessmundi.com Tue Nov 15 20:02:00 2011 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 15 Nov 2011 18:02:00 +0100 Subject: [Freeswitch-dev] How can I detect the cause of a potential memory leak? In-Reply-To: References: <1321291827.20161.66.camel@luna.madrid.commsmundi.com> <1321349116.20161.101.camel@luna.madrid.commsmundi.com> Message-ID: <1321376520.20161.135.camel@luna.madrid.commsmundi.com> Cool! I should do that more often... I'll first try to figure out how you found those leaks from the report. Thanks! Fran?ois. On Tue, 2011-11-15 at 10:02 -0600, Anthony Minessale wrote: > That report showed me 2 leaks. > > > 1) in mod_cdr_sqlite where it was expanding a variable template and > not checking if it needed to free the results. > 2) in mod_dptools in the user channel code. > > > Wait, there's more. > > > Fixing the leak in the user channel then led me to realize there was a > regression in the change that introduced that leak that was breaking > the code that copies all the variables from the {} into the far end > channel. > > > anyway, try out GIT head. > > > On Tue, Nov 15, 2011 at 3:25 AM, Fran?ois Delawarde > wrote: > I did a small valgrind test with git HEAD: > 1. turn on, make 5 calls or so > 2. leave FS on for the night (without calls) > 3. shut it down cleanly in the morning > > During execution, some "Warning: invalid file descriptor -1 in > syscall > close()" (4-5 for each call + some extras). > > After shutting down, something strikes me in the vg.log file: > "definitely lost: 173,407,885 bytes in 84,680 blocks" > > Is this normal? > > valgrind log: http://pastebin.freeswitch.org/17779 > > Thanks, > Fran?ois. > > > > On Mon, 2011-11-14 at 11:41 -0600, Anthony Minessale wrote: > > valgrind is your best bet: > > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full > > --leak-resolution=high > > --show-reachable=yes /usr/local/freeswitch/bin/freeswitch > -vg > > > > > > run this on normal traffic for a while and get the log file. > > > > > > if you unload mods it would not help with a leak but it > would with a > > swelling pool. > > > > > > > > On Mon, Nov 14, 2011 at 11:30 AM, Fran?ois Delawarde > > wrote: > > Hello, > > > > I'm experiencing what I think is a memory leak in a > production > > system > > with a recent git (2-Nov), with FS consuming >5GB > and > > increasing after a > > few days with never more than 10 calls at once. > > > > While the users are not experiencing problems so > far, I'm a > > bit worried. > > So I have a few questions: > > > > - How can I be sure it's a leak and not some memory > pool thing > > that FS > > would free when the system needs (system has 10GB > total)? > > > > - If I unload modules one by one, is the memory used > by this > > module > > freed immediately? > > > > - What would be a "good" way to try and narrow down > the cause? > > Is > > valgrind a good tool for that? > > > > Of course I'll try with GIT HEAD and try my best to > find the > > cause > > before considering to add a new Jira issue, > otherwise it's > > quite > > useless. > > > > > > Thanks, > > Fran?ois. > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org From anthony.minessale at gmail.com Wed Nov 16 03:20:51 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Nov 2011 18:20:51 -0600 Subject: [Freeswitch-dev] Help Wanted [Consultants and Developers] Message-ID: Anyone out there interested in getting some consulting leads and doing regular work regarding FS or anyone who is ambitious and interested in a larger more permanent position doing development frontend and backend for Barracuda Networks drop us a line. email jobs at freeswitch.org with relevant contact info and CV -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111115/50fcc0df/attachment.html From msc at freeswitch.org Wed Nov 16 19:38:54 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Nov 2011 08:38:54 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conf Call Today Message-ID: Hello all, We have a really light agenda for the conf call today: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_16 Today would be a good day to come talk about your FreeSWITCH questions! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111116/4086773a/attachment.html From SPapineni at enghouse.com Wed Nov 16 20:20:28 2011 From: SPapineni at enghouse.com (Papineni, Suneel) Date: Wed, 16 Nov 2011 17:20:28 +0000 Subject: [Freeswitch-dev] Issue with mod_diangling during compilation Message-ID: <9438D04074E0DE45A49CD76099821272F86BDF@CORP-MAIL-002.edge.local> Hi, I am trying to understand mod_diangling and followed documentation at http://wiki.freeswitch.org/wiki/Dingaling As I am running on Windows, Installed GnuTls and generated libgnutls-26.lib file. Added additional include directory and dependencies to iksemel library project. Now when I compile, it should generate respective DLLs right. Unfortunately I am not finding the TLS DLLs' as mentioned in the link. I tried to compile whole solution and also compiled iksemel project and mod_diangling project separately. Still I didn't find the TLS DLLs in output (Release) folder. Can someone please let me know what I am missing here. Also please let me know if I need to follow any other procedure. Note: I am using latest version of FreeSwitch from GIT and using Visual Studio 2010. Thanks & Regards Suneel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111116/db573535/attachment.html From fdelawarde at wirelessmundi.com Thu Nov 17 17:15:24 2011 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Thu, 17 Nov 2011 15:15:24 +0100 Subject: [Freeswitch-dev] How can I detect the cause of a potential memory leak? In-Reply-To: <1321376520.20161.135.camel@luna.madrid.commsmundi.com> References: <1321291827.20161.66.camel@luna.madrid.commsmundi.com> <1321349116.20161.101.camel@luna.madrid.commsmundi.com> <1321376520.20161.135.camel@luna.madrid.commsmundi.com> Message-ID: <1321539324.16656.26.camel@luna.madrid.commsmundi.com> Master Anthony thanks again, your teachings were not in vain: http://jira.freeswitch.org/browse/FS-3702 Regards, Fran?ois. On Tue, 2011-11-15 at 18:02 +0100, Fran?ois Delawarde wrote: > Cool! I should do that more often... > > I'll first try to figure out how you found those leaks from the report. > > Thanks! > Fran?ois. > > On Tue, 2011-11-15 at 10:02 -0600, Anthony Minessale wrote: > > That report showed me 2 leaks. > > > > > > 1) in mod_cdr_sqlite where it was expanding a variable template and > > not checking if it needed to free the results. > > 2) in mod_dptools in the user channel code. > > > > > > Wait, there's more. > > > > > > Fixing the leak in the user channel then led me to realize there was a > > regression in the change that introduced that leak that was breaking > > the code that copies all the variables from the {} into the far end > > channel. > > > > > > anyway, try out GIT head. > > > > > > On Tue, Nov 15, 2011 at 3:25 AM, Fran?ois Delawarde > > wrote: > > I did a small valgrind test with git HEAD: > > 1. turn on, make 5 calls or so > > 2. leave FS on for the night (without calls) > > 3. shut it down cleanly in the morning > > > > During execution, some "Warning: invalid file descriptor -1 in > > syscall > > close()" (4-5 for each call + some extras). > > > > After shutting down, something strikes me in the vg.log file: > > "definitely lost: 173,407,885 bytes in 84,680 blocks" > > > > Is this normal? > > > > valgrind log: http://pastebin.freeswitch.org/17779 > > > > Thanks, > > Fran?ois. > > > > > > > > On Mon, 2011-11-14 at 11:41 -0600, Anthony Minessale wrote: > > > valgrind is your best bet: > > > > > > valgrind --tool=memcheck --log-file=vg.log --leak-check=full > > > --leak-resolution=high > > > --show-reachable=yes /usr/local/freeswitch/bin/freeswitch > > -vg > > > > > > > > > run this on normal traffic for a while and get the log file. > > > > > > > > > if you unload mods it would not help with a leak but it > > would with a > > > swelling pool. > > > > > > > > > > > > On Mon, Nov 14, 2011 at 11:30 AM, Fran?ois Delawarde > > > wrote: > > > Hello, > > > > > > I'm experiencing what I think is a memory leak in a > > production > > > system > > > with a recent git (2-Nov), with FS consuming >5GB > > and > > > increasing after a > > > few days with never more than 10 calls at once. > > > > > > While the users are not experiencing problems so > > far, I'm a > > > bit worried. > > > So I have a few questions: > > > > > > - How can I be sure it's a leak and not some memory > > pool thing > > > that FS > > > would free when the system needs (system has 10GB > > total)? > > > > > > - If I unload modules one by one, is the memory used > > by this > > > module > > > freed immediately? > > > > > > - What would be a "good" way to try and narrow down > > the cause? > > > Is > > > valgrind a good tool for that? > > > > > > Of course I'll try with GIT HEAD and try my best to > > find the > > > cause > > > before considering to add a new Jira issue, > > otherwise it's > > > quite > > > useless. > > > > > > > > > Thanks, > > > Fran?ois. > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > > Server > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-dev mailing list > > > FreeSWITCH-dev at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > > 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_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > googletalk:conf+888 at conference.freeswitch.org > > > pstn:+19193869900 > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-dev mailing list > > > FreeSWITCH-dev at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org From SPapineni at enghouse.com Thu Nov 17 19:15:28 2011 From: SPapineni at enghouse.com (Papineni, Suneel) Date: Thu, 17 Nov 2011 16:15:28 +0000 Subject: [Freeswitch-dev] Joining a call from Gtalk to a conference on FreeSwitch Message-ID: <9438D04074E0DE45A49CD76099821272F86F86@CORP-MAIL-002.edge.local> Hi, I got the new GIT version and enabled mod_dingling and compiled. Everything went through and able to establish call to an extension if I configure that extension number in "client" profile. What I am trying to do is, I want to bridge or join a call coming from GTalk to an existing conference in FreeSwitch. For this purpose I configured a different number on "client profile" and created a dial-plan for this number to 'park' the call first before trying to join to the conference. Then using eventSockets I am trying to join this call to conference and issued following command. (tried with "uuid_bridge" command as well) "api uuid_transfer [Unique-ID] conference:xyz at default inline" Command is successful and also I can hear a sound that someone joined in the conference, but I didn't hear any voice at either side. I couldn't see any RTP flow as well (checked wireshark traces at FS). After sometime like 30 seconds call at GTalk is disconnected automatically. I am not sure why nothing is heard at both sides and why call got disconnected. Also tried answering the call first (after Park) and then bridging to conference, still got the same issue. Could someone please let me know if I am missing anything or need to configure in a different way for conferencing. Thanks & Regards Suneel Client.xml Dial-plan.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111117/c4e75b67/attachment-0001.html From ahe.sanath at gmail.com Thu Nov 17 11:25:23 2011 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Thu, 17 Nov 2011 13:55:23 +0530 Subject: [Freeswitch-dev] Connect Voice mail flow before sending Answer. Message-ID: Hi all, I am new to Freeswitch & need to do following. When called party not available, I need to route call to voice mail system without sending answer.(or OK) If send answer, calling party will be charged. So I need to avoid that. After that I need to prompt message as "Press 1 for go to voicemail or Press 2 to exit" If user press1 then send answer (Charging is start) & connect to voice mail call flow. For doing above, shall I need to do code level changes ?? Pls advice. Br, Sanath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111117/fe851774/attachment.html From john.gubba at rediffmail.com Thu Nov 17 15:54:50 2011 From: john.gubba at rediffmail.com (john ) Date: 17 Nov 2011 12:54:50 -0000 Subject: [Freeswitch-dev] =?utf-8?q?event_dispatch_thread_should_exit_when?= =?utf-8?q?_idle?= Message-ID: <20111117125450.11195.qmail@f4mail-235-133.rediffmail.com> Dear Friends, I was going through the code of event mechanism in freeswitch. In switch_event.cThe switch event thread may spawn more dispatch threads in case heavy load.But I am wandering why there isn't any exit mechanism of the event dispatch thread. event dispatch thread only exits when shutdown. Current event dispatch queue size is 10,000. At some point of time event thread detects 12,000 events to be dispatched. It spawns a new dispatch thread - 2, to deliver remaining 2000 events. but after that dispatch thread -2 remains active & idle. I think we need to kill this idle thread+queue, so that it can be re spawned latter on. and max no of dispatch threads never get exceed. Please reply me if anyone has implemented this or have any idea abt this. thnx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111117/a5d6190d/attachment.html From anthony.minessale at gmail.com Thu Nov 17 22:31:52 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Nov 2011 13:31:52 -0600 Subject: [Freeswitch-dev] event dispatch thread should exit when idle In-Reply-To: <20111117125450.11195.qmail@f4mail-235-133.rediffmail.com> References: <20111117125450.11195.qmail@f4mail-235-133.rediffmail.com> Message-ID: once the thread is up its asleep anyway, it would be possible to make it go away but its not really hurting anything and if the same situation is likely to continue to happen its more efficient to be ready for it the subsequent times the load occurs. On Thu, Nov 17, 2011 at 6:54 AM, john wrote: > Dear Friends, > > I was going through the code of event mechanism in freeswitch. In > switch_event.c > The switch event thread may spawn more dispatch threads in case heavy load. > But I am wandering why there isn't any exit mechanism of the event > dispatch thread. > > event dispatch thread only exits when shutdown. > > Current event dispatch queue size is 10,000. At some point of time event > thread detects 12,000 events to be dispatched. It spawns a new dispatch > thread - 2, to deliver remaining 2000 events. but after that dispatch > thread -2 remains active & idle. > > I think we need to kill this idle thread+queue, so that it can be re > spawned latter on. and max no of dispatch threads never get exceed. > > Please reply me if anyone has implemented this or have any idea abt this. > > thnx > > > > > Follow *Rediff Deal ho jaye! > * to get exciting offers in your city everyday. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111117/0f3d0b88/attachment.html From freeswitch-list at puzzled.xs4all.nl Fri Nov 18 02:33:44 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Fri, 18 Nov 2011 00:33:44 +0100 Subject: [Freeswitch-dev] Joining a call from Gtalk to a conference on FreeSwitch In-Reply-To: <9438D04074E0DE45A49CD76099821272F86F86@CORP-MAIL-002.edge.local> References: <9438D04074E0DE45A49CD76099821272F86F86@CORP-MAIL-002.edge.local> Message-ID: <4EC599D8.7080107@puzzled.xs4all.nl> On 11/17/2011 05:15 PM, Papineni, Suneel wrote: [snip] > Could someone please let me know if I am missing anything or need to > configure in a different way for conferencing. You posted to the wrong mailing list. The FreeSWITCH-dev mailing list is for discussing the development of FreeSWITCH and not for user questions like yours. Also do not cross-post your question to multiple mailing lists at the same time. It will not get you an answer any quicker. Probably more the contrary as cross-posting is frowned upon. Regards, Patrick From rml at tollfreeforwarding.com Fri Nov 18 12:59:55 2011 From: rml at tollfreeforwarding.com (RaviRaj Mulasa) Date: Fri, 18 Nov 2011 09:59:55 +0000 Subject: [Freeswitch-dev] Loading an IVR menu on the fly through an URL Message-ID: <8B94625BC339264DBA61E314BE9EC2CF230EC8BA@EXCH125.IFN.com> Hi FreeSWITCH Enthusiastics/Gurus Requirement 1. Customers can build a custom IVR using a UI. 2. Back end code persists the IVR details in a DB. 3. A URL will be exposed and will return the a XML file in the format as follows > 4. Load and start executing the IVR 'CUSTOMER_ID_ivr' from events socket on the fly via HTTP URI. Gave it a shot with current setup ivr.conf.xml The ivr.conf.xml is auto loaded at startup of FreeSWITCH. To load any new IVR menus , we need to call API command reloadxml which reloads the whole XML configs in autload_configs folder. The reloadxml might take long time based on the number of XML files present in the folder "/mnt/<>/ivr_menus" Good /Hope to have for the IVR menu 1. Load /Unload IVR menus on the fly using API command(s) freeswitch at localhost> loadivr << URL to fetch the XML describing the IVR menu(s)>> freeswitch at localhost> unloadivr <> 2. From the event socket SendMsg call-command: execute execute-app-name: ivr execute-app-arg: <>SPACE <> Educated/Wild Guess We might be able to play IVR menu on the fly by combining the mod_xml_curl and dialplan tool - ivr. Please let us know how can we achieve the functionality , we are more interested in the second approach(Event Socket) under Good /Hope to have for the IVR menu. Thanks RaviRaj. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111118/d27335c0/attachment-0001.html From msc at freeswitch.org Sat Nov 19 00:12:35 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Nov 2011 13:12:35 -0800 Subject: [Freeswitch-dev] Connect Voice mail flow before sending Answer. In-Reply-To: References: Message-ID: On Thu, Nov 17, 2011 at 12:25 AM, Sanath Prasanna wrote: > Hi all, > I am new to Freeswitch & need to do following. > > When called party not available, I need to route call to voice mail system > without sending answer.(or OK) If send answer, calling party will be > charged. So I need to avoid that. After that I need to prompt message as > "Press 1 for go to voicemail or Press 2 to exit" > > If user press1 then send answer (Charging is start) & connect to voice > mail call flow. > For doing above, shall I need to do code level changes ?? Pls advice. > What is your scenario? Is the called party on your FreeSWITCH system? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111118/4ed32015/attachment.html From msc at freeswitch.org Tue Nov 22 06:03:41 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Nov 2011 19:03:41 -0800 Subject: [Freeswitch-dev] Wednesday Conf Call - SIP Presence? Message-ID: Hi all, I would like to have someone speak on the subject of SIP presence this coming Wednesday. If you are up for giving the community a brief introduction to the concepts involved with SIP presence please let me know. Also, if you are experienced and/or knowledgeable on the subject but would rather not give a presentation also let me know as you may be in a position to type up some material that could be used by the presenter. Please email me off list. Thanks all! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111121/be7a9ebd/attachment.html From succer110 at tiscali.it Tue Nov 22 16:38:13 2011 From: succer110 at tiscali.it (succer110 at tiscali.it) Date: Tue, 22 Nov 2011 14:38:13 +0100 (CET) Subject: [Freeswitch-dev] Other-Leg-uuid before bridge Message-ID: <31448356.99021321969093910.JavaMail.defaultUser@defaultHost> Hi everybody! I'm executing a script with exec_on_answer and exec_on_pre_answer in a bridge application. Since the execution of my application is triggered before bridging the two legs (that's exactly hat what i was looking for) i do not have the "Other-Leg-uuid" var setted in my session's channel variable. Is there any way to get the uuid of the bleg before the bridge? So: in the dial.lua i need to have the uuid of "sofia/external/1111 at 192.168.1.1". Thank you in advice! (and thank you all for freeswitch!!) E' nata indoona : chiama, videochiama e messaggia Gratis. Scarica indoona per iPhone, Android e PC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111122/1599dd6c/attachment.html From xin at ind.rwth-aachen.de Tue Nov 22 17:31:10 2011 From: xin at ind.rwth-aachen.de (Han Xin) Date: Tue, 22 Nov 2011 15:31:10 +0100 Subject: [Freeswitch-dev] why does app "record_session" record extra frequency range Message-ID: <38f3f3fccf08a3eacad1eba105f0772d@gw.ind.rwth-aachen.de> Hi all, I have a bridged call and a home brew module which adds a media bug to the local and partner session before the bridge app in the dial plan. The original speech codec used is G722, which sampling rate is 16KHz. I add a media bug which does low pass filtering the speech to 4KHz. But the spectrum of the recorded .wav file showed that it always contains the freq range from 0K to 8K. The desired spectrum should contain only 0 to 4KHz. I am confused, can anyone show me some hints? Thanks in advance. Best Regards, Han -- From anthony.minessale at gmail.com Tue Nov 22 18:54:41 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Nov 2011 09:54:41 -0600 Subject: [Freeswitch-dev] why does app "record_session" record extra frequency range In-Reply-To: <38f3f3fccf08a3eacad1eba105f0772d@gw.ind.rwth-aachen.de> References: <38f3f3fccf08a3eacad1eba105f0772d@gw.ind.rwth-aachen.de> Message-ID: Do you have an example of how you're doing it? On Tue, Nov 22, 2011 at 8:31 AM, Han Xin wrote: > Hi all, > > I have a bridged call and a home brew module which adds a media bug to the > local and partner session before the bridge app in the dial plan. The > original speech codec used is G722, which sampling rate is 16KHz. I add a > media bug which does low pass filtering the speech to 4KHz. But the > spectrum of the recorded .wav file showed that it always contains the freq > range from 0K to 8K. The desired spectrum should contain only 0 to 4KHz. > > I am confused, can anyone show me some hints? Thanks in advance. > > Best Regards, > > Han > -- > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111122/e3b010b3/attachment.html From jbaclor at ezuce.com Tue Nov 22 10:35:55 2011 From: jbaclor at ezuce.com (Joegen Baclor) Date: Tue, 22 Nov 2011 15:35:55 +0800 Subject: [Freeswitch-dev] Submitted a patch to correct compile error of mod_shout inFC 15 & 16 Message-ID: <4ECB50DB.2010302@ezuce.com> Jira is here http://jira.freeswitch.org/browse/FS-3711 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111122/4eca3d4c/attachment.html From Giovanni.Visciano at italtel.it Tue Nov 22 16:57:12 2011 From: Giovanni.Visciano at italtel.it (Visciano Giovanni) Date: Tue, 22 Nov 2011 14:57:12 +0100 Subject: [Freeswitch-dev] Sofia - RFC2396 URI escape Message-ID: According to RFC2396 a sip URI should not contain reserved character in uri syntax (ex: #<>"). It seems sofia does not apply the escape function. For example using the cli to originate a call: > originate sofia/external/*2<1#@192.168.1.45 55 Result in: INVITE sip:*2<1#@192.168.1.45 SIP/2.0 Via: SIP/2.0/UDP 138.132.50.77:5080;rport;branch=z9hG4bK6KDeZ78yH4tFS Max-Forwards: 70 Regards Giovanni Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). 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If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From xin at ind.rwth-aachen.de Tue Nov 22 19:12:15 2011 From: xin at ind.rwth-aachen.de (Han Xin) Date: Tue, 22 Nov 2011 17:12:15 +0100 Subject: [Freeswitch-dev] why does app "record_session" record extra frequency range In-Reply-To: Message-ID: <8b5f97c9c40e7579b4cbe7eed5f2b531@gw.ind.rwth-aachen.de> Hi Anthony, Problem is solved, I reverse the order of record_session and the low pass filtering module in the dial plan. Then it turns out to be working as expected. I am wondering how the order of media bugs are added and processed now. :P ----------------urspr?ngliche Nachricht----------------- Von: "Anthony Minessale" anthony.minessale at gmail.com An: freeswitch-dev at lists.freeswitch.org Datum: Tue, 22 Nov 2011 09:54:41 -0600 ------------------------------------------------- > Do you have an example of how you're doing it? > > On Tue, Nov 22, 2011 at 8:31 AM, Han Xin xin at ind.rwth-aachen.de wrote: > >> Hi all, >> >> I have a bridged call and a home brew module which adds a media bug to the >> local and partner session before the bridge app in the dial plan. The >> original speech codec used is G722, which sampling rate is 16KHz. I add a >> media bug which does low pass filtering the speech to 4KHz. But the >> spectrum of the recorded .wav file showed that it always contains the freq >> range from 0K to 8K. The desired spectrum should contain only 0 to 4KHz. >> >> I am confused, can anyone show me some hints? Thanks in advance. >> >> Best Regards, >> >> Han >> -- >> >> >> >> >> >> __________________________________________________________________ >> _______ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch >> -dev >> 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > __________________________________________________ > > ____________________________________________________________________ > _____ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-d > ev > http://www.freeswitch.org > -- From anthony.minessale at gmail.com Tue Nov 22 19:30:17 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Nov 2011 10:30:17 -0600 Subject: [Freeswitch-dev] Submitted a patch to correct compile error of mod_shout inFC 15 & 16 In-Reply-To: <4ECB50DB.2010302@ezuce.com> References: <4ECB50DB.2010302@ezuce.com> Message-ID: This seems somewhat ridiculous of a change. (the change to patch not your patch to fix the patched patch) Why bother not allowing that when you can just as easily cat ../../../file | patch making it pointless to change the behavior. I'm not trying to shoot the messenger but it's fairly silly. On Tue, Nov 22, 2011 at 1:35 AM, Joegen Baclor wrote: > Jira is here http://jira.freeswitch.org/browse/FS-3711 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111122/a20be0a1/attachment.html From msc at freeswitch.org Wed Nov 23 03:16:11 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Nov 2011 16:16:11 -0800 Subject: [Freeswitch-dev] Other-Leg-uuid before bridge In-Reply-To: <31448356.99021321969093910.JavaMail.defaultUser@defaultHost> References: <31448356.99021321969093910.JavaMail.defaultUser@defaultHost> Message-ID: You can set the uuid yourself: Check out the create_uuid API and origination_uuid chan var on the wiki... -MC On Tue, Nov 22, 2011 at 5:38 AM, succer110 at tiscali.it wrote: > Hi everybody! > > I'm executing a script with exec_on_answer and exec_on_pre_answer in a > bridge application. > Since the execution of my application is triggered before bridging the two > legs (that's exactly hat what i was looking for) i do not have the "Other-Leg-uuid" > var setted in my session's channel variable. Is there any way to get the > uuid of the bleg before the bridge? > > data="nolocal:execute_on_pre_answer=lua dial.lua PRGS"/> > > > > So: in the dial.lua i need to have the uuid of "sofia/external/ > 1111 at 192.168.1.1". > Thank you in advice! > (and thank you all for freeswitch!!) > > E' nata indoona : chiama, videochiama e messaggia Gratis. > Scarica indoona per iPhone, > Android e PC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111122/9b332ce5/attachment.html From callum.guy at x-on.co.uk Wed Nov 23 12:44:59 2011 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 23 Nov 2011 09:44:59 +0000 Subject: [Freeswitch-dev] Other-Leg-uuid before bridge In-Reply-To: References: <31448356.99021321969093910.JavaMail.defaultUser@defaultHost> Message-ID: Hi Michael, This brings me neatly on to a long outstanding query of mine - does this UUID stick? All testing and evidence suggests that the originating_uuid specified continues as the channel uuid after answer, however the wiki ( http://wiki.freeswitch.org/wiki/Mod_commands) states the following (UUID specification is key to part of my system and has left me worrying that this may cause a headache in the future): Wiki text: You can specify the UUID of an originated call by doing the following: - Use create_uuid to generate a UUID to use. - This will allow you to kill an originated call before it is answered by using uuid_kill. - The UUID of the answered call leg will not be the same UUID as the origination_uuid specified (Each call leg always gets its own UUID) Please tell me the wiki is out of date! Thanks, Callum On 23 November 2011 00:16, Michael Collins wrote: > You can set the uuid yourself: > > > data="{origination_uuid=${my_uuid}}/sofia/foo/bar"/> > > Check out the create_uuid API and origination_uuid chan var on the wiki... > > -MC > > On Tue, Nov 22, 2011 at 5:38 AM, succer110 at tiscali.it < > succer110 at tiscali.it> wrote: > >> Hi everybody! >> >> I'm executing a script with exec_on_answer and exec_on_pre_answer in a >> bridge application. >> Since the execution of my application is triggered before bridging the >> two legs (that's exactly hat what i was looking for) i do not have the "Other-Leg-uuid" >> var setted in my session's channel variable. Is there any way to get the >> uuid of the bleg before the bridge? >> >> > data="nolocal:execute_on_pre_answer=lua dial.lua PRGS"/> >> >> >> >> So: in the dial.lua i need to have the uuid of "sofia/external/ >> 1111 at 192.168.1.1". >> Thank you in advice! >> (and thank you all for freeswitch!!) >> >> E' nata indoona : chiama, videochiama e messaggia Gratis. >> Scarica indoona per iPhone, >> Android e PC >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111123/35acb80a/attachment.html From msc at freeswitch.org Wed Nov 23 19:35:29 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Nov 2011 08:35:29 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_23 There have been a number of changes/additions to mod_conference, so we are going to talk about those today. Also, there are several FreeSWITCH updates to talk about. If we have time and if enough subject matter experts are available we will also talk about SIP presence. See you soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111123/4e7db7e0/attachment-0001.html From anthony.minessale at gmail.com Wed Nov 23 22:26:39 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Nov 2011 13:26:39 -0600 Subject: [Freeswitch-dev] Other-Leg-uuid before bridge In-Reply-To: References: <31448356.99021321969093910.JavaMail.defaultUser@defaultHost> Message-ID: if you specify origination_uuid it will remain the uuid for the whole session On Wed, Nov 23, 2011 at 3:44 AM, Callum Guy wrote: > Hi Michael, > > This brings me neatly on to a long outstanding query of mine - does this > UUID stick? All testing and evidence suggests that the originating_uuid > specified continues as the channel uuid after answer, however the wiki ( > http://wiki.freeswitch.org/wiki/Mod_commands) states the following (UUID > specification is key to part of my system and has left me worrying that > this may cause a headache in the future): > > Wiki text: > > You can specify the UUID of an originated call by doing the following: > > - Use create_uuid to generate a UUID to use. > - This will allow you to kill an originated call before it is answered > by using uuid_kill. > - The UUID of the answered call leg will not be the same UUID as the > origination_uuid specified (Each call leg always gets its own UUID) > > Please tell me the wiki is out of date! > > Thanks, > > Callum > > > > > On 23 November 2011 00:16, Michael Collins wrote: > >> You can set the uuid yourself: >> >> >> > data="{origination_uuid=${my_uuid}}/sofia/foo/bar"/> >> >> Check out the create_uuid API and origination_uuid chan var on the wiki... >> >> -MC >> >> On Tue, Nov 22, 2011 at 5:38 AM, succer110 at tiscali.it < >> succer110 at tiscali.it> wrote: >> >>> Hi everybody! >>> >>> I'm executing a script with exec_on_answer and exec_on_pre_answer in a >>> bridge application. >>> Since the execution of my application is triggered before bridging the >>> two legs (that's exactly hat what i was looking for) i do not have the "Other-Leg-uuid" >>> var setted in my session's channel variable. Is there any way to get >>> the uuid of the bleg before the bridge? >>> >>> >> data="nolocal:execute_on_pre_answer=lua dial.lua PRGS"/> >>> >>> >>> >>> So: in the dial.lua i need to have the uuid of "sofia/external/ >>> 1111 at 192.168.1.1". >>> Thank you in advice! >>> (and thank you all for freeswitch!!) >>> >>> E' nata indoona : chiama, videochiama e messaggia Gratis. >>> Scarica indoona per iPhone, >>> Android e PC >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111123/2970651f/attachment.html From callum.guy at x-on.co.uk Thu Nov 24 12:29:37 2011 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 24 Nov 2011 09:29:37 +0000 Subject: [Freeswitch-dev] Other-Leg-uuid before bridge In-Reply-To: References: <31448356.99021321969093910.JavaMail.defaultUser@defaultHost> Message-ID: That's great, I appreciate the confirmation. I've updated the wiki to clear this up for new users. ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 23 November 2011 19:26, Anthony Minessale wrote: > if you specify origination_uuid it will remain the uuid for the whole > session > > > On Wed, Nov 23, 2011 at 3:44 AM, Callum Guy wrote: > >> Hi Michael, >> >> This brings me neatly on to a long outstanding query of mine - does this >> UUID stick? All testing and evidence suggests that the originating_uuid >> specified continues as the channel uuid after answer, however the wiki ( >> http://wiki.freeswitch.org/wiki/Mod_commands) states the following (UUID >> specification is key to part of my system and has left me worrying that >> this may cause a headache in the future): >> >> Wiki text: >> >> You can specify the UUID of an originated call by doing the following: >> >> - Use create_uuid to generate a UUID to use. >> - This will allow you to kill an originated call before it is >> answered by using uuid_kill. >> - The UUID of the answered call leg will not be the same UUID as the >> origination_uuid specified (Each call leg always gets its own UUID) >> >> Please tell me the wiki is out of date! >> >> Thanks, >> >> Callum >> >> >> >> >> On 23 November 2011 00:16, Michael Collins wrote: >> >>> You can set the uuid yourself: >>> >>> >>> >> data="{origination_uuid=${my_uuid}}/sofia/foo/bar"/> >>> >>> Check out the create_uuid API and origination_uuid chan var on the >>> wiki... >>> >>> -MC >>> >>> On Tue, Nov 22, 2011 at 5:38 AM, succer110 at tiscali.it < >>> succer110 at tiscali.it> wrote: >>> >>>> Hi everybody! >>>> >>>> I'm executing a script with exec_on_answer and exec_on_pre_answer in a >>>> bridge application. >>>> Since the execution of my application is triggered before bridging the >>>> two legs (that's exactly hat what i was looking for) i do not have the "Other-Leg-uuid" >>>> var setted in my session's channel variable. Is there any way to get >>>> the uuid of the bleg before the bridge? >>>> >>>> >>> data="nolocal:execute_on_pre_answer=lua dial.lua PRGS"/> >>>> >>> data="nolocal:execute_on_answer=lua dial.lua ANSW"/> >>>> >>>> >>>> So: in the dial.lua i need to have the uuid of "sofia/external/ >>>> 1111 at 192.168.1.1". >>>> Thank you in advice! >>>> (and thank you all for freeswitch!!) >>>> >>>> E' nata indoona : chiama, videochiama e messaggia Gratis. >>>> Scarica indoona per iPhone, >>>> Android e >>>> PC >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-dev mailing list >>>> FreeSWITCH-dev at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111124/f82cf1f6/attachment-0001.html From bhrugumehta at gmail.com Thu Nov 24 15:57:23 2011 From: bhrugumehta at gmail.com (Bhrugu Mehta) Date: Thu, 24 Nov 2011 18:27:23 +0530 Subject: [Freeswitch-dev] condition chaking.. Message-ID: hi, all asterisk setup: exten => _X.,1,GotoIf($[${status1}=on]?n1:n2) exten => _X.,n(n1),GotoIf($[${status2}=off]?n3:n4) exten => _X.,(n2),GotoIf($[${status3}=X]?n5:n6) ... ... How do i implement in Freeswitch... Regards, -- Bhrugu Mehta Sr. S/W Engineer VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) India -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111124/81676f05/attachment.html From bhrugumehta at gmail.com Thu Nov 24 15:57:53 2011 From: bhrugumehta at gmail.com (Bhrugu Mehta) Date: Thu, 24 Nov 2011 18:27:53 +0530 Subject: [Freeswitch-dev] condition check... Message-ID: hi, all asterisk setup: exten => _X.,1,GotoIf($[${status1}=on]?n1:n2) exten => _X.,n(n1),GotoIf($[${status2}=off]?n3:n4) exten => _X.,(n2),GotoIf($[${status3}=X]?n5:n6) ... ... How do i implement in Freeswitch... Regards, -- Bhrugu Mehta Sr. S/W Engineer VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) India -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111124/f69802e5/attachment.html From bhrugumehta at gmail.com Thu Nov 24 15:58:26 2011 From: bhrugumehta at gmail.com (Bhrugu Mehta) Date: Thu, 24 Nov 2011 18:28:26 +0530 Subject: [Freeswitch-dev] condition check.. Message-ID: hi, all asterisk setup: exten => _X.,1,GotoIf($[${status1}=on]?n1:n2) exten => _X.,n(n1),GotoIf($[${status2}=off]?n3:n4) exten => _X.,(n2),GotoIf($[${status3}=X]?n5:n6) ... ... How do i implement in Freeswitch... Regards, -- Bhrugu Mehta Sr. S/W Engineer VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) India -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111124/a97974fc/attachment.html From freeswitch-list at puzzled.xs4all.nl Thu Nov 24 18:04:49 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Thu, 24 Nov 2011 16:04:49 +0100 Subject: [Freeswitch-dev] condition check.. In-Reply-To: References: Message-ID: <4ECE5D11.3020208@puzzled.xs4all.nl> You posted to the wrong mailing list. The FreeSWITCH-dev mailing list is for discussing the development of FreeSWITCH and not for user questions like yours. On 24-11-11 13:58, Bhrugu Mehta wrote: > hi, all > asterisk setup: > exten => _X.,1,GotoIf($[${status1}=on]?n1:n2) > exten => _X.,n(n1),GotoIf($[${status2}=off]?n3:n4) > exten => _X.,(n2),GotoIf($[${status3}=X]?n5:n6) > ... > ... > > How do i implement in Freeswitch... Buy the excellent FreeSWITCH book and read it. It has most answers. Next time maybe you can first Google for "FreeSWITCH GotoIf" or any other question you have before posting to the *right* mailing list. http://lmgtfy.com/?q=freeswitch+gotoif Regards, Patrick From Giovanni.Visciano at italtel.it Thu Nov 24 17:33:29 2011 From: Giovanni.Visciano at italtel.it (Visciano Giovanni) Date: Thu, 24 Nov 2011 15:33:29 +0100 Subject: [Freeswitch-dev] Sofia - RFC2396 URI escape In-Reply-To: References: Message-ID: I have modified my dialplan configuration to apply url_encode() to the destination number I want to bridge to. So sofia receive a correctly escaped URI. Thanks. Giovanni -----Original Message----- From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Visciano Giovanni Sent: marted? 22 novembre 2011 14.57 To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] Sofia - RFC2396 URI escape According to RFC2396 a sip URI should not contain reserved character in uri syntax (ex: #<>"). It seems sofia does not apply the escape function. For example using the cli to originate a call: > originate sofia/external/*2<1#@192.168.1.45 55 Result in: INVITE sip:*2<1#@192.168.1.45 SIP/2.0 Via: SIP/2.0/UDP 138.132.50.77:5080;rport;branch=z9hG4bK6KDeZ78yH4tFS Max-Forwards: 70 Regards Giovanni Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org From brian at freeswitch.org Sat Nov 26 01:38:02 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 25 Nov 2011 16:38:02 -0600 Subject: [Freeswitch-dev] Sofia - RFC2396 URI escape In-Reply-To: References: Message-ID: <138A6267-11BF-4F6A-B7EF-3D66582F3EC7@freeswitch.org> We could url encode it ... but it would break things out there that don't properly URL decode... ie nextone if I recall. /b On Nov 24, 2011, at 8:33 AM, Visciano Giovanni wrote: > I have modified my dialplan configuration to apply url_encode() to the destination number I want to bridge to. > > > > So sofia receive a correctly escaped URI. > > Thanks. > Giovanni -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111125/298bf718/attachment.html From karol at tls.pl Mon Nov 28 11:31:52 2011 From: karol at tls.pl (Karol Golab) Date: Mon, 28 Nov 2011 09:31:52 +0100 Subject: [Freeswitch-dev] Possible deadlock in src/switch_event.c ? Message-ID: <4ED346F8.6030604@tls.pl> Hi! I've noticed some strange code in src/switch_event.c - could someone smarter (& knowing the mentioned code) please take a look? The problem is in the order of calls to locks BLOCK & RWLOCK - it differs between functions: switch_event_bind_removable switch_mutex_lock(BLOCK); switch_thread_rwlock_wrlock(RWLOCK); switch_event_unbind_callback switch_thread_rwlock_wrlock(RWLOCK); switch_mutex_lock(BLOCK); switch_event_unbind switch_thread_rwlock_wrlock(RWLOCK); switch_mutex_lock(BLOCK); As I understand this may lead to an outright deadlock between bind and unbind. I stumbled upon this code while debugging what seems like deadlocks in our test FS installation put under some heavy load. The load is about 25 new event subscribers per second (destroyed few seconds later) and about 700 events per second. Regards, Karol -- e-mail: Karol.Golab at tls.pl www: http://www.tls.pl signature: not found company: TLS-Technologie Sp??ka z ograniczon? odpowiedzialno?ci? z siedzib? w Warszawie przy ul. Polnej 50 00-644; KRS 0000073337 (XII Wydzia? Gospodarczy KRS); NIP 526-25-78-091; VAT ID PL5262578091; REGON 017349442; kap. zak. 50000,00 z? From brian at freeswitch.org Mon Nov 28 19:27:06 2011 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Nov 2011 10:27:06 -0600 Subject: [Freeswitch-dev] Possible deadlock in src/switch_event.c ? In-Reply-To: <4ED346F8.6030604@tls.pl> References: <4ED346F8.6030604@tls.pl> Message-ID: Please post it to jira.freeswitch.org /b On Nov 28, 2011, at 2:31 AM, Karol Golab wrote: > > Hi! > > I've noticed some strange code in src/switch_event.c - could someone > smarter (& knowing the mentioned code) please take a look? > > The problem is in the order of calls to locks BLOCK & RWLOCK - it > differs between functions: > > switch_event_bind_removable > switch_mutex_lock(BLOCK); > switch_thread_rwlock_wrlock(RWLOCK); > > switch_event_unbind_callback > switch_thread_rwlock_wrlock(RWLOCK); > switch_mutex_lock(BLOCK); > > switch_event_unbind > switch_thread_rwlock_wrlock(RWLOCK); > switch_mutex_lock(BLOCK); > > As I understand this may lead to an outright deadlock between bind > and unbind. > > I stumbled upon this code while debugging what seems like deadlocks > in our test FS installation put under some heavy load. The load is about > 25 new event subscribers per second (destroyed few seconds later) and > about 700 events per second. > > Regards, > Karol -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111128/d48e1d80/attachment.html From anthony.minessale at gmail.com Mon Nov 28 19:36:49 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Nov 2011 10:36:49 -0600 Subject: [Freeswitch-dev] Possible deadlock in src/switch_event.c ? In-Reply-To: References: <4ED346F8.6030604@tls.pl> Message-ID: you're probably right but I wonder what you are doing that is binding and unbinding so much? The bind and unbind was designed to be called once when the module is loaded and called again when it exits. My fear is you are using the embedded scripting event handers in some quick repeating fashion. I did a commit to flip that case around but I don't encourage you to use embedded scripts for heavy event traffic, that's what event socket is for and if you have the C skills to debug this, you might want to try a C app ;) 2011/11/28 Brian West > Please post it to jira.freeswitch.org > > /b > > On Nov 28, 2011, at 2:31 AM, Karol Golab wrote: > > > Hi! > > I've noticed some strange code in src/switch_event.c - could someone > smarter (& knowing the mentioned code) please take a look? > > The problem is in the order of calls to locks BLOCK & RWLOCK - it > differs between functions: > > switch_event_bind_removable > switch_mutex_lock(BLOCK); > switch_thread_rwlock_wrlock(RWLOCK); > > switch_event_unbind_callback > switch_thread_rwlock_wrlock(RWLOCK); > switch_mutex_lock(BLOCK); > > switch_event_unbind > switch_thread_rwlock_wrlock(RWLOCK); > switch_mutex_lock(BLOCK); > > As I understand this may lead to an outright deadlock between bind > and unbind. > > I stumbled upon this code while debugging what seems like deadlocks > in our test FS installation put under some heavy load. The load is about > 25 new event subscribers per second (destroyed few seconds later) and > about 700 events per second. > > Regards, > Karol > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111128/b361f66e/attachment-0001.html From karol at tls.pl Mon Nov 28 23:13:23 2011 From: karol at tls.pl (Karol Golab) Date: Mon, 28 Nov 2011 21:13:23 +0100 Subject: [Freeswitch-dev] Possible deadlock in src/switch_event.c ? In-Reply-To: References: <4ED346F8.6030604@tls.pl> Message-ID: <4ED3EB63.4020203@tls.pl> Well, I have a Perl script run as part of a dialplan and the script creates an EventConsumer. Thus I have one bind / unbind pair for each connection. Is this somehow wrong / against what the EventConsumer API was planned for? How could I have a single bind/unbind in this case? As for choosing event socket and C application vs a script - what is the main difference / reason to switch: performance of some other issues? My CPU seems to have some cycles to spare. Regards, Karol On 11/28/2011 05:36 PM, Anthony Minessale wrote: > you're probably right but I wonder what you are doing that is binding > and unbinding so much? > The bind and unbind was designed to be called once when the module is > loaded and called again when it exits. > > My fear is you are using the embedded scripting event handers in some > quick repeating fashion. > > I did a commit to flip that case around but I don't encourage you to > use embedded scripts for heavy event traffic, that's what event socket > is for and if you have the C skills to debug this, you might want to > try a C app ;) > > > 2011/11/28 Brian West > > > Please post it to jira.freeswitch.org > > /b > > On Nov 28, 2011, at 2:31 AM, Karol Golab wrote: > >> >> Hi! >> >> I've noticed some strange code in src/switch_event.c - could >> someone >> smarter (& knowing the mentioned code) please take a look? >> >> The problem is in the order of calls to locks BLOCK & RWLOCK - it >> differs between functions: >> >> switch_event_bind_removable >> switch_mutex_lock(BLOCK); >> switch_thread_rwlock_wrlock(RWLOCK); >> >> switch_event_unbind_callback >> switch_thread_rwlock_wrlock(RWLOCK); >> switch_mutex_lock(BLOCK); >> >> switch_event_unbind >> switch_thread_rwlock_wrlock(RWLOCK); >> switch_mutex_lock(BLOCK); >> >> As I understand this may lead to an outright deadlock between bind >> and unbind. >> >> I stumbled upon this code while debugging what seems like deadlocks >> in our test FS installation put under some heavy load. The load >> is about >> 25 new event subscribers per second (destroyed few seconds later) and >> about 700 events per second. >> >> Regards, >> Karol > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org Karol -- e-mail: Karol.Golab at tls.pl www: http://www.tls.pl signature: not found company: TLS-Technologie Sp??ka z ograniczon? odpowiedzialno?ci? z siedzib? w Warszawie przy ul. Polnej 50 00-644; KRS 0000073337 (XII Wydzia? Gospodarczy KRS); NIP 526-25-78-091; VAT ID PL5262578091; REGON 017349442; kap. zak. 50000,00 z? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111128/ddb01fb8/attachment.html From anthony.minessale at gmail.com Mon Nov 28 23:26:25 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Nov 2011 14:26:25 -0600 Subject: [Freeswitch-dev] Possible deadlock in src/switch_event.c ? In-Reply-To: <4ED3EB63.4020203@tls.pl> References: <4ED346F8.6030604@tls.pl> <4ED3EB63.4020203@tls.pl> Message-ID: you can do whatever you wish. I was just warning you that nobody has really pushed event binding and unbinding in such a way before which may explain how you uncovered that issue. Usually things that bind stay resident for a long time. 2011/11/28 Karol Golab > ** > > Well, I have a Perl script run as part of a dialplan and the script > creates an EventConsumer. Thus I have one bind / unbind pair for each > connection. Is this somehow wrong / against what the EventConsumer API was > planned for? > How could I have a single bind/unbind in this case? > > As for choosing event socket and C application vs a script - what is the > main difference / reason to switch: performance of some other issues? My > CPU seems to have some cycles to spare. > > Regards, > Karol > > > > On 11/28/2011 05:36 PM, Anthony Minessale wrote: > > you're probably right but I wonder what you are doing that is binding and > unbinding so much? > The bind and unbind was designed to be called once when the module is > loaded and called again when it exits. > > My fear is you are using the embedded scripting event handers in some > quick repeating fashion. > > I did a commit to flip that case around but I don't encourage you to use > embedded scripts for heavy event traffic, that's what event socket is for > and if you have the C skills to debug this, you might want to try a C app ;) > > > 2011/11/28 Brian West > >> Please post it to jira.freeswitch.org >> >> /b >> >> On Nov 28, 2011, at 2:31 AM, Karol Golab wrote: >> >> >> Hi! >> >> I've noticed some strange code in src/switch_event.c - could someone >> smarter (& knowing the mentioned code) please take a look? >> >> The problem is in the order of calls to locks BLOCK & RWLOCK - it >> differs between functions: >> >> switch_event_bind_removable >> switch_mutex_lock(BLOCK); >> switch_thread_rwlock_wrlock(RWLOCK); >> >> switch_event_unbind_callback >> switch_thread_rwlock_wrlock(RWLOCK); >> switch_mutex_lock(BLOCK); >> >> switch_event_unbind >> switch_thread_rwlock_wrlock(RWLOCK); >> switch_mutex_lock(BLOCK); >> >> As I understand this may lead to an outright deadlock between bind >> and unbind. >> >> I stumbled upon this code while debugging what seems like deadlocks >> in our test FS installation put under some heavy load. The load is about >> 25 new event subscribers per second (destroyed few seconds later) and >> about 700 events per second. >> >> Regards, >> Karol >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org > > > > Karol > > -- > e-mail: Karol.Golab at tls.pl > www: http://www.tls.pl > signature: not found > > company: TLS-Technologie Sp??ka z ograniczon? odpowiedzialno?ci? z siedzib? w Warszawie przy ul. Polnej 50 00-644; KRS 0000073337 (XII Wydzia? Gospodarczy KRS); NIP 526-25-78-091; VAT ID PL5262578091; REGON 017349442; kap. zak. 50000,00 z? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111128/e66b4d07/attachment-0001.html From karol at tls.pl Mon Nov 28 23:33:02 2011 From: karol at tls.pl (Karol Golab) Date: Mon, 28 Nov 2011 21:33:02 +0100 Subject: [Freeswitch-dev] Possible deadlock in src/switch_event.c ? In-Reply-To: References: <4ED346F8.6030604@tls.pl> <4ED3EB63.4020203@tls.pl> Message-ID: <4ED3EFFE.6040705@tls.pl> Thanks! The idea of using EventConsumer per connection seemed pretty natural for me - thus my surprise at your warning as I'd expect that it's a standard way to use events. Well, it shows I'm a newbie to FS ;) Regards, Karol On 11/28/2011 09:26 PM, Anthony Minessale wrote: > you can do whatever you wish. I was just warning you that nobody has > really pushed event binding and unbinding in such a way before which > may explain how you uncovered that issue. Usually things that bind > stay resident for a long time. > > > 2011/11/28 Karol Golab > > > > Well, I have a Perl script run as part of a dialplan and the > script creates an EventConsumer. Thus I have one bind / unbind > pair for each connection. Is this somehow wrong / against what the > EventConsumer API was planned for? > How could I have a single bind/unbind in this case? > > As for choosing event socket and C application vs a script - > what is the main difference / reason to switch: performance of > some other issues? My CPU seems to have some cycles to spare. > > Regards, > Karol > > > > On 11/28/2011 05:36 PM, Anthony Minessale wrote: >> you're probably right but I wonder what you are doing that is >> binding and unbinding so much? >> The bind and unbind was designed to be called once when the >> module is loaded and called again when it exits. >> >> My fear is you are using the embedded scripting event handers in >> some quick repeating fashion. >> >> I did a commit to flip that case around but I don't encourage you >> to use embedded scripts for heavy event traffic, that's what >> event socket is for and if you have the C skills to debug this, >> you might want to try a C app ;) >> >> >> 2011/11/28 Brian West > > >> >> Please post it to jira.freeswitch.org >> >> >> /b >> >> On Nov 28, 2011, at 2:31 AM, Karol Golab wrote: >> >>> >>> Hi! >>> >>> I've noticed some strange code in src/switch_event.c - >>> could someone >>> smarter (& knowing the mentioned code) please take a look? >>> >>> The problem is in the order of calls to locks BLOCK & >>> RWLOCK - it >>> differs between functions: >>> >>> switch_event_bind_removable >>> switch_mutex_lock(BLOCK); >>> switch_thread_rwlock_wrlock(RWLOCK); >>> >>> switch_event_unbind_callback >>> switch_thread_rwlock_wrlock(RWLOCK); >>> switch_mutex_lock(BLOCK); >>> >>> switch_event_unbind >>> switch_thread_rwlock_wrlock(RWLOCK); >>> switch_mutex_lock(BLOCK); >>> >>> As I understand this may lead to an outright deadlock >>> between bind >>> and unbind. >>> >>> I stumbled upon this code while debugging what seems like >>> deadlocks >>> in our test FS installation put under some heavy load. The >>> load is about >>> 25 new event subscribers per second (destroyed few seconds >>> later) and >>> about 700 events per second. >>> >>> Regards, >>> Karol >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> 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_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org > > > Karol > > -- > e-mail:Karol.Golab at tls.pl > www:http://www.tls.pl > signature: not found > > company: TLS-Technologie Sp??ka z ograniczon? odpowiedzialno?ci? z siedzib? w Warszawie przy ul. Polnej 50 00-644; KRS 0000073337 (XII Wydzia? Gospodarczy KRS); NIP 526-25-78-091; VAT ID PL5262578091; REGON 017349442; kap. zak. 50000,00 z? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org Karol -- e-mail: Karol.Golab at tls.pl www: http://www.tls.pl signature: not found company: TLS-Technologie Sp??ka z ograniczon? odpowiedzialno?ci? z siedzib? w Warszawie przy ul. Polnej 50 00-644; KRS 0000073337 (XII Wydzia? Gospodarczy KRS); NIP 526-25-78-091; VAT ID PL5262578091; REGON 017349442; kap. zak. 50000,00 z? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111128/597b0623/attachment-0001.html From daleiliu at gmail.com Tue Nov 29 07:37:01 2011 From: daleiliu at gmail.com (Dalei Liu) Date: Mon, 28 Nov 2011 20:37:01 -0800 Subject: [Freeswitch-dev] unofficial build 1.0.8beta1dl Message-ID: Hi, I am testing the release procedure and generated a few binary packages based on the recent git repository. The build is base on a fork on github freeswitch project, which seems synchronized with official freeswitch repository. You may find the packages for debian5, debian 6, centos 5, centos 6 (all 32-bit), and win32 here: https://github.com/daleiliu/FreeSWITCH/downloads md5sum: f24dea4da2d0be74c7b7db2e77156213 freeswitch-1.0.8beta1dl-centos5.tar 9961e0bb1d0cafea653cc8c3ae498b4b freeswitch-1.0.8beta1dl-centos6.tar d8fe7034ae0aa8df9812155cb9c3e663 freeswitch-1.0.8beta1dl-debian5.tar ab5d934d79709b6f961882e5a2cc3d7c freeswitch-1.0.8beta1dl-debian6.tar f18d2511c82d67098477f817a1fc7ddf freeswitch-1.0.8beta1dl-win32.zip I tagged this version as 1.0.8beta1dl in the b1.0dl branch of my github branch. But please be aware that this is NOT an official freeswitch release, but just a personal experiment. This cut is simply based on the time I started building them and not fully tested. I hope it would be a static reference point for testing and debugging. Appreciate if you have comments or suggestions. Dalei Liu From msc at freeswitch.org Tue Nov 29 21:37:31 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Nov 2011 10:37:31 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Tomorrow: Special Guest Message-ID: Hello all! I just wanted to give everyone a heads up about tomorrow's conference call. We have a special guest scheduled to join us: Jean-Marc Valin from Mozilla! Jean-Marc is a codec guru who is working diligently on the OPUS codec. He will be coming to talk to us about the codec and to answer any questions that you may have. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_30 We look forward to having you on our call tomorrow! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111129/752c0b03/attachment.html From msc at freeswitch.org Tue Nov 29 22:05:26 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Nov 2011 11:05:26 -0800 Subject: [Freeswitch-dev] condition check... In-Reply-To: References: Message-ID: Bhrugu, Did you ever figure this out? Most likely you need to take a step back and look at the big picture. In FreeSWITCH you don't normally need to have a bunch of if-then type structures. What is the problem you are trying to solve? Chances are there is a much more elegant solution than trying to emulate these crude GotoIf statements. -MC On Thu, Nov 24, 2011 at 4:57 AM, Bhrugu Mehta wrote: > hi, all > > asterisk setup: > exten => _X.,1,GotoIf($[${status1}=on]?n1:n2) > exten => _X.,n(n1),GotoIf($[${status2}=off]?n3:n4) > exten => _X.,(n2),GotoIf($[${status3}=X]?n5:n6) > ... > ... > > How do i implement in Freeswitch... > > Regards, > > -- > Bhrugu Mehta > Sr. S/W Engineer > VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) > India > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111129/0e2893f7/attachment.html From anthony.minessale at gmail.com Wed Nov 30 01:01:12 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Nov 2011 16:01:12 -0600 Subject: [Freeswitch-dev] condition check... In-Reply-To: References: Message-ID: Ha, guess who wrote the gotoif app....? =p its unnecessary in the FreeSWITCH paradigm you can just the XML dialplan. On Thu, Nov 24, 2011 at 6:57 AM, Bhrugu Mehta wrote: > hi, all > > asterisk setup: > exten => _X.,1,GotoIf($[${status1}=on]?n1:n2) > exten => _X.,n(n1),GotoIf($[${status2}=off]?n3:n4) > exten => _X.,(n2),GotoIf($[${status3}=X]?n5:n6) > ... > ... > > How do i implement in Freeswitch... > > Regards, > > -- > Bhrugu Mehta > Sr. S/W Engineer > VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) > India > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111129/190aedbd/attachment.html From msc at freeswitch.org Wed Nov 30 01:13:34 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Nov 2011 14:13:34 -0800 Subject: [Freeswitch-dev] condition check... In-Reply-To: References: Message-ID: On Tue, Nov 29, 2011 at 2:01 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Ha, guess who wrote the gotoif app....? =p > > I guess we need to add that one to http://cluecon.com/anthm.html. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111129/3678f537/attachment.html From sholapuram at gmail.com Wed Nov 30 16:42:53 2011 From: sholapuram at gmail.com (sham) Date: Wed, 30 Nov 2011 19:12:53 +0530 Subject: [Freeswitch-dev] Not able to compile FS Message-ID: Hello all, i am not able to compile the FS on my ubuntu server 10.04 http://pastebin.freeswitch.org/17896 can anyone help? -- Thanks and Regards, Sham V. Kota, Mobile No. (+91)9970244889 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111130/1bcae5e4/attachment.html From krice at freeswitch.org Wed Nov 30 19:13:19 2011 From: krice at freeswitch.org (Ken Rice) Date: Wed, 30 Nov 2011 10:13:19 -0600 Subject: [Freeswitch-dev] Not able to compile FS In-Reply-To: Message-ID: The pastebin is pretty obvious that you are trying to build H323 support and didn?t installed the pre-requisites... On 11/30/11 7:42 AM, "sham" wrote: > Hello all, > > i am not able to compile the FS on my ubuntu server 10.04 > > http://pastebin.freeswitch.org/17896 > > can anyone help? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111130/ec94f461/attachment.html From msc at freeswitch.org Wed Nov 30 20:12:21 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Nov 2011 09:12:21 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: Don't forget: Jean-Marc Valin from Mozilla will be in to discuss the OPUS codec today! Here's the agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_30 Talk to you soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20111130/eebb7633/attachment.html