Re: [Freeswitch-users] Skypiax, same skype user, multiple channels
On Tue, Feb 24, 2009 at 10:33 AM, Eric Chamberlain wrote: > I was reading through the Skypiax documentation and saw the comment > that it's not possible to run multiple skype clients on the same linux > machine, all using the same skype user account. > > It's possible to run multiple skype clients with the same skype user > account, as long as the skype clients are not accessing the same Skype > dbpath. > > We use runuser to run multiple skype clients. All the clients use the > same skype user, but each instance uses a different home directory, > each with its own .Skype folder. > > In such a configuration, will Skypiax support multiple channels using > the same skype username? Hi Eric, yes, definitely yes. If you give me more details I would like to integrate this use case both in the docs and in my testings. BTW: I'm about to move on your previous *very useful* suggestions and feature requests, please continue to send it :-) Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 > > -- > Eric Chamberlain, Founder > RF.com - http://RF.com/ > > > > > > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH
On Mon, Feb 23, 2009 at 5:26 PM, Carlos Talbot wrote: > Were you planning to check in the sample skype.conf.xml into the default > FreeSWITCH conf folder? If so, just be aware the default config causes > freeswitch to hang right after a "load mod_skypiax" (if you do not have > skype running or specify a nonexistant skype user). Carlos, many thanks for reporting! I'll fix this this evening, if you have time to file a Jira for it would be wonderful. ciao for now, giovanni > > regards, > > > Carlos > On Thu, Feb 19, 2009 at 7:32 PM, Giovanni Maruzzelli > wrote: >> >> On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot >> wrote: >> >> > One question I have, is ringback suppose to work with mod_skypiax? >> > Whenever >> > I dial a number I get a few seconds of dead air before the call is >> > answered. >> > I've tried adding ringback and transfer_ringback into the dialplan just >> > before the bridge command but no go. Am I missing something? Thanks. >> >> Carlos, >> >> ringback now works without tricks, and Skypiax is in trunk. >> >> Both remote ringing and early media are treated as remote ringing >> right now (eg: no early media, just ringing). >> >> I'll add early media support in the near future. >> >> Thanks a lot for testing and exercising skypiax, and please let me >> know any hint, suggestion, feature request, etc >> >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> = >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot >> wrote: >> > Giovannia, >> > >> > great work on mod_skypiax. I've been testing it under Windows and it >> > sounds >> > great including PSTN calls. I plan to include it as part of the Windows >> > MSI >> > build. >> > >> > One question I have, is ringback suppose to work with mod_skypiax? >> > Whenever >> > I dial a number I get a few seconds of dead air before the call is >> > answered. >> > I've tried adding ringback and transfer_ringback into the dialplan just >> > before the bridge command but no go. Am I missing something? Thanks. >> > >> > regards, >> > >> > Carlos >> > >> > >> > >> > >> > ___ >> > Freeswitch-users mailing list >> > Freeswitch-users@lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] New build gives error message for default grammar file??
Hey Brian, Where abouts do you keep the Window MSI 1.0.3 build that isn't in SVN trunk. Installing from the wiki installation page gets me a build with the same error. Thanks. Mark. -Original Message- From: Brian West To: freeswitch-users@lists.freeswitch.org Sent: Tue, 24 Feb 2009 11:45 am Subject: Re: [Freeswitch-users] New build gives error message for default grammar file?? You're not in 1.0.3 you're in SVN trunk... The reason I know this is that wasn't changed till AFTER 1.0.3 was tagged and a new file set was used... please go into your libs dir and wipe out pocketsphinx and sphinx base.. then let it redownload them. ?I'll make you a new tarball of the new grammar files which are in the jsgf format. ?An example was added to scripts yes_no.gram ... word of warning the new pocketsphinx loads the entire dictionary on port open which on a machine of any speed should load the entire thing in 2 seconds or less... but here is the WARNING WARNING WARNING... If you happen to use words that aren't in the dictionary pocketsphinx WILL CRASH.. I am already on this with the dev's to work out a fix for this its been a long standing bug apparently in the lib. /b On Feb 24, 2009, at 1:36 PM, mszla...@aol.com wrote: I'm getting this error message trying out the pizza demo in FS 1.0.3: ?"Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. = ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] howto originate fs call from webapp (python)
On Tue, Feb 24, 2009 at 1:09 PM, Alexander de Greiff
wrote:
> hi all,
>
> i come from asterisk an i am new to freeswitch. after my with days with
> freeswitch i am very excited!
Welcome to FreeSWITCH!
>
> but trying to migrate our deployment i have three challenges. one of them is:
>
> i need to call freeswitch from a webapp (e.g. python) and pass number1 and
> number2. i then need freeswitch to call number1. as soon as it is picked up
> say a short confirmaton text, call number2 and bridge the two.
>
> my first approach was to call via xml_rpc like described in the wiki but when
> i call like
>
> server.freeswitch.api("originate","sofia/gategay/gateway1/{number1}
> &bridge(sofia/gateway/gateway2/{number2})")
>
> but in this case both numbers are called in parallel and the first number to
> pick up gets a ringback tone until the other number picks up. how can i get
> the sequence described above?
>
> thanks for your help
> alex
Do you have any other requirements? For example, what happens if the
first bridge fails? Does your Python app need to "do anything"? Just
curious.
Thanks,
MC
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Re: [Freeswitch-users] howto originate fs call from webapp (python)
> my first approach was to call via xml_rpc like described in the wiki
> but when i call like
>
> server.freeswitch.api("originate","sofia/gategay/gateway1/{number1}
> &bridge(sofia/gateway/gateway2/{number2})")
>
> but in this case both numbers are called in parallel and the first
> number to pick up gets a ringback tone until the other number picks
> up. how can i get the sequence described above?
>
> thanks for your help alex
You are probably getting early media when dialing number 1. Try :
server.freeswitch.api("originate","{ignore_early_media=true}sofia/gategay/gateway1/{number1}
&bridge(sofia/gateway/gateway2/{number2})")
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[Freeswitch-users] howto originate fs call from webapp (python)
hi all,
i come from asterisk an i am new to freeswitch. after my with days with
freeswitch i am very excited!
but trying to migrate our deployment i have three challenges. one of them is:
i need to call freeswitch from a webapp (e.g. python) and pass number1 and
number2. i then need freeswitch to call number1. as soon as it is picked up say
a short confirmaton text, call number2 and bridge the two.
my first approach was to call via xml_rpc like described in the wiki but when i
call like
server.freeswitch.api("originate","sofia/gategay/gateway1/{number1}
&bridge(sofia/gateway/gateway2/{number2})")
but in this case both numbers are called in parallel and the first number to
pick up gets a ringback tone until the other number picks up. how can i get the
sequence described above?
thanks for your help
alex
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Re: [Freeswitch-users] Web-based forum?
On Tue, Feb 24, 2009 at 4:19 PM, Michael Collins wrote: > > Maybe this question has been raised before, but if not: There's so > > much traffic in this mailing list that I was wondering if adding a > > web-based forum on the site was in the works? > > We are upgrading the freeswitch.org site soon to drupal 6.9. We are > considering turning on the forum feature there. No definitive decision > has been made but this request has come in several times. However, we > are trying to make it so that the devs don't have yet another place to > have to monitor for user questions, etc. so we will need to figure out > a way to make it easy to use for the experts... -1 for a forum. > > > -MC > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Arnaldo M Pereira ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] New build gives error message for default grammar file??
Hi Brian, It sounds like I'd be better off with 1.0.3 than SVN and will waiting for the fix? But thanks for the files and info. Mark. -Original Message- From: Brian West To: freeswitch-users@lists.freeswitch.org Sent: Tue, 24 Feb 2009 11:45 am Subject: Re: [Freeswitch-users] New build gives error message for default grammar file?? You're not in 1.0.3 you're in SVN trunk... The reason I know this is that wasn't changed till AFTER 1.0.3 was tagged and a new file set was used... please go into your libs dir and wipe out pocketsphinx and sphinx base.. then let it redownload them. ?I'll make you a new tarball of the new grammar files which are in the jsgf format. ?An example was added to scripts yes_no.gram ... word of warning the new pocketsphinx loads the entire dictionary on port open which on a machine of any speed should load the entire thing in 2 seconds or less... but here is the WARNING WARNING WARNING... If you happen to use words that aren't in the dictionary pocketsphinx WILL CRASH.. I am already on this with the dev's to work out a fix for this its been a long standing bug apparently in the lib. /b On Feb 24, 2009, at 1:36 PM, mszla...@aol.com wrote: I'm getting this error message trying out the pizza demo in FS 1.0.3: ?"Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. = ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Recording and outbound rtp
no, I'm matching the incoming sip call via the destination number in my public context and executing the javascript appliaction. This app directly answers the call and records it until the user hangs up. D- - Original Message - From: "Anthony Minessale" To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording and outbound rtp is it during a bridged call? On Tue, Feb 24, 2009 at 11:49 AM, Dan < freeswitch-us...@digitaldan.com > wrote: Hi, I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great. The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch. Thanks! Dan- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/ PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] New build gives error message for default grammar file??
You're not in 1.0.3 you're in SVN trunk... The reason I know this is that wasn't changed till AFTER 1.0.3 was tagged and a new file set was used... please go into your libs dir and wipe out pocketsphinx and sphinx base.. then let it redownload them. I'll make you a new tarball of the new grammar files which are in the jsgf format. An example was added to scripts yes_no.gram ... word of warning the new pocketsphinx loads the entire dictionary on port open which on a machine of any speed should load the entire thing in 2 seconds or less... but here is the WARNING WARNING WARNING... If you happen to use words that aren't in the dictionary pocketsphinx WILL CRASH.. I am already on this with the dev's to work out a fix for this its been a long standing bug apparently in the lib. /b On Feb 24, 2009, at 1:36 PM, mszla...@aol.com wrote: I'm getting this error message trying out the pizza demo in FS 1.0.3: "Can't open dictionary C:\Program Files\FreeSWITCH\grammar \default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] New build gives error message for default grammar file??
http://www.bkw.org/pizza_gram.tar.gz /b On Feb 24, 2009, at 1:36 PM, mszla...@aol.com wrote: I'm getting this error message trying out the pizza demo in FS 1.0.3: "Can't open dictionary C:\Program Files\FreeSWITCH\grammar \default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] New build gives error message for default grammar file??
I'm getting this error message trying out the pizza demo in FS 1.0.3: ?"Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Web-based forum?
The web version of this list is available at: http://www.nabble.com/Freeswitch-users-f32209.html Mike On Feb 24, 2009, at 2:08 PM, Fred wrote: > Hello > > Maybe this question has been raised before, but if not: There's so > much traffic in this mailing list that I was wondering if adding a > web-based forum on the site was in the works? > > Cheers, > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Web-based forum?
> Maybe this question has been raised before, but if not: There's so > much traffic in this mailing list that I was wondering if adding a > web-based forum on the site was in the works? We are upgrading the freeswitch.org site soon to drupal 6.9. We are considering turning on the forum feature there. No definitive decision has been made but this request has come in several times. However, we are trying to make it so that the devs don't have yet another place to have to monitor for user questions, etc. so we will need to figure out a way to make it easy to use for the experts... -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Web-based forum?
Hello Maybe this question has been raised before, but if not: There's so much traffic in this mailing list that I was wondering if adding a web-based forum on the site was in the works? Cheers, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Recording and outbound rtp
is it during a bridged call? On Tue, Feb 24, 2009 at 11:49 AM, Dan wrote: > Hi, > > I have a small javascript application that accepts a call, retrieves some > dtmf digits and then records the call to an icecast server. This works > great. > > The problem I'm having is that when the call is being recorded freeswitch > is no longer sending rtp packets back to the originating caller, in my case > a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, > since no voice data back is being generated. Unfortunately my Cisco gear > has rtp inactivity timers set up to hang up a call after 3 minutes of no > incoming rtp packets, this is a global setting that cannot be configured for > a single dial peer. Does anyone have a suggestion to generate rtp packets > every once in a while? I tried setting comfort noise which did not seem to > send anything. I could try playing a empty/short wav file every minute or > so but the javascript call session.record is blocking, would a traditional > javascript timer and callback to play a wav file be my best bet or is there > a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian > etch. > > Thanks! > Dan- > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] file directory.conf.xml
On Tue, Feb 24, 2009 at 9:24 AM, Ali Al-Rubaie wrote: > Hi, > > The file directory.conf.xml had been mentioned in the documentation many > times but there is not such file in the conf folder. Do you mean default.xml > in directory folder? > > Thanks! Can you tell me where you see that file name listed? It's possible that it should be "dialplan_directory.conf.xml" but I don't know for sure. I will check it out. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Recording and outbound rtp
Hi, I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great. The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch. Thanks! Dan- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] new ilbc lib
The problem comes up that the default is 30... the chances are that your phone doesn't set the mode= line so we default to 30 when this takes place. Not setting the mode= line in the FMTP usually means 30ms... which is the default. So to force this always to 30 you can allow i...@30i, because if you invite to me with 20 and I 200 ok you 30.. you are to use 30 no exceptions. Most phones do not obey this rule. /b On Feb 24, 2009, at 11:28 AM, Alex Gusak wrote: > Hello. > > After upgrade to version 1.0.3 we have a problem with the codec iLBC > (I think that this is due to the transition to a new ilbc libs 1 week > ago). > Very poor quality for calls to the codec iLBC mode=20 (crack in the > dynamic). iLBC mode=30 works well. > Tested with phones and Zoiper SJPhone. > > After a rollback to the old version of FreeSWITCH 1.0.2 this is not a > problem, iLBC works fine in both modes (mode = 20 and mode = 30). > > What could be the problem? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] new ilbc lib
Hello. After upgrade to version 1.0.3 we have a problem with the codec iLBC (I think that this is due to the transition to a new ilbc libs 1 week ago). Very poor quality for calls to the codec iLBC mode=20 (crack in the dynamic). iLBC mode=30 works well. Tested with phones and Zoiper SJPhone. After a rollback to the old version of FreeSWITCH 1.0.2 this is not a problem, iLBC works fine in both modes (mode = 20 and mode = 30). What could be the problem? -- Alex Gusak ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] file directory.conf.xml
Hi, The file directory.conf.xml had been mentioned in the documentation many times but there is not such file in the conf folder. Do you mean default.xml in directory folder? Thanks! --- On Tue, 2/24/09, freeswitch-users-requ...@lists.freeswitch.org wrote: From: freeswitch-users-requ...@lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 32, Issue 181 To: freeswitch-users@lists.freeswitch.org Date: Tuesday, February 24, 2009, 3:34 AM Send Freeswitch-users mailing list submissions to freeswitch-users@lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-requ...@lists.freeswitch.org You can reach the person managing the list at freeswitch-users-ow...@lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: SIP dump to DB (kokoska.rokoska) 2. FREESwitch on Windows Server 2003 (Stephen Walker) 3. Re: mod_erlang_event compile problem (Andrew Thompson) 4. Re: FREESwitch on Windows Server 2003 (Carlos Talbot) 5. Re: SIP dump to DB (Joseph Bajin) 6. Re: SIP dump to DB (kokoska.rokoska) 7. Re: mod_portaudio: Do not accept next call after Hangup (Rene Pankratz) 8. Patch for openzap concerning finding a free channel. (Helmut Kuper) -- Message: 1 Date: Mon, 23 Feb 2009 23:32:26 +0100 From: "kokoska.rokoska" Subject: Re: [Freeswitch-users] SIP dump to DB To: freeswitch-users@lists.freeswitch.org Message-ID: <49a323fa.8000...@post.cz> Content-Type: text/plain; charset=ISO-8859-1 Joseph Bajin napsal(a): > Basically, you are trying to build what Empirix has with their Hammer tool. > Thank you very much, Joseph, for your interest! I have never heard about Empirix (I'll look at it), but what I'm trying to build is something like SER/Kamailio/OpenSIPS sip_trace module. > You can create an application that is basically a mix of tshark and a > database feeder. > You sniff with tshark and going to basically pipe it to another > application that will read the pcap file, parse it, and load it into the > db for you. There are plenty of modules out there that will read pcap > for you. > Thank you once more, Joseph, for suggestion! I think about it - it will be challenge for me to write robust and still fast enough (thousands messages per second) SIP parser + DB feeder :-) Best regards, kokoska.rokoska -- Message: 2 Date: Mon, 23 Feb 2009 14:47:13 -0800 From: "Stephen Walker" Subject: [Freeswitch-users] FREESwitch on Windows Server 2003 To: Message-ID: <3b93e0500b57d04cbae85520b750cff04ca...@exchange.sonasearch.com> Content-Type: text/plain; charset="us-ascii" Hello: I have successfully loaded the Windows implementation (SVN 11602 - 02/02/09) from your site and it runs fine. I configured a Linksys SPA 2102 and have acquired dial tone and the '999X' tests work. I have not been able to establish connection with either FreeWorldDialup or Broadvoice as of yet. Which files do I need to edit and what are the proper entries to enable connection to FreeWorldDialup and Broadvoice? Example files and where they reside in the file structure would be very much appreciated. Thank you All the Best, Steve Steve Walker President SONASEARCH, INC 425/883-1984 NOTICE: The information contained in this document is intended by Sonasearch, Inc. or one of its subsidiaries for the use of the named individuals or entities to which it is addressed and may contain information that is privileged or otherwise confidential. It is not intended for transmission to, or receipt by, any individual or entity other than the named addressee (or a person authorized to deliver it to the named addressee) except as otherwise expressly permitted in this document. If you have received this document in error, please destroy it without copying or forwarding it, and notify the sender of the error by calling Sonasearch at (425) 883-1984. -- next part -- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/7d514817/attachment-0001.html -- Message: 3 Date: Mon, 23 Feb 2009 19:22:08 -0500 From: Andrew Thompson Subject: Re: [Freeswitch-users] mod_erlang_event compile problem To: freeswitch-users@lists.freeswitch.org Message-ID: <20090224002207.gf13...@hijacked.us> Content-Type: text/plain; charset=us-ascii Leon, I think I found the problem. I shouldn't have been defaulting to binding to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the module to actually bind to 0.0.0.0 correctly and made it the default in the config file. Erlan
Re: [Freeswitch-users] mod_erlang_event compile problem
On Tue, Feb 24, 2009 at 10:49:24AM +0100, Leon de Rooij wrote: > Well, this works, I feel a bit stupid now :-] Now it's time to play > with it.. > Nah, bad choice of defaults on my part. Defaulting to 0.0.0.0 is much more consistant and compatible. For some reason I was trying to emulate the event socket, not an erlang node. Thanks for finally making me solve the problem instead of just working around it. Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup
What direction is the original call? Are you sure you do not have the auto_answer enabled? On Tue, Feb 24, 2009 at 1:27 AM, Rene Pankratz < r.pankr...@fh-wolfenbuettel.de> wrote: > No, unfortunately the problem still persists. Portaudio still > automatically accepts/takes the next call. > > René > > On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz > > wrote: > > > >> Hello, > >> when hanging up a call with portaudio automatically the next call that > >> is incoming or held is accepted. > >> Is it possible to configure PA that way, that after hanging up (doesn't > >> matter whether caller or callee) no call is activated automatically? I > >> want to choose if I accept the next call or not. > >> > >> Thanks in advance > >>René > >> > >> > > Just following up - did this get resolved? > > -MC > > > > ___ > > Freeswitch-users mailing list > > Freeswitch-users@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup
Please report this bug to jira.freeswitch.org. On Feb 24, 2009, at 2:27 AM, Rene Pankratz wrote: > No, unfortunately the problem still persists. Portaudio still > automatically accepts/takes the next call. > > René >> On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz >> wrote: >> >>> Hello, >>> when hanging up a call with portaudio automatically the next call >>> that >>> is incoming or held is accepted. >>> Is it possible to configure PA that way, that after hanging up >>> (doesn't >>> matter whether caller or callee) no call is activated >>> automatically? I >>> want to choose if I accept the next call or not. >>> >>> Thanks in advance >>> René >>> >>> >> Just following up - did this get resolved? >> -MC >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Skypiax, same skype user, multiple channels
I was reading through the Skypiax documentation and saw the comment that it's not possible to run multiple skype clients on the same linux machine, all using the same skype user account. It's possible to run multiple skype clients with the same skype user account, as long as the skype clients are not accessing the same Skype dbpath. We use runuser to run multiple skype clients. All the clients use the same skype user, but each instance uses a different home directory, each with its own .Skype folder. In such a configuration, will Skypiax support multiple channels using the same skype username? -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_fax and sending a fax
Hi, the clue for sending fax is to use the originate command in the CLI: originate sofia/example/1...@10.10.10.10 &txfax(/path_to_fax_file) this command will send the fax file via profile example to fax machine 100 reachable via 10.10.10.10 Hope this could help others :p regards, rod. Javier Aristizábal wrote: > Hi Rod, i just play with rx_fax and work for me. I didn't work with > tx_fax but i understand, that you need a .tiff file to send > passthrough the rx_fax. Maybe that can help you > > regards > javar > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_erlang_event compile problem
Well, this works, I feel a bit stupid now :-] Now it's time to play with it.. Thanks a lot ! kind regards, Leon On Feb 24, 2009, at 1:22 AM, Andrew Thompson wrote: > Leon, > > I think I found the problem. I shouldn't have been defaulting to > binding > to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the > module to actually bind to 0.0.0.0 correctly and made it the default > in > the config file. Erlang nodes by default bind to 0.0.0.0, so I decided > to make mod_erlang_event follow suit. > > Please give that a shot and see if it fixes things. > > Andrew > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_erlang_event compile problem
Andrew, I think you're right, packets are indeed sent to 172.31.0.13 while mod_erlang_event is listening at 127.0.0.1 ! Why didn't I see that ! ;-) Will test it now and let you know how it goes.. regards, Leon On Feb 24, 2009, at 1:22 AM, Andrew Thompson wrote: > Leon, > > I think I found the problem. I shouldn't have been defaulting to > binding > to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the > module to actually bind to 0.0.0.0 correctly and made it the default > in > the config file. Erlang nodes by default bind to 0.0.0.0, so I decided > to make mod_erlang_event follow suit. > > Please give that a shot and see if it fixes things. > > Andrew > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_openzap stops working after some calls Update
Hello, just to keep you informed about this problem. As mentioned I added a hack to free allocated channels depending on last event time. I enhanced oz dump as well to display "last event time" and "InUse"-Flag. What I found is this: 1. InUse channel flag wasn't set for inbound calls. I fixed that as far as I understood the openzap code ;) and I tested the patch successfully for 7 days now... 2. In my setup (AVAYA as remote end for a E1) channels tend to hang in a state <> DOWN after terminating a call. Then I found TOMANYCALLS entries in FS log. I had to restart FS resp. openzap module. The hack I added is in production and works for 7 days now. No channels hanging anymore. Of course, the hack is not the final solution, but it seems to solve at least my problems in production until openzap has state timers. If the board wants, I can upload the hack as well. regards Helmut On 12.02.2009 14:03, Helmut Kuper wrote: > Hi Mike, > > at least for incoming calls this shouldn't be too brutal, cause far > end seems to know that the channel should be free otherwise it never > would allocate it. By now the hack works at least for me quite good. > Nobody from AVAYA side moaned about it, yet. But I have to wait one or > two further days to be sure ... I guess I have to talk to stkn in irc > to get an idea how long I have to use it. > > regards > helmut ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Patch for openzap concerning finding a free channel.
Hello, today I uploaded a little patch for openzap into trunk (r667). It marks now inbound channels as "inUse" which is conform with outbound channel handling. This should solve some problems finding a free channel in ozmod_isdn.c for inbound and outbound calls. regards Helmut ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
