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Can I restrict a gateway to 1 call at a time?
by Scott Ellis 22 Jan '09

22 Jan '09

2 1

mod_managed failing to load on CentOS 5.2
by Tim B 22 Jan '09

22 Jan '09

I changed the filename of the dll to FreeSWITCH.Managed.dll then tried to restart. FS now no longer starts. Says mono error.... with a dump. I don't have the exact message because I am not on location with the machine. I know it does compile, load and execute on a windows machine. Just not on Centos. > From: freeswitch-users-request(a)lists.freeswitch.org> Subject: Freeswitch-users Digest, Vol 31, Issue 77> To: freeswitch-users(a)lists.freeswitch.org> Date: Thu, 15 Jan 2009 00:20:53 -0800> > Send Freeswitch-users mailing list submissions to> freeswitch-users(a)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-request(a)lists.freeswitch.org> > You can reach the person managing the list at> freeswitch-users-owner(a)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. mod_managed failing to load on CentOS 5.2 (Tim B)> 2. Re: Using mod_managed Linux/Mono 2.02 (Michael Giagnocavo)> 3. zapata.conf immediate=yes in Asterisk - Freeswitch> equivalent? (Scott Ellis)> 4. Country specific tones - how to contribute? (Scott Ellis)> 5. Re: Country specific tones - how to contribute? (Jason White)> 6. Changes in PlayAndGetDigits (Juan Backson)> 7. Re: OpenZAP parse error [-3012] [Q931E_INVALID_CRV] (Peter P GMX)> > > ----------------------------------------------------------------------> > Message: 1> Date: Wed, 14 Jan 2009 20:13:27 -0500> From: Tim B <timb0311(a)hotmail.com>> Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2> To: <freeswitch-users(a)lists.freeswitch.org>> Message-ID: <BAY117-W5065D9F4AAF59BB1B86359DED70(a)phx.gbl>> Content-Type: text/plain; charset="windows-1252"> > > Got mod_managed compiled and installed. Now it isn't loading. See below...> > > 1) Donwloaded fresh from SVN> > 2) Compiled... and installed.. OK> [root@phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig> [root@phone2 mod_managed]# make> [root@phone2 mod_managed]# make install> > 3) Added to modules.conf.xml :> <load module="mod_managed"/>> > 4) Started freeswitch from command line ... Error:> 2009-01-14 20:01:42 [ERR] mod_managed.cpp:220 loadRuntime() mono_domain_assembly_open failed.> 2009-01-14 20:01:42 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_managed.so> **Module load routine returned an error**> > 5) I know mono2 is working because I compiled and executed a helloworld test class on machine.> > Any ideas?> > > > _________________________________________________________________> Windows Live?: Keep your life in sync. > http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_012…> -------------- next part --------------> An HTML attachment was scrubbed...> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114… > > ------------------------------> > Message: 2> Date: Thu, 15 Jan 2009 00:34:20 -0500> From: Michael Giagnocavo <mgg(a)giagnocavo.net>> Subject: Re: [Freeswitch-users] Using mod_managed Linux/Mono 2.02> To: "freeswitch-users(a)lists.freeswitch.org"> <freeswitch-users(a)lists.freeswitch.org>> Message-ID:> <6E8D2069C08AA84A83D336E996AE4C670235BBB97F(a)mse17be1.mse17.exchange.ms>> > Content-Type: text/plain; charset="us-ascii"> > The managed assembly should be the same on both platforms. The correct name is FreeSWITCH.Managed.dll. I'll get a patch to the mod_managed/managed/Makefile.> > Meanwhile, simply renaming mod_managed_lib.dll should work.> > After that, make sure there's a "managed" subdirectory where the modules are.> > -Michael> > > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Adam Long> Sent: Wednesday, January 14, 2009 3:45 PM> To: freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] Using mod_managed Linux/Mono 2.02> > Has anyone had any luck using mod_managed under linux with mono yet?> The Wiki looks to still be lacking some linux installation instructions.> I feel like I'm close but missing something simple.> > I got as far as adding "languages/mod_managed" to the /usr/src/freeswitch-1.0.2/modules.conf without quotes obviously.> > My installed mono version is> [root@sipcore-alpha mod]# mono -V> Mono JIT compiler version 2.2 (tarball Wed Jan 14 09:44:57 PST 2009)> Copyright (C) 2002-2008 Novell, Inc and Contributors. www.mono-project.com> TLS: __thread> GC: Included Boehm (with typed GC)> SIGSEGV: altstack> Notifications: epoll> Architecture: x86> Disabled: none> > I can successful compile freeswitch and it indeed compiles mod_managed.so> > I added <load module="mod_managed" />> to my /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml> > I did also create the /usr/local/freeswitch/mod/managed directory as stated in the wiki as requirement.> > But when I start freeswitch I get the following in regards to the mod_managed loading...> > 2009-01-14 14:19:12 [INFO] mod_managed.cpp:309 mod_managed_load() Loading mod_managed (Common Language Infrastructure), Mono Version> 2009-01-14 14:19:12 [INFO] mod_managed.cpp:213 loadRuntime() Calling mono_assembly_loaded.> 2009-01-14 14:19:12 [INFO] mod_managed.cpp:217 loadRuntime() Calling mono_domain_assembly_open.> 2009-01-14 14:19:12 [ERR] mod_managed.cpp:220 loadRuntime() mono_domain_assembly_open failed.> 2009-01-14 14:19:12 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_managed.so> **Module load routine returned an error**> > One thing I think I may be missing is a FreeSWITCH.Managed.dll (this exists on windows environment but doesn't seem to be compiled under linux)> I thought perhaps mod_managed_lib.dll was the linux equivalent but that exists and still no a no go.> > Any ideas would be very welcome? Thank you!> > > > Regards,> -Adam> > > > > -------------- next part --------------> An HTML attachment was scrubbed...> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115… > > ------------------------------> > Message: 3> Date: Thu, 15 Jan 2009 17:50:30 +1100> From: Scott Ellis <scott.ellis(a)novatex.com.au>> Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk -> Freeswitch equivalent?> To: "freeswitch-users(a)lists.freeswitch.org"> <freeswitch-users(a)lists.freeswitch.org>> Message-ID: <496EDCB6.4020802(a)novatex.com.au>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed> > Searched the wiki and mailing lists as best I can, but with no luck.> > How do I get OpenZap to answer a call immediately? (I do not need caller id)> > Scott> > > > > > ------------------------------> > Message: 4> Date: Thu, 15 Jan 2009 18:16:13 +1100> From: Scott Ellis <scott.ellis(a)novatex.com.au>> Subject: [Freeswitch-users] Country specific tones - how to> contribute?> To: "freeswitch-users(a)lists.freeswitch.org"> <freeswitch-users(a)lists.freeswitch.org>> Message-ID: <496EE2BD.2050102(a)novatex.com.au>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed> > I have tracked down a set of au tones from the mailing list, which I am > going to verify. How do I go about getting these added into the default > build so that they are available for all in future?> > I tried <action application="set" data="ringback=${au-ring}"/> and this > did not work - where does it try and load the ring tone from? I have > entries in the tones.conf file, but these do not seem to be used.> > Scott> > > > > > > ------------------------------> > Message: 5> Date: Thu, 15 Jan 2009 18:24:05 +1100> From: Jason White <jason(a)jasonjgw.net>> Subject: Re: [Freeswitch-users] Country specific tones - how to> contribute?> To: freeswitch-users(a)lists.freeswitch.org> Message-ID: <20090115072405.GA15789(a)jdc.jasonjgw.net>> Content-Type: text/plain; charset=us-ascii> > Scott Ellis <scott.ellis(a)novatex.com.au> wrote:> > I have tracked down a set of au tones from the mailing list, which I am > > going to verify. How do I go about getting these added into the default > > build so that they are available for all in future?> > Maybe by posting a patch to the bug tracking system or the development list?> > > > I tried <action application="set" data="ringback=${au-ring}"/> and this > > did not work - where does it try and load the ring tone from? I have > > entries in the tones.conf file, but these do not seem to be used.> > us-ring and uk-ring are defined in vars.xml. Note that they are global> variables, referenced with the $${variable-name} syntax.> > There's an ITU document referred to on the wiki with the official definitions> of ringback and other tones for various countries.> > > > > ------------------------------> > Message: 6> Date: Thu, 15 Jan 2009 15:43:20 +0800> From: "Juan Backson" <juanbackson(a)gmail.com>> Subject: [Freeswitch-users] Changes in PlayAndGetDigits> To: freeswitch-users(a)lists.freeswitch.org> Message-ID:> <27c25bc40901142343l34a3e99ftecf0df971e8e32f6(a)mail.gmail.com>> Content-Type: text/plain; charset=ISO-8859-1> > Hi,> > Is there a change in the playAndGetDigits api? In the old release,> 11102, my lua script is working but is not working in the latest> release.> The error I am getting is " Error in playAndGetDigits expected 10..10> args, got 9 ".> > Thanks,> JB> > > > ------------------------------> > Message: 7> Date: Thu, 15 Jan 2009 09:20:18 +0100> From: Peter P GMX <Prometheus001(a)gmx.net>> Subject: Re: [Freeswitch-users] OpenZAP parse error [-3012]> [Q931E_INVALID_CRV]> To: freeswitch-users(a)lists.freeswitch.org> Message-ID: <496EF1C2.8020607(a)gmx.net>> Content-Type: text/plain; charset=ISO-8859-1> > Hello Michael,> > how much $$ are we talking about? I need this issue to be solved quickly> and it's worth to spend some money.> > I've read the following post:> > http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg05792.…> and have the same symptom with "after hundreds of calls I start to get b> channels that are stuck in states like "TERMINATING" or "HANGUP""> > Best regards> Peter> > Michael Collins schrieb:> > I believe these are all symptoms of something that Stefan is working> > on: better Q931 timers. It's been on the todo list for some time but> > we've had absolutely NOBODY willing to pony up serious $$ to support> > OpenZAP development which means it is progressing at the speed of> > developers' free time.> >> > -MC> >> > On Wed, Jan 14, 2009 at 9:44 AM, Peter P GMX <Prometheus001(a)gmx.net> wrote:> > > >> After a time I receive the following error when a call comes in on our> >> OpenZap span 2:> >> parse error [-3012] [Q931E_INVALID_CRV]> >>> >> Here's the log> >> 2009-01-14 13:14:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got> >> an event! Type:[4d] Size:[103] CRV: 23 (0x17, CTX: Originator)> >> 2009-01-14 13:14:11 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0> >> (-1:-1) source isdn_data->channels_remote_crv[0x17]> >> 2009-01-14 13:14:11 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received> >> Release with no matching channel 0> >> 2009-01-14 13:14:11 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse> >> error [-3012] [Q931E_INVALID_CRV]> >> 2009-01-14 13:14:15 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5> >> --------------------------------------------------------------------------------> >>> >> When freeswitch is restarted or mod_openzap is reloaded, the error is> >> gone away.> >>> >> Any idea what this can be?> >>> >> Best regards> >> Peter> >>> >>> >> _______________________________________________> >> Freeswitch-users mailing list> >> Freeswitch-users(a)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(a)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(a)lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org> > > End of Freeswitch-users Digest, Vol 31, Issue 77> ************************************************ _________________________________________________________________ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009

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zapata.conf immediate=yes in Asterisk - Freeswitch equivalent?
by Scott Ellis 22 Jan '09

22 Jan '09

Searched the wiki and mailing lists as best I can, but with no luck. How do I get OpenZap to answer a call immediately? (I do not need caller id) Scott

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firing events from javascript - working example needed
by Stephen Crosby 21 Jan '09

21 Jan '09

I noticed the wiki has an example of sending a custom event from javascript: http://wiki.freeswitch.org/wiki/Javascript_Event , but I can't make it work. It doesn't fail or cause an error. But I never see an event on my listener script. Can someone confirm that this example does in fact work? or provide me with one that does? --Stephen

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sample code of how to use the C level API instead of dialplan
by Dave 21 Jan '09

21 Jan '09

Just like to know whether there is any kind of document available for FS integration at its C level, not at its dialplan level. The reason I ask this is that some of the local process simply impossible to be integrated with the Dialplan, but should not be that hard if the C level API is available. Any URL or suggestion is welcome. Thanks. ___________________________________________________________ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.com.

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Re: [Freeswitch-users] Recording ULAW files
by freeswitch-users＠digitaldan.com 21 Jan '09

21 Jan '09

Thanks, that's what I'm currently doing now. ----- Original Message ----- From: "Anthony Minessale" <anthony.minessale(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 21, 2009 12:28:19 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording ULAW files The default audio framework operates on raw audio. The mod_native_file triggers a special flag that tells the higher level api's for recording not to transcode the audio first. It would probably be easier for you to use the api_hangup_hook variable to trigger a sox command to wrap the files in a wav or use a batch process in cron to do so than to try to figure it out in the code. On Wed, Jan 21, 2009 at 1:05 PM, < freeswitch-users(a)digitaldan.com > wrote: Thanks for the quick reply. I'm new to this project so I'm not familiar with the inner workings just yet but at looking at mod_native_file.c it seems this is a thin wrapper around the switch's own file input and output routines? Would it be best to change this class or register a new file type, like .ul? If so, where would be a good starting point. ----- Original Message ----- From: "Anthony Minessale" < anthony.minessale(a)gmail.com > To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 21, 2009 10:56:45 AM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording ULAW files no, there is currently no way to do that. It would be feasible to add an option to mod_native_file to write wav headers around the raw data but it has not been attempted. On Wed, Jan 21, 2009 at 10:48 AM, < freeswitch-users(a)digitaldan.com > wrote: Hi, I'm recording files using the pcmu extension in order to save them in the g.711 ulaw format, which is what everything in my network uses. It appears that the recorded file is just raw data without a header. Is there any way to save this as a wav type with a header (keeping the ulaw format)? for example, running the unix command 'file' on the recording prints: /tmp/185065_f7bb8e0c-e641-11dd-800d-5ffe41c540dd.PCMU: data if I run it through sox which just outputs the same data (ulaw,8000,mono) but specifying the type as wav /usr/bin/sox -t .ul -r 8000 -c 1 -b -U file.pcmu -t wav -r 8000 -c 1 -b -U file.wav it produces a file that shows: /tmp/185065_f7bb8e0c-e641-11dd-800d-5ffe41c540dd.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz if I try and save it within freeswitch using the wav extension, it trans-codes it to a pcm format RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Is there any way to have freeswitch record the file as ulaw with the RIFF wav header? Thanks! _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)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_minessale@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)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(a)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_minessale@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Recording ULAW files
by freeswitch-users＠digitaldan.com 21 Jan '09

21 Jan '09

Hi, I'm recording files using the pcmu extension in order to save them in the g.711 ulaw format, which is what everything in my network uses. It appears that the recorded file is just raw data without a header. Is there any way to save this as a wav type with a header (keeping the ulaw format)? for example, running the unix command 'file' on the recording prints: /tmp/185065_f7bb8e0c-e641-11dd-800d-5ffe41c540dd.PCMU: data if I run it through sox which just outputs the same data (ulaw,8000,mono) but specifying the type as wav /usr/bin/sox -t .ul -r 8000 -c 1 -b -U file.pcmu -t wav -r 8000 -c 1 -b -U file.wav it produces a file that shows: /tmp/185065_f7bb8e0c-e641-11dd-800d-5ffe41c540dd.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz if I try and save it within freeswitch using the wav extension, it trans-codes it to a pcm format RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Is there any way to have freeswitch record the file as ulaw with the RIFF wav header? Thanks!

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Re: [Freeswitch-users] How to bridge without Answer? (Anthony Minessale)
by Kareem Hamdy 21 Jan '09

21 Jan '09

Hello everyone: I think what Anthony wants is (please excuse me if I'm wrong - but what I'm assuming is) a call to come in - let's say that its DID goes to person A. He wants to ring person A, let person A pick up, and then bridge the call. When working at an Asterisk VoIP vendor, I had a call in which a gentleman wanted just that. I think they paid for incoming calls or something. Anthony, please let us know if that's accurate. Thanks, Kareem -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Wednesday, January 21, 2009 6:54 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 31, Issue 125 Send Freeswitch-users mailing list submissions to freeswitch-users(a)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-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)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: How to bridge without Answer? (Anthony Minessale) 2. Re: firing events from javascript - working example needed (Michael Collins) 3. Re: Problem with digium te220p (Krzysztof Zimnicki) 4. Re: Problem with digium te220p (Michael Collins) 5. Re: Hang up not received (Michael Collins) 6. ATA-answering machine question/recommendation (jonathan augenstine) ---------------------------------------------------------------------- Message: 1 Date: Wed, 21 Jan 2009 07:56:51 -0600 From: Anthony Minessale <anthony.minessale(a)gmail.com> Subject: Re: [Freeswitch-users] How to bridge without Answer? To: freeswitch-users(a)lists.freeswitch.org Message-ID: <191c3a030901210556n2d443179n17d8bbb9ed24b8ab(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" You can't. Why would you need that? Are you trying to forward inbound calls from the pstn to an ivr without answering them? That could get you in trouble FYI. On Wed, Jan 21, 2009 at 7:40 AM, shehzad p <pmhshz(a)gmail.com> wrote: > > Hi all, > > When I dial a number from Originator Gateway, It will route to Freeswitch > Server and then FS will bridge the call to Terminator Gateway as below. > Terminator Answer the call (and runs playback, and look for DTMF). > > |Originator Gateway|---------------> |FreeSwitch |------------------> > |Terminator Gateway| > > I used bridge application to route call to Terminator. > But my requirement is that when Terminator answer the call (Respnd with > 200OK) , Freeswitch should NOT Answer call for A leg (Originater Gateway). > > How can be this done? > > Thanks in advance. > msp. > -- > View this message in context: > http://www.nabble.com/How-to-bridge-without-Answer--tp21583334p21583334.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)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_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Hang up not received
by Tomás 21 Jan '09

21 Jan '09

Hi all, I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch. I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN. When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing... I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log. I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up) I've googled the list, and I've found several people with a similar problem but no solution... That's my pastebin with the most importants printouts and config files: http://pastebin.freeswitch.org/6822 Thank you very much in advance.

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Problem with digium te220p
by Krzysztof Zimnicki 21 Jan '09

21 Jan '09

>Can you join irc later today? I will be on as mercutioviz. I would >like to discuss this more. > >-MC >Sent from my iPhone Sorry, i can't join to irc. Can you put your questions here? I'll try to answer. Our CallCenter have strange situation, because now is working on Asterisk and we can only put this card in other machine after 22 pm. Thanks.

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