From jason at jasonjgw.net Thu Jan 1 00:19:28 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 1 Jan 2009 19:19:28 +1100 Subject: [Freeswitch-users] fs_cli help command In-Reply-To: <20090101072045.GA12582@jdc.jasonjgw.net> References: <20090101072045.GA12582@jdc.jasonjgw.net> Message-ID: <20090101081928.GA13138@jdc.jasonjgw.net> This patch implements the /help idea, but I'm still not sure that it's the right solution. -------------- next part -------------- A non-text attachment was scrubbed... Name: fs_cli.patch Type: text/x-diff Size: 555 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090101/8d78ac83/attachment.bin From Prometheus001 at gmx.net Thu Jan 1 03:28:51 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 01 Jan 2009 12:28:51 +0100 Subject: [Freeswitch-users] uuid_playback In-Reply-To: References: <495B8190.2080207@gmx.net> Message-ID: <495CA8F3.4010203@gmx.net> Here's the link http://wiki.freeswitch.org/wiki/Mod_commands#uuid_playback Brian West schrieb: > Wiki link please. > > /b > > On Dec 31, 2008, at 8:28 AM, Peter P GMX wrote: > > >> As I see on the Wiki page uuid_playback seems to be implemented, >> however >> it doesn't work on the console or via event_socket. >> Also in the code I could not find it (svn 10438). >> >> So for now I use uuid_brodcast to play announcements to one or both >> parties. >> >> Question: What is the status of uuid_playback? >> >> Best regards >> Peter >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Jan 1 06:41:26 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 1 Jan 2009 08:41:26 -0600 Subject: [Freeswitch-users] uuid_playback In-Reply-To: <495CA8F3.4010203@gmx.net> References: <495B8190.2080207@gmx.net> <495CA8F3.4010203@gmx.net> Message-ID: <0E427671-E836-455B-87CF-DC99B42B3EF0@freeswitch.org> Fixed. /b On Jan 1, 2009, at 5:28 AM, Peter P GMX wrote: > Here's the link > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_playback > > Brian West schrieb: >> Wiki link please. >> >> /b >> >> On Dec 31, 2008, at 8:28 AM, Peter P GMX wrote: >> >> >>> As I see on the Wiki page uuid_playback seems to be implemented, >>> however >>> it doesn't work on the console or via event_socket. >>> Also in the code I could not find it (svn 10438). >>> >>> So for now I use uuid_brodcast to play announcements to one or both >>> parties. >>> >>> Question: What is the status of uuid_playback? >>> >>> Best regards >>> Peter >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From excelsio at gmx.net Thu Jan 1 03:27:56 2009 From: excelsio at gmx.net (excelsio at gmx.net) Date: Thu, 01 Jan 2009 12:27:56 +0100 Subject: [Freeswitch-users] encryption infos needed Message-ID: <20090101112756.181740@gmx.net> Hi, we want to enhance our old Siemens Hicom 300 and replace it step by step. Therefore we decided to try out opensource solutions ourselves. One requirement is that the solution has to encrypt all data. So try let?s look at Asterisk was our first thought. Well, there seem to be unoffical patches for Asterisk 1.4.x with SRTP/SIPS support. So, unofficial. With 1.6.x the support for it hasn?t been fully integrated, yet. So, what?s next out there? => freeswitch But what about encryption support? SRTP is end to end encryption between phones, SIPS is used for the encryption of signaling "hop-by-hop", well which hop? Talking about encryption, it seems there are many different scenarios to consider, which looks like they couldn?t encrypted? Let?s look at our planed setup public telefon network <--ISDN/S2M--> Patton 4960 <--ISDN/S2M--> Siemens Hicom 300 Patton 4960 <--IP--> freeswitch <--IP--> Snom 320 SIP Provider <--IP-- freeswitch <--IP-- Snom 320 1. Incoming calls shoud be reached via landline: [e.g. telefon network --ISDN/S2M--> Patton 4960 --IP--> freeswitch --IP--> Snom 320 users] So, what about encryption between the Patton 4960, the freeswitch and der Snom 320? Is it possible to encrypt the whole path? Well, how? Is it supported with freeswitch? 2. Outcoming calls should go to a SIP provider which supports sip trunking and DDI, well SIPconnect: [e.g. SIP Provider <--SIP trunk-- freeswitch <--IP-- Snom 320 users] Same question here: What about encryption between the Patton 4960, the freeswitch and der Snom 320? Is it possible to encrypt the whole path? Well, how? Is it supported with freeswitch? 2.1 Outcoming calls should be forwarded locally, if the SIP trunk between the SIP provider and the freeswitch server fails [e.g. telefon network <-- ISDN/S2M-- Patton 4960 <--IP-- freeswitch <--IP-- Snom 320 users] Same question here: What about encryption between the Patton 4960, the freeswitch and der Snom 320? Is it possible to encrypt the whole IP path? Well, how? Is it supported with freeswitch? 3. The next thing is the encryption of voice and signaling data in general. Does the freeswitch solution support this? I think it?s an end to end encryption between the users? As freeswitch seems to play a proxy part, I guess yes? [e.g. freeswitch <--IP--> Snom 320 users <--SRTP/SIPS --> Snom 320] 4. Another problem is the encryption of the voice and signaling data between our LAN and the SIP provider. Is it possible to encrypt all data between those with the freeswitch solution? Do I need something additionally? [e.g SIP Provider <--encrypted SIP trunk ??? --> freeswitch] So what can be done with freeswitch? What else can be done support all scenarios above? -- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger From excelsio at gmx.net Thu Jan 1 04:49:05 2009 From: excelsio at gmx.net (excelsio at gmx.net) Date: Thu, 01 Jan 2009 13:49:05 +0100 Subject: [Freeswitch-users] Direct inward dialling Message-ID: <20090101124905.245010@gmx.net> Hallo Peter, could you please post your complete working config for DID and QSC? thanks in advance Tim -- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger From yudha2008 at gmail.com Fri Jan 2 01:17:17 2009 From: yudha2008 at gmail.com (Baskar) Date: Fri, 2 Jan 2009 14:47:17 +0530 Subject: [Freeswitch-users] busy tone detection In-Reply-To: <965B5A92-4182-489B-8AA0-6E6173C1931B@jerris.com> References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com> <87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com> <51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com> <2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org> <965B5A92-4182-489B-8AA0-6E6173C1931B@jerris.com> Message-ID: *Hi Michael Jerris,* *1. When we call a busy number wont it detect busy signal?* *2. If we get the SIP response code for busy as 486 can we detect the tone?* *3. And more over in the client side only we are using softphone, the other end is connected with E1 in the audiocode, so since it seems to be a e1 line here.* *The connection is like this,* *softphone---> freeswitch----> audiocode---> e1 line.* * Thanks with Regards, * *N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/c89cf14b/attachment-0001.html From oseslija at gmail.com Fri Jan 2 02:32:16 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Fri, 2 Jan 2009 11:32:16 +0100 Subject: [Freeswitch-users] encryption infos needed In-Reply-To: <20090101112756.181740@gmx.net> References: <20090101112756.181740@gmx.net> Message-ID: <4468a6770901020232x67066259jdcd40c78c796a446@mail.gmail.com> Hello, I'm using FreeSWITCH mostly as a PBX for multi tenants. Secure calling is supported fully by FreeSWITCH and to my knowledge it is the only open-source solution where it works w/o any hacks or tweaks. Current major brand of phones supporting SRTP and TLS that I've tested are Linksys and Snom. I'm told on the IRC channel that there are more working. FreeSWITCH as an SIP B2BUA can be configured to offer SRTP in SDP negogitation in the B-leg (just use export sip_secure_media param). That means that if a phone and the other softswitch/gateway supports SRTP you can have whole path encrypted. I have tried following scenario: Linkys phone calling 9888 ext which is a conference server at conference.freeswitch.org, so we have Linksys SRTP -> FS SRTP -> other FS SRTP (whole media path between a phone and two FS servers encrypted). The only question left to answer is does Patton offer SRTP/TLS. FreeSWITCH won't be an issue here. Regards, Ognjen On Thu, Jan 1, 2009 at 12:27 PM, wrote: > Hi, > > we want to enhance our old Siemens Hicom 300 and replace it step by step. > Therefore we decided to try out opensource solutions ourselves. One > requirement > is that the solution has to encrypt all data. So try let?s look at Asterisk > was > our first thought. Well, there seem to be unoffical patches for Asterisk > 1.4.x > with SRTP/SIPS support. So, unofficial. With 1.6.x the support for it > hasn?t > been fully integrated, yet. > > So, what?s next out there? => freeswitch > But what about encryption support? > SRTP is end to end encryption between phones, SIPS is used for the > encryption of signaling "hop-by-hop", well which hop? > Talking about encryption, it seems there are many different scenarios to > consider, which looks like they couldn?t encrypted? > > Let?s look at our planed setup > > public telefon network <--ISDN/S2M--> Patton 4960 <--ISDN/S2M--> Siemens > Hicom 300 > Patton 4960 <--IP--> freeswitch <--IP--> > Snom 320 > SIP Provider <--IP-- freeswitch <--IP-- Snom > 320 > > 1. Incoming calls shoud be reached via landline: > > [e.g. telefon network --ISDN/S2M--> Patton 4960 --IP--> freeswitch --IP--> > Snom > 320 users] > > So, what about encryption between the Patton 4960, the freeswitch and der > Snom > 320? Is it possible to encrypt the whole path? Well, how? Is it supported > with > freeswitch? > > > 2. Outcoming calls should go to a SIP provider which supports sip trunking > and > DDI, well SIPconnect: > > [e.g. SIP Provider <--SIP trunk-- freeswitch <--IP-- Snom 320 > users] > > Same question here: > What about encryption between the Patton 4960, the freeswitch and der Snom > 320? > Is it possible to encrypt the whole path? Well, how? Is it supported with > freeswitch? > > > 2.1 Outcoming calls should be forwarded locally, if the SIP trunk between > the > SIP provider and the freeswitch server fails > > [e.g. telefon network <-- ISDN/S2M-- Patton 4960 <--IP-- freeswitch <--IP-- > Snom > 320 users] > > Same question here: > What about encryption between the Patton 4960, the freeswitch and der Snom > 320? Is > it possible to encrypt the whole IP path? Well, how? Is it supported with > freeswitch? > > > 3. The next thing is the encryption of voice and signaling data in general. > Does the freeswitch solution support this? I think it?s an end to end > encryption > between the users? As freeswitch seems to play a proxy part, I guess yes? > > [e.g. freeswitch <--IP--> Snom 320 users <--SRTP/SIPS --> Snom 320] > > > > 4. Another problem is the encryption of the voice and signaling data > between > our LAN and the SIP provider. Is it possible to encrypt all data between > those > with the freeswitch solution? Do I need something additionally? > > [e.g SIP Provider <--encrypted SIP trunk ??? --> freeswitch] > > > So what can be done with freeswitch? What else can be done support all > scenarios above? > -- > Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: > http://www.gmx.net/de/go/multimessenger > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/2fd4a097/attachment.html From fidibus83 at aol.com Fri Jan 2 03:30:23 2009 From: fidibus83 at aol.com (fidibus83) Date: Fri, 2 Jan 2009 12:30:23 +0100 Subject: [Freeswitch-users] (no subject) Message-ID: <00fd01c96ccd$810ce050$6445310a@Franzi> Hello! Can FS do something like an ACD? I have configure a Supportqueue like this: And my Agents are in a Call Group. How can I transfer the call to a free agent? I saw the example in mod_fifo. But my Agent can?t wait in the queue all the time. Is it possible that the agents are called back when a call is waiting in the queue? Thanks. Regards, fidibus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/e38acd53/attachment.html From fidibus83 at aol.com Fri Jan 2 03:31:27 2009 From: fidibus83 at aol.com (fidibus83) Date: Fri, 2 Jan 2009 12:31:27 +0100 Subject: [Freeswitch-users] ACD - Queue and Call Group Message-ID: <010501c96ccd$a7459320$6445310a@Franzi> Hello! Can FS do something like an ACD? I have configure a Supportqueue like this: And my Agents are in a Call Group. How can I transfer the call to a free agent? I saw the example in mod_fifo. But my Agent can?t wait in the queue all the time. Is it possible that the agents are called back when a call is waiting in the queue? Thanks. Regards, fidibus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/0d177bcb/attachment.html From yudha2008 at gmail.com Fri Jan 2 06:58:11 2009 From: yudha2008 at gmail.com (Baskar) Date: Fri, 2 Jan 2009 20:28:11 +0530 Subject: [Freeswitch-users] Conference In-Reply-To: <914fc92a0811121154w219ce22ekb31b6d8ce3abe8d8@mail.gmail.com> References: <914fc92a0811121154w219ce22ekb31b6d8ce3abe8d8@mail.gmail.com> Message-ID: Hi, Conference is working well in recent days, but suddenly it in not working when i transfer the call ,the call get disconnected with the voice file playing Bye....... What is the error? When i transfer the call why the call get hangup api originate sofia/internal/1003%172.20.191.227 &bridge(sofia/default/ 79841799874 at 172.20.191.228) Content-Type: api/response Content-Length: 41 +OK b98ddfca-6e84-4945-a02c-e7d455b58c37 api uuid_transfer b98ddfca-6e84-4945-a02c-e7d455b58c37 -both 3003 Content-Type: api/response Content-Length: 4 +OK Correct me were i am Wrong. -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/309764eb/attachment-0001.html From swjinjin at volans.kr Fri Jan 2 07:18:04 2009 From: swjinjin at volans.kr (Sangwoo Jin) Date: Sat, 3 Jan 2009 00:18:04 +0900 Subject: [Freeswitch-users] The transferred call is not hanged up. Message-ID: <012b01c96ced$4f291070$ed7b3150$@kr> I have met the non-hanged up call after being transferred. The following is the test scenario. 1. A makes a call to B 2. B answers the call of A and holds the call of A. 3. B makes a call to C and transfers the call of A to C before C answers the call of B. 4. A hangs up the call before C answers the call of A. In this state, C's phone is ringing endless. Why is not the channel of C hanged up? Thanks for reading my mail. Sangwoo Jin. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090103/7c2d950e/attachment.html From intralanman at freeswitch.org Fri Jan 2 07:41:51 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 02 Jan 2009 15:41:51 +0000 Subject: [Freeswitch-users] fs_cli help command In-Reply-To: <20090101081928.GA13138@jdc.jasonjgw.net> References: <20090101072045.GA12582@jdc.jasonjgw.net> <20090101081928.GA13138@jdc.jasonjgw.net> Message-ID: <495E35BF.1050501@freeswitch.org> Jason White wrote: > This patch implements the /help idea, but I'm still not sure that it's the > right solution. > jira would be a better place to put this -Ray From msc at freeswitch.org Fri Jan 2 08:30:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 2 Jan 2009 08:30:25 -0800 Subject: [Freeswitch-users] fs_cli help command In-Reply-To: <495E35BF.1050501@freeswitch.org> References: <20090101072045.GA12582@jdc.jasonjgw.net> <20090101081928.GA13138@jdc.jasonjgw.net> <495E35BF.1050501@freeswitch.org> Message-ID: <87f2f3b90901020830k56ab5fd1r8db6dc4500e3509e@mail.gmail.com> Good point, Ray. FYI, MikeJ fixed this already. "Help" will do a FreeSWITCH help and "/help" will do a fs_cli help. -MC On Fri, Jan 2, 2009 at 7:41 AM, Raymond Chandler wrote: > Jason White wrote: > > This patch implements the /help idea, but I'm still not sure that it's > the > > right solution. > > > jira would be a better place to put this > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/c019f43a/attachment.html From mike at jerris.com Fri Jan 2 09:53:04 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 2 Jan 2009 12:53:04 -0500 Subject: [Freeswitch-users] busy tone detection In-Reply-To: References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com> <87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com> <51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com> <2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org> <965B5A92-4182-489B-8AA0-6E6173C1931B@jerris.com> Message-ID: On Jan 2, 2009, at 4:17 AM, Baskar wrote: > Hi Michael Jerris, > 1. When we call a busy number wont it detect busy signal? > not if your getting a response code and trying to detect a tone. > 2. If we get the SIP response code for busy as 486 can we detect the > tone? > A 486 response code is not a tone, so there is no tone to detect. > 3. And more over in the client side only we are using softphone, the > other end is connected with E1 in the audiocode, so since it seems > to be a e1 line here. > I would guess on an e1 pri you would never get a tone. > The connection is like this, > > softphone---> freeswitch----> audiocode---> e1 line. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/1a991335/attachment.html From gminet-ml at mcit.be Fri Jan 2 02:59:24 2009 From: gminet-ml at mcit.be (=?ISO-8859-1?Q?Ga=EBtan_Minet?=) Date: Fri, 2 Jan 2009 11:59:24 +0100 Subject: [Freeswitch-users] MOH and bypass media Message-ID: Hi, As a newcomer on Freeswitch, first of all thanks you for all your work on it. I'm currently evaluating it as a replacement for * and really appreciate working (ok, playing) with it. One of our requirements is to optimize the network flows as much as possible, and as such avoid any unnecessary rtp proxying. When a call is in bypass media and one of the phones puts the call on hold, the remote phone doesn't play moh. Of course I guess this is expected in bypass media, and moh should be the responsibility of the UA just like in a pure sip proxy environment ( so the phone that holds the line should for example invite the moh server in a new dialog and swap sdp with the remote phone or something similar). Snom phones do this just fine when you configure the moh server uri in the phone, but others like Thomsons ST2030 do not have that setting (... I know, real crap, but we already deployed tons of them so we cannot ignore them...) . So when a Thomson puts another phone on hold, there is no music at all. I'd like to achieve something similar to what asterisk does with (re)invites during the hold. I read about and tried the resume-media- on-hold option and this works fine, the phone correctly gets the moh. However, when the call is unheld, the media stream continues in media mode and I'd like to avoid this as much as possible. Is there a way to return to bypass media mode automatically when the call is unheld ? BTW, is the resume-media-on-hold really implemented for server moh support, or as a more generic way to put a call back in media mode (so we can listen for inband/rfc dtmf for example) ? I also noticed that the remote phone only sees/display the hold state when in bypass media mode. I guess no reinvite (a=sendonly) at all is sent to the held party when in media mode ? Again, is there a solution for this ? Many thanks ! Kind regards Gaetan From gminet-ml at mcit.be Fri Jan 2 02:59:24 2009 From: gminet-ml at mcit.be (=?ISO-8859-1?Q?Ga=EBtan_Minet?=) Date: Fri, 2 Jan 2009 11:59:24 +0100 Subject: [Freeswitch-users] MOH and bypass media Message-ID: Hi, As a newcomer on Freeswitch, first of all thanks you for all your work on it. I'm currently evaluating it as a replacement for * and really appreciate working (ok, playing) with it. One of our requirements is to optimize the network flows as much as possible, and as such avoid any unnecessary rtp proxying. When a call is in bypass media and one of the phones puts the call on hold, the remote phone doesn't play moh. Of course I guess this is expected in bypass media, and moh should be the responsibility of the UA just like in a pure sip proxy environment ( so the phone that holds the line should for example invite the moh server in a new dialog and swap sdp with the remote phone or something similar). Snom phones do this just fine when you configure the moh server uri in the phone, but others like Thomsons ST2030 do not have that setting (... I know, real crap, but we already deployed tons of them so we cannot ignore them...) . So when a Thomson puts another phone on hold, there is no music at all. I'd like to achieve something similar to what asterisk does with (re)invites during the hold. I read about and tried the resume-media- on-hold option and this works fine, the phone correctly gets the moh. However, when the call is unheld, the media stream continues in media mode and I'd like to avoid this as much as possible. Is there a way to return to bypass media mode automatically when the call is unheld ? BTW, is the resume-media-on-hold really implemented for server moh support, or as a more generic way to put a call back in media mode (so we can listen for inband/rfc dtmf for example) ? I also noticed that the remote phone only sees/display the hold state when in bypass media mode. I guess no reinvite (a=sendonly) at all is sent to the held party when in media mode ? Again, is there a solution for this ? Many thanks ! Kind regards Gaetan From gminet-ml at mcit.be Fri Jan 2 08:35:44 2009 From: gminet-ml at mcit.be (=?ISO-8859-1?Q?Ga=EBtan_Minet?=) Date: Fri, 2 Jan 2009 17:35:44 +0100 Subject: [Freeswitch-users] bind_meta_app with sip info dtmf in bypass media mode Message-ID: Hi, It looks like that even when using sip info dtmf, the bind_meta_app doesn't react on them when rtp is in bypass media mode. Of course rfc2833/inband cannot be processed in bypass media mode, but doesn't sip info still go through freeswitch ?. Is this a limitation of the current implementation, or simply not possible due to some technical limitations ? Or is there any option to control this ? I'd like to implement (dtmf-triggered) server-side features but still use bypass media mode. Thank you Kind regards Gaetan From mike at jerris.com Fri Jan 2 09:55:32 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 2 Jan 2009 12:55:32 -0500 Subject: [Freeswitch-users] (no subject) In-Reply-To: <00fd01c96ccd$810ce050$6445310a@Franzi> References: <00fd01c96ccd$810ce050$6445310a@Franzi> Message-ID: <491E1798-26E7-48C9-950C-8BEBA40BB928@jerris.com> Yes, callback agents are now part of mod_fifo as well. On Jan 2, 2009, at 6:30 AM, fidibus83 wrote: > Hello! > > Can FS do something like an ACD? > > I have configure a Supportqueue like this: > > > > > > > > > And my Agents are in a Call Group. How can I transfer the call to a > free agent? > > I saw the example in mod_fifo. But my Agent can?t wait in the queue > all the time. Is it possible that the agents are called back when a > call is waiting in the queue? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/01b87bda/attachment-0001.html From mike at jerris.com Fri Jan 2 09:56:44 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 2 Jan 2009 12:56:44 -0500 Subject: [Freeswitch-users] Conference In-Reply-To: References: <914fc92a0811121154w219ce22ekb31b6d8ce3abe8d8@mail.gmail.com> Message-ID: On Jan 2, 2009, at 9:58 AM, Baskar wrote: > Hi, > > Conference is working well in recent days, but suddenly it in not > working when i transfer the call ,the call get disconnected with the > voice file playing Bye....... > > What is the error? > When i transfer the call why the call get hangup > > api originate sofia/internal/1003%172.20.191.227 &bridge(sofia/default/79841799874 at 172.20.191.228 > ) > > Content-Type: api/response > Content-Length: 41 > > +OK b98ddfca-6e84-4945-a02c-e7d455b58c37 > api uuid_transfer b98ddfca-6e84-4945-a02c-e7d455b58c37 -both 3003 > > Content-Type: api/response > Content-Length: 4 > > +OK > > Correct me were i am Wrong. Look at the debug log to see what happens. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/0b86103e/attachment.html From brian at freeswitch.org Fri Jan 2 09:57:13 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Jan 2009 11:57:13 -0600 Subject: [Freeswitch-users] MOH and bypass media In-Reply-To: References: Message-ID: Please see: in sip_profiles/internal.xml /b PS you can't get out of the media path when you unhold... FS will be in the media path hold button moving forward unless you use the uuid_media command to turn it back off. On Jan 2, 2009, at 4:59 AM, Ga?tan Minet wrote: > Hi, > > As a newcomer on Freeswitch, first of all thanks you for all your work > on it. I'm currently evaluating it as a replacement for * and really > appreciate working (ok, playing) with it. > > One of our requirements is to optimize the network flows as much as > possible, and as such avoid any unnecessary rtp proxying. > > When a call is in bypass media and one of the phones puts the call on > hold, the remote phone doesn't play moh. > Of course I guess this is expected in bypass media, and moh should be > the responsibility of the UA just like in a pure sip proxy environment > ( so the phone that holds the line should for example invite the moh > server in a new dialog and swap sdp with the remote phone or something > similar). > > Snom phones do this just fine when you configure the moh server uri in > the phone, but others like Thomsons ST2030 do not have that setting > (... I know, real crap, but we already deployed tons of them so we > cannot ignore them...) . So when a Thomson puts another phone on hold, > there is no music at all. > > I'd like to achieve something similar to what asterisk does with > (re)invites during the hold. I read about and tried the resume-media- > on-hold option and this works fine, the phone correctly gets the moh. > However, when the call is unheld, the media stream continues in media > mode and I'd like to avoid this as much as possible. > > Is there a way to return to bypass media mode automatically when the > call is unheld ? > BTW, is the resume-media-on-hold really implemented for server moh > support, or as a more generic way to put a call back in media mode (so > we can listen for inband/rfc dtmf for example) ? > > I also noticed that the remote phone only sees/display the hold state > when in bypass media mode. I guess no reinvite (a=sendonly) at all is > sent to the held party when in media mode ? Again, is there a solution > for this ? > > > Many thanks ! > > Kind regards > > Gaetan > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Fri Jan 2 09:58:00 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 2 Jan 2009 12:58:00 -0500 Subject: [Freeswitch-users] The transferred call is not hanged up. In-Reply-To: <012b01c96ced$4f291070$ed7b3150$@kr> References: <012b01c96ced$4f291070$ed7b3150$@kr> Message-ID: Are these sip phones? how are you transfering? what model phones are they? What version of freeswitch are you using? Mike On Jan 2, 2009, at 10:18 AM, Sangwoo Jin wrote: > I have met the non-hanged up call after being transferred. > The following is the test scenario. > > 1. A makes a call to B > 2. B answers the call of A and holds the call of A. > 3. B makes a call to C and transfers the call of A to C before C > answers the call of B. > 4. A hangs up the call before C answers the call of A. > > In this state, C?s phone is ringing endless. > Why is not the channel of C hanged up? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/3b685954/attachment.html From brian at freeswitch.org Fri Jan 2 09:57:43 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Jan 2009 11:57:43 -0600 Subject: [Freeswitch-users] bind_meta_app with sip info dtmf in bypass media mode In-Reply-To: References: Message-ID: You need to enable sip info on your sofia profile. /b On Jan 2, 2009, at 10:35 AM, Ga?tan Minet wrote: > Hi, > > It looks like that even when using sip info dtmf, the bind_meta_app > doesn't react on them when rtp is in bypass media mode. Of course > rfc2833/inband cannot be processed in bypass media mode, but doesn't > sip info still go through freeswitch ?. > Is this a limitation of the current implementation, or simply not > possible due to some technical limitations ? Or is there any option to > control this ? > > I'd like to implement (dtmf-triggered) server-side features but still > use bypass media mode. > > Thank you > > Kind regards > > Gaetan From anthony.minessale at gmail.com Fri Jan 2 10:00:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 2 Jan 2009 12:00:55 -0600 Subject: [Freeswitch-users] bind_meta_app with sip info dtmf in bypass media mode In-Reply-To: References: Message-ID: <191c3a030901021000q468bb432pae780f480b77c3b5@mail.gmail.com> yes sorry, you cannot use the 2 features together. On Fri, Jan 2, 2009 at 10:35 AM, Ga?tan Minet wrote: > Hi, > > It looks like that even when using sip info dtmf, the bind_meta_app > doesn't react on them when rtp is in bypass media mode. Of course > rfc2833/inband cannot be processed in bypass media mode, but doesn't > sip info still go through freeswitch ?. > Is this a limitation of the current implementation, or simply not > possible due to some technical limitations ? Or is there any option to > control this ? > > I'd like to implement (dtmf-triggered) server-side features but still > use bypass media mode. > > Thank you > > Kind regards > > Gaetan > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/4cd6ba12/attachment.html From anthony.minessale at gmail.com Fri Jan 2 10:21:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 2 Jan 2009 12:21:57 -0600 Subject: [Freeswitch-users] Conference In-Reply-To: References: <914fc92a0811121154w219ce22ekb31b6d8ce3abe8d8@mail.gmail.com> Message-ID: <191c3a030901021021w71dec3d8lcf90d339a192a9c@mail.gmail.com> I'm guessing you are doing it too fast before the call is completed. On Fri, Jan 2, 2009 at 11:56 AM, Michael Jerris wrote: > > On Jan 2, 2009, at 9:58 AM, Baskar wrote: > > Hi, > > Conference is working well in recent days, but suddenly it in not working > when i transfer the call ,the call get disconnected with the voice file > playing Bye....... > > What is the error? > When i transfer the call why the call get hangup > > api originate sofia/internal/1003%172.20.191.227 &bridge(sofia/default/ > 79841799874 at 172.20.191.228) > > Content-Type: api/response > Content-Length: 41 > > +OK b98ddfca-6e84-4945-a02c-e7d455b58c37 > api uuid_transfer b98ddfca-6e84-4945-a02c-e7d455b58c37 -both 3003 > > Content-Type: api/response > Content-Length: 4 > > +OK > > Correct me were i am Wrong. > > > Look at the debug log to see what happens. > > Mike > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/15205961/attachment-0001.html From gminet-ml at mcit.be Fri Jan 2 11:52:05 2009 From: gminet-ml at mcit.be (=?ISO-8859-1?Q?Ga=EBtan_Minet?=) Date: Fri, 2 Jan 2009 20:52:05 +0100 Subject: [Freeswitch-users] MOH and bypass media In-Reply-To: References: Message-ID: Hi Brian Thank you. Indeed I'm using those options. My question was really about getting out of the media path on unhold. Gaetan NB, sorry for the double post, I didn't realize the list was moderated. On 02/01/2009, at 18:57, Brian West wrote: > Please see: > > > > > > in sip_profiles/internal.xml > > /b > PS you can't get out of the media path when you unhold... FS will be > in the media path hold button moving forward unless you use the > uuid_media command to turn it back off. > > > On Jan 2, 2009, at 4:59 AM, Ga?tan Minet wrote: > >> Hi, >> >> As a newcomer on Freeswitch, first of all thanks you for all your >> work >> on it. I'm currently evaluating it as a replacement for * and really >> appreciate working (ok, playing) with it. >> >> One of our requirements is to optimize the network flows as much as >> possible, and as such avoid any unnecessary rtp proxying. >> >> When a call is in bypass media and one of the phones puts the call on >> hold, the remote phone doesn't play moh. >> Of course I guess this is expected in bypass media, and moh should be >> the responsibility of the UA just like in a pure sip proxy >> environment >> ( so the phone that holds the line should for example invite the moh >> server in a new dialog and swap sdp with the remote phone or >> something >> similar). >> >> Snom phones do this just fine when you configure the moh server uri >> in >> the phone, but others like Thomsons ST2030 do not have that setting >> (... I know, real crap, but we already deployed tons of them so we >> cannot ignore them...) . So when a Thomson puts another phone on >> hold, >> there is no music at all. >> >> I'd like to achieve something similar to what asterisk does with >> (re)invites during the hold. I read about and tried the resume-media- >> on-hold option and this works fine, the phone correctly gets the moh. >> However, when the call is unheld, the media stream continues in media >> mode and I'd like to avoid this as much as possible. >> >> Is there a way to return to bypass media mode automatically when the >> call is unheld ? >> BTW, is the resume-media-on-hold really implemented for server moh >> support, or as a more generic way to put a call back in media mode >> (so >> we can listen for inband/rfc dtmf for example) ? >> >> I also noticed that the remote phone only sees/display the hold state >> when in bypass media mode. I guess no reinvite (a=sendonly) at all is >> sent to the held party when in media mode ? Again, is there a >> solution >> for this ? >> >> >> Many thanks ! >> >> Kind regards >> >> Gaetan >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gminet-ml at mcit.be Fri Jan 2 12:14:59 2009 From: gminet-ml at mcit.be (=?ISO-8859-1?Q?Ga=EBtan_Minet?=) Date: Fri, 2 Jan 2009 21:14:59 +0100 Subject: [Freeswitch-users] bind_meta_app with sip info dtmf in bypass media mode In-Reply-To: References: Message-ID: <775EF1C6-7A25-417C-B431-C8393208E209@mcit.be> Hi If you are talking about , yes it is enabled. Reading the source of mod_sofia i think this is only related to server (re)generated dtmf as it seems to be only used in sofia_send_dtmf(...) ? Gaetan On 02/01/2009, at 18:57, Brian West wrote: > You need to enable sip info on your sofia profile. > > /b > > On Jan 2, 2009, at 10:35 AM, Ga?tan Minet wrote: > >> Hi, >> >> It looks like that even when using sip info dtmf, the bind_meta_app >> doesn't react on them when rtp is in bypass media mode. Of course >> rfc2833/inband cannot be processed in bypass media mode, but doesn't >> sip info still go through freeswitch ?. >> Is this a limitation of the current implementation, or simply not >> possible due to some technical limitations ? Or is there any option >> to >> control this ? >> >> I'd like to implement (dtmf-triggered) server-side features but still >> use bypass media mode. >> >> Thank you >> >> Kind regards >> >> Gaetan > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Fri Jan 2 12:17:21 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Jan 2009 14:17:21 -0600 Subject: [Freeswitch-users] bind_meta_app with sip info dtmf in bypass media mode In-Reply-To: <775EF1C6-7A25-417C-B431-C8393208E209@mcit.be> References: <775EF1C6-7A25-417C-B431-C8393208E209@mcit.be> Message-ID: As anthony said.. it won't work. /b On Jan 2, 2009, at 2:14 PM, Ga?tan Minet wrote: > Hi > > If you are talking about , yes > it is enabled. > Reading the source of mod_sofia i think this is only related to > server (re)generated dtmf as it seems to be only used in > sofia_send_dtmf(...) ? > > > Gaetan From gminet-ml at mcit.be Fri Jan 2 12:34:26 2009 From: gminet-ml at mcit.be (=?ISO-8859-1?Q?Ga=EBtan_Minet?=) Date: Fri, 2 Jan 2009 21:34:26 +0100 Subject: [Freeswitch-users] bind_meta_app with sip info dtmf in bypass media mode In-Reply-To: <191c3a030901021000q468bb432pae780f480b77c3b5@mail.gmail.com> References: <191c3a030901021000q468bb432pae780f480b77c3b5@mail.gmail.com> Message-ID: <934F8590-86C1-49A9-AEAF-173B6D24D906@mcit.be> Hi Anthony Ok, thank you. Gaetan On 02/01/2009, at 19:00, Anthony Minessale wrote: > yes sorry, you cannot use the 2 features together. > > > On Fri, Jan 2, 2009 at 10:35 AM, Ga?tan Minet > wrote: > Hi, > > It looks like that even when using sip info dtmf, the bind_meta_app > doesn't react on them when rtp is in bypass media mode. Of course > rfc2833/inband cannot be processed in bypass media mode, but doesn't > sip info still go through freeswitch ?. > Is this a limitation of the current implementation, or simply not > possible due to some technical limitations ? Or is there any option to > control this ? > > I'd like to implement (dtmf-triggered) server-side features but still > use bypass media mode. > > Thank you > > Kind regards > > Gaetan > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/a2a34ef9/attachment.html From kristian.kielhofner at gmail.com Fri Jan 2 14:59:09 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 2 Jan 2009 17:59:09 -0500 Subject: [Freeswitch-users] mod_event_multicast TTL? Message-ID: <2d9149cd0901021459o68fb27f5k971c0f4339b35e5d@mail.gmail.com> Hey FS Devs/Users, Happy New Year! Is it just me or should mod_event_multicast allow you to specify a TTL on the outbound packet? I know the wiki says you should use VLANs (which I usually do anyway) but the more "standard" way of restricting multicast packets is to set the TTL extremely low (1, etc). I didn't want to be another "gimme gimme gimme" user; I tried looking through the code but I think even my neighbors could here the "whooshing" sound as it went over my head... Thoughts? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From swjinjin at volans.kr Fri Jan 2 18:19:35 2009 From: swjinjin at volans.kr (Sangwoo Jin) Date: Sat, 3 Jan 2009 11:19:35 +0900 Subject: [Freeswitch-users] The transferred call is not hanged up. In-Reply-To: References: <012b01c96ced$4f291070$ed7b3150$@kr> Message-ID: <016f01c96d49$be4da6d0$3ae8f470$@kr> Yes, I'm using SIP phones, Moimstone IP-250. Freeswitch's version is lastest svn. My SIP phone's transfer is using REFER reqeuest and transfer is working well except hanging up. Thanks for reply. Sangwoo. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Saturday, January 03, 2009 2:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] The transferred call is not hanged up. Are these sip phones? how are you transfering? what model phones are they? What version of freeswitch are you using? Mike On Jan 2, 2009, at 10:18 AM, Sangwoo Jin wrote: I have met the non-hanged up call after being transferred. The following is the test scenario. 1. A makes a call to B 2. B answers the call of A and holds the call of A. 3. B makes a call to C and transfers the call of A to C before C answers the call of B. 4. A hangs up the call before C answers the call of A. In this state, C's phone is ringing endless. Why is not the channel of C hanged up? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090103/e3f1eea0/attachment-0001.html From brian at freeswitch.org Fri Jan 2 18:37:56 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Jan 2009 20:37:56 -0600 Subject: [Freeswitch-users] The transferred call is not hanged up. In-Reply-To: <016f01c96d49$be4da6d0$3ae8f470$@kr> References: <012b01c96ced$4f291070$ed7b3150$@kr> <016f01c96d49$be4da6d0$3ae8f470$@kr> Message-ID: <8B96FCE6-6851-4BD3-AF14-DE99626A4642@freeswitch.org> I realized what you're describing here is a behavior that is undefined, and we try to work around it... I have tested this scenario with linksys and polycom and it works fine. This is the scenario where the user doesn't have a clue how to use their phone. You're converting an attended transfer into a blind transfer mid call. If your users are wanting to do a blind transfer its best they do that before dialing C and letting it ring. I'll suspect your phone is putting the call back on hold or loosing the dialog of the first leg in the process causing this to happen. Can you get me a sip trace? and console log with debug on? /b On Jan 2, 2009, at 8:19 PM, Sangwoo Jin wrote: > I have met the non-hanged up call after being transferred. > The following is the test scenario. > > 1. A makes a call to B > 2. B answers the call of A and holds the call of A. > 3. B makes a call to C and transfers the call of A to C before C > answers the call of B. > 4. A hangs up the call before C answers the call of A. > > In this state, C?s phone is ringing endless. > Why is not the channel of C hanged up? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/936d290e/attachment.html From paul at necxgen.com Fri Jan 2 15:54:47 2009 From: paul at necxgen.com (Calvin Paul) Date: Fri, 02 Jan 2009 18:54:47 -0500 Subject: [Freeswitch-users] SCTP Support Message-ID: <495EA947.1050104@necxgen.com> Hello, I need some help to run a sip or iax trunk between two freeswitch boxes using the SCTP transport protocol. I am looking at the documentation and this list and I am pretty sure there is nothing covering SCTP. I am interested in doing this using the freeswitch package in pfsense but I will take anything I can get. The only mention I can find of SCTP is in the features list, an instruction to compile freeswitch with SCTP support and a reference to use TLS with SCTP. Any and all help will be welcomed. PS Can encryption be done on IAX? Paul From brian at freeswitch.org Fri Jan 2 18:43:53 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Jan 2009 20:43:53 -0600 Subject: [Freeswitch-users] SCTP Support In-Reply-To: <495EA947.1050104@necxgen.com> References: <495EA947.1050104@necxgen.com> Message-ID: <05511D17-1405-4CA4-934C-43731B413559@freeswitch.org> Configure sofia with --enable-sctp on linux then dial strings are done like sofia/profile/number at remote;transport=sctp /b PS: more below. On Jan 2, 2009, at 5:54 PM, Calvin Paul wrote: > > Hello, > I need some help to run a sip or iax trunk between two > freeswitch boxes using the SCTP transport protocol. I am looking at > the documentation and this list and I am pretty sure there is nothing > covering SCTP. I am interested in doing this using the freeswitch > package in pfsense but I will take anything I can get. The only > mention I can find of SCTP is in the features list, an instruction to > compile freeswitch with SCTP support and a reference to use TLS with > SCTP. > > Any and all help will be welcomed. > > PS > > Can encryption be done on IAX? Not with ours. > > > Paul From jason at jasonjgw.net Fri Jan 2 18:45:07 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 3 Jan 2009 13:45:07 +1100 Subject: [Freeswitch-users] Enabling SIP traces from the command line interface Message-ID: <20090103024507.GA7443@jdc.jasonjgw.net> The sofia loglevel command is very useful. However, even with sofia loglevel 9, I still don't get a SIP trace. I know all about shutting down FreeSWITCH and setting the environment variables to obtain a SIP trace and voluminous debugging output: # source this file in the current shell, then # run FreeSWITCH. At the FreeSWITCH prompt, # press F8, then make the call that requires debugging, and # capture the output. # type shutdown to end the FreeSWITCH session export SOFIA_DEBUG=9 export NUA_DEBUG=9 export NTA_DEBUG=9 export TPORT_DEBUG=9 export TPORT_LOG=1 this is fine for me, but for people in production for whom uptime is significant, I'm wondering whether there are plans to allow tracing to be dynamically enabled/disabled from the CLI, or whether this is possible now with options that I haven't yet discovered. Please don't interpret this as a complaint or as one of those nagging feature request posts! As mentioned, it doesn't affect my usage scenario - testing and experimentation at home on my desktop Linux box. From brian at freeswitch.org Fri Jan 2 19:11:37 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Jan 2009 21:11:37 -0600 Subject: [Freeswitch-users] The transferred call is not hanged up. In-Reply-To: <016f01c96d49$be4da6d0$3ae8f470$@kr> References: <012b01c96ced$4f291070$ed7b3150$@kr> <016f01c96d49$be4da6d0$3ae8f470$@kr> Message-ID: Please try Committed revision 11061. Next time please open a jira and report soon as possible. btw 1.0.2 came out yesterday. /b On Jan 2, 2009, at 8:19 PM, Sangwoo Jin wrote: > Yes, I?m using SIP phones, Moimstone IP-250. > Freeswitch?s version is lastest svn. > > My SIP phone?s transfer is using REFER reqeuest and transfer is > working well except hanging up. > > Thanks for reply. > Sangwoo. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Jerris > Sent: Saturday, January 03, 2009 2:58 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] The transferred call is not hanged up. > > Are these sip phones? how are you transfering? what model phones > are they? What version of freeswitch are you using? > > Mike > > On Jan 2, 2009, at 10:18 AM, Sangwoo Jin wrote: > > > I have met the non-hanged up call after being transferred. > The following is the test scenario. > > 1. A makes a call to B > 2. B answers the call of A and holds the call of A. > 3. B makes a call to C and transfers the call of A to C before C > answers the call of B. > 4. A hangs up the call before C answers the call of A. > > In this state, C?s phone is ringing endless. > Why is not the channel of C hanged up? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090102/e685d039/attachment-0001.html From brian at freeswitch.org Fri Jan 2 19:30:14 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Jan 2009 21:30:14 -0600 Subject: [Freeswitch-users] Enabling SIP traces from the command line interface In-Reply-To: <20090103024507.GA7443@jdc.jasonjgw.net> References: <20090103024507.GA7443@jdc.jasonjgw.net> Message-ID: Jason, Its something that needs to be added to the sofia lib. Till then its gonna be on the wish list. ;) /b On Jan 2, 2009, at 8:45 PM, Jason White wrote: > The sofia loglevel command is very useful. However, even with sofia > loglevel > 9, I still don't get a SIP trace. > > I know all about shutting down FreeSWITCH and setting the environment > variables to obtain a SIP trace and voluminous debugging output: > > # source this file in the current shell, then > # run FreeSWITCH. At the FreeSWITCH prompt, > # press F8, then make the call that requires debugging, and > # capture the output. > # type shutdown to end the FreeSWITCH session > export SOFIA_DEBUG=9 > export NUA_DEBUG=9 > export NTA_DEBUG=9 > export TPORT_DEBUG=9 > export TPORT_LOG=1 > > this is fine for me, but for people in production for whom uptime is > significant, I'm wondering whether there are plans to allow tracing > to be > dynamically enabled/disabled from the CLI, or whether this is > possible now > with options that I haven't yet discovered. > > Please don't interpret this as a complaint or as one of those > nagging feature > request posts! > As mentioned, it doesn't affect my usage scenario - testing and > experimentation at home on my desktop Linux box. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From swjinjin at volans.kr Fri Jan 2 23:42:21 2009 From: swjinjin at volans.kr (Sangwoo Jin) Date: Sat, 3 Jan 2009 16:42:21 +0900 Subject: [Freeswitch-users] The transferred call is not hanged up. In-Reply-To: References: <012b01c96ced$4f291070$ed7b3150$@kr> <016f01c96d49$be4da6d0$3ae8f470$@kr> Message-ID: <018601c96d76$cff63f00$6fe2bd00$@kr> My problem is fixed in rev. 11061. Thank you for fast fix. Sangwoo. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Saturday, January 03, 2009 12:12 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] The transferred call is not hanged up. Please try Committed revision 11061. Next time please open a jira and report soon as possible. btw 1.0.2 came out yesterday. /b On Jan 2, 2009, at 8:19 PM, Sangwoo Jin wrote: Yes, I'm using SIP phones, Moimstone IP-250. Freeswitch's version is lastest svn. My SIP phone's transfer is using REFER reqeuest and transfer is working well except hanging up. Thanks for reply. Sangwoo. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Saturday, January 03, 2009 2:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] The transferred call is not hanged up. Are these sip phones? how are you transfering? what model phones are they? What version of freeswitch are you using? Mike On Jan 2, 2009, at 10:18 AM, Sangwoo Jin wrote: I have met the non-hanged up call after being transferred. The following is the test scenario. 1. A makes a call to B 2. B answers the call of A and holds the call of A. 3. B makes a call to C and transfers the call of A to C before C answers the call of B. 4. A hangs up the call before C answers the call of A. In this state, C's phone is ringing endless. Why is not the channel of C hanged up? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090103/108bed33/attachment.html From kristjan.ugrin at gmail.com Sat Jan 3 00:43:02 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Sat, 03 Jan 2009 09:43:02 +0100 Subject: [Freeswitch-users] Changing internal profile Message-ID: I'm trying to change internal address to something else than default external, so I would be able to make calls from phone on wireless to e.g. gtalk users. I already tried this scenario at different location and it work, but everything was on same lan, now I have eth0 interface with external ip e.g. 212.235.180.41 eth1 interface with internal ip 192.168.0.1 on eth1 is wireless router attached with wireless clients (my phone is 192.168.0.102), messages are going trough, no problem with that. If I got it right I have to change: to 192.168.0.1, after that I modified acl.conf.xml domains section to: but I'm getting forbidden: 2009-01-03 09:39:42 [WARNING] sofia_reg.c:1533 sofia_reg_parse_auth() Can't find user [1000 at 192.168.0.1] You must define a domain called '192.168.0.1' in your directory and add a user with the id="1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. What directory? Should I delete added node from acl.conf.xml? -- kriko From yudha2008 at gmail.com Sat Jan 3 01:44:00 2009 From: yudha2008 at gmail.com (Baskar) Date: Sat, 3 Jan 2009 15:14:00 +0530 Subject: [Freeswitch-users] Conference In-Reply-To: <191c3a030901021021w71dec3d8lcf90d339a192a9c@mail.gmail.com> References: <914fc92a0811121154w219ce22ekb31b6d8ce3abe8d8@mail.gmail.com> <191c3a030901021021w71dec3d8lcf90d339a192a9c@mail.gmail.com> Message-ID: *Hi, Conference is working well before. Suddenly it is not working i have tested manually by calling 3002 conference room from Soft phone .But call get hangup directly with the ivr message BYE. I have given "^(30\d{2})$" in default.xml . what is the error? I have pasted the console logs in this path http://pastebin.freeswitch.org/6617 Try to correct where is error and what is the error???? -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090103/1f21f6b8/attachment.html From jason at jasonjgw.net Sat Jan 3 01:59:25 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 3 Jan 2009 20:59:25 +1100 Subject: [Freeswitch-users] Enabling SIP traces from the command line interface In-Reply-To: References: <20090103024507.GA7443@jdc.jasonjgw.net> Message-ID: <20090103095925.GA9551@jdc.jasonjgw.net> Brian West wrote: > Its something that needs to be added to the sofia lib. Till then its > gonna be on the wish list. ;) Meanwhile (and I hope the Sofia developers eventually implement it), you can set it in the Sip profile configuration. Running sofia profile restart reloadxml is enough to enable it; I've just tested it. From brian at freeswitch.org Sat Jan 3 08:07:35 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 3 Jan 2009 10:07:35 -0600 Subject: [Freeswitch-users] Changing internal profile In-Reply-To: References: Message-ID: What is the output of "sofia status"? And what have you changed from the default config ie in vars.xml? /b On Jan 3, 2009, at 2:43 AM, kriko wrote: > I'm trying to change internal address to something else than default > external, so I would be able to make calls from phone on wireless > to e.g. gtalk users. I already tried this scenario at different > location and it work, but everything was on same lan, now I have > eth0 interface with external ip e.g. 212.235.180.41 > eth1 interface with internal ip 192.168.0.1 > > on eth1 is wireless router attached with wireless clients (my phone > is 192.168.0.102), messages are going trough, no problem with that. > If I got it right I have to change: > > > > to 192.168.0.1, after that I modified acl.conf.xml domains section to: > > > > > > > but I'm getting forbidden: > 2009-01-03 09:39:42 [WARNING] sofia_reg.c:1533 > sofia_reg_parse_auth() Can't find user [1000 at 192.168.0.1] > You must define a domain called '192.168.0.1' in your directory and > add a user with the id="1000" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > > What directory? Should I delete added node from acl.conf.xml? > > -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090103/14ff9d23/attachment.html From anthony.minessale at gmail.com Sat Jan 3 08:09:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 3 Jan 2009 10:09:38 -0600 Subject: [Freeswitch-users] Conference In-Reply-To: References: <914fc92a0811121154w219ce22ekb31b6d8ce3abe8d8@mail.gmail.com> <191c3a030901021021w71dec3d8lcf90d339a192a9c@mail.gmail.com> Message-ID: <191c3a030901030809q404a2977y1d4337fb5fd93556@mail.gmail.com> you might want to press f8 first so you actually get debug log. On Sat, Jan 3, 2009 at 3:44 AM, Baskar wrote: > *Hi, > > Conference is working well before. Suddenly it is not working i have tested > manually by calling 3002 conference room from Soft phone .But call get > hangup directly with the ivr message BYE. > > I have given "^(30\d{2})$" in default.xml . what is the error? > > I have pasted the console logs in this path > http://pastebin.freeswitch.org/6617 > > Try to correct where is error and what is the error???? > > -- > Warm Regards, > N.Baskar > * > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090103/bb1b8830/attachment.html From kristjan.ugrin at gmail.com Sat Jan 3 09:23:44 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Sat, 03 Jan 2009 18:23:44 +0100 Subject: [Freeswitch-users] Changing internal profile In-Reply-To: References: Message-ID: This is sofia status: http://pastebin.com/m2d6d4a80 When registering I get: http://pastebin.com/m2381d6a5 I can also provide a wireshark trace, but I don't see anything unusual or network related-problem. FS actually replies to the phone with 403 Forbidden On Sat, 03 Jan 2009 17:07:35 +0100, Brian West wrote: > What is the output of "sofia status"? > > And what have you changed from the default config ie in vars.xml? > > /b > > > On Jan 3, 2009, at 2:43 AM, kriko wrote: > >> I'm trying to change internal address to something else than default >> external, so I would be able to make calls from phone on wireless >> to e.g. gtalk users. I already tried this scenario at different >> location and it work, but everything was on same lan, now I have >> eth0 interface with external ip e.g. 212.235.180.41 >> eth1 interface with internal ip 192.168.0.1 >> >> on eth1 is wireless router attached with wireless clients (my phone >> is 192.168.0.102), messages are going trough, no problem with that. >> If I got it right I have to change: >> >> >> >> to 192.168.0.1, after that I modified acl.conf.xml domains section to: >> >> >> >> >> >> >> but I'm getting forbidden: >> 2009-01-03 09:39:42 [WARNING] sofia_reg.c:1533 >> sofia_reg_parse_auth() Can't find user [1000 at 192.168.0.1] >> You must define a domain called '192.168.0.1' in your directory and >> add a user with the id="1000" attribute >> and you must configure your device to use the proper domain in it's >> authentication credentials. >> >> What directory? Should I delete added node from acl.conf.xml? >> >> -- > -- kriko From brian at freeswitch.org Sat Jan 3 09:42:04 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 3 Jan 2009 11:42:04 -0600 Subject: [Freeswitch-users] Changing internal profile In-Reply-To: References: Message-ID: You changed the ip and domain ? Sent from my iPhone On Jan 3, 2009, at 11:23 AM, kriko wrote: > This is sofia status: > http://pastebin.com/m2d6d4a80 > > When registering I get: > http://pastebin.com/m2381d6a5 > > I can also provide a wireshark trace, but I don't see anything > unusual or network related-problem. > FS actually replies to the phone with 403 Forbidden > > On Sat, 03 Jan 2009 17:07:35 +0100, Brian West > wrote: > >> What is the output of "sofia status"? >> >> And what have you changed from the default config ie in vars.xml? >> >> /b >> >> >> On Jan 3, 2009, at 2:43 AM, kriko wrote: >> >>> I'm trying to change internal address to something else than default >>> external, so I would be able to make calls from phone on wireless >>> to e.g. gtalk users. I already tried this scenario at different >>> location and it work, but everything was on same lan, now I have >>> eth0 interface with external ip e.g. 212.235.180.41 >>> eth1 interface with internal ip 192.168.0.1 >>> >>> on eth1 is wireless router attached with wireless clients (my phone >>> is 192.168.0.102), messages are going trough, no problem with that. >>> If I got it right I have to change: >>> >>> >>> >>> to 192.168.0.1, after that I modified acl.conf.xml domains section >>> to: >>> >>> >>> >>> >>> >>> >>> but I'm getting forbidden: >>> 2009-01-03 09:39:42 [WARNING] sofia_reg.c:1533 >>> sofia_reg_parse_auth() Can't find user [1000 at 192.168.0.1] >>> You must define a domain called '192.168.0.1' in your directory and >>> add a user with the id="1000" attribute >>> and you must configure your device to use the proper domain in it's >>> authentication credentials. >>> >>> What directory? Should I delete added node from acl.conf.xml? >>> >>> -- >> > > > > -- > kriko > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kristjan.ugrin at gmail.com Sat Jan 3 09:46:11 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Sat, 03 Jan 2009 18:46:11 +0100 Subject: [Freeswitch-users] Changing internal profile In-Reply-To: References: Message-ID: I changed exactly what I wrote in previous mails. Everything else is default. Is there anything else to change? On Sat, 03 Jan 2009 18:42:04 +0100, Brian West wrote: > You changed the ip and domain ? > > Sent from my iPhone > > On Jan 3, 2009, at 11:23 AM, kriko wrote: > >> This is sofia status: >> http://pastebin.com/m2d6d4a80 >> >> When registering I get: >> http://pastebin.com/m2381d6a5 >> >> I can also provide a wireshark trace, but I don't see anything >> unusual or network related-problem. >> FS actually replies to the phone with 403 Forbidden >> >> On Sat, 03 Jan 2009 17:07:35 +0100, Brian West >> wrote: >> >>> What is the output of "sofia status"? >>> >>> And what have you changed from the default config ie in vars.xml? >>> >>> /b >>> >>> >>> On Jan 3, 2009, at 2:43 AM, kriko wrote: >>> >>>> I'm trying to change internal address to something else than default >>>> external, so I would be able to make calls from phone on wireless >>>> to e.g. gtalk users. I already tried this scenario at different >>>> location and it work, but everything was on same lan, now I have >>>> eth0 interface with external ip e.g. 212.235.180.41 >>>> eth1 interface with internal ip 192.168.0.1 >>>> >>>> on eth1 is wireless router attached with wireless clients (my phone >>>> is 192.168.0.102), messages are going trough, no problem with that. >>>> If I got it right I have to change: >>>> >>>> >>>> >>>> to 192.168.0.1, after that I modified acl.conf.xml domains section >>>> to: >>>> >>>> >>>> >>>> >>>> >>>> >>>> but I'm getting forbidden: >>>> 2009-01-03 09:39:42 [WARNING] sofia_reg.c:1533 >>>> sofia_reg_parse_auth() Can't find user [1000 at 192.168.0.1] >>>> You must define a domain called '192.168.0.1' in your directory and >>>> add a user with the id="1000" attribute >>>> and you must configure your device to use the proper domain in it's >>>> authentication credentials. >>>> >>>> What directory? Should I delete added node from acl.conf.xml? >>>> >>>> -- >>> >> >> >> >> -- >> kriko >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- kriko From brian at freeswitch.org Sat Jan 3 10:21:39 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 3 Jan 2009 12:21:39 -0600 Subject: [Freeswitch-users] Changing internal profile In-Reply-To: References: Message-ID: <0427A651-04AA-486C-A4C5-F6D3A2C2309E@freeswitch.org> The detected domain is 212.235.180.41 which caused this problem. You have two options here. Since we detected and set your default domain to 212.235.180.41 on start up... you're registering to the 192.168.0.1 ip aka the internal interface. The inbound register packet has: From: We take the part before the @ aka the username, then we take the part after the @ aka the domain name. FreeSWITCH will then look thru your directory looking for domain which in this case is 192.168.0.1 which it can't find because we detected your public IP and set it up as 212.235.180.41. So the error message is telling you that you do not have a domain called 192.168.0.1 with a user 1000 in it. So what you have to do here is understand that SIP like email works on the concept of domains. user at host. A few things you need to know are this: (SOMEONE WIKIFY THIS PLEASE and expand on it. Find me on IRC if you have questions) sofia profile params: challenge-realm: (default configuration uses auto_from) Choose the realm challenge key. Default is auto_to if not set. auto_from - uses the from field as the value for the sip realm. auto_to - uses the to field as the value for the sip realm. - you can input any value to use for the sip realm. force-register-domain: This will force the profile to ignore the domain in the to or from packet and force it to the value listed here for this param. This will store the info into the database with the user@ force-register-db-domain: This will work in conjunction with force-register-domain so that the forced domain is stored in the database also. ATTENTION HERE IS WHAT YOU SHOULD DO: So what I recommend for you is to open up internal.xml and uncomment the force-register-domain and force-register-db-domain params on the internal profile. Also make sure internal.xml has sip-ip and rtp-ip set to 192.168.0.1 and make sure your phones register to 192.168.0.1. /b On Jan 3, 2009, at 11:46 AM, kriko wrote: > I changed exactly what I wrote in previous mails. > Everything else is default. > > Is there anything else to change? From ronmccar at gmail.com Sun Jan 4 09:55:44 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Sun, 4 Jan 2009 10:55:44 -0700 Subject: [Freeswitch-users] Forward calls that need to be transcoded to a Asterisk box Message-ID: <3885f4fe0901040955h8acabedmeb2052285db154ab@mail.gmail.com> Hi List, I have ready in places that we can run regex on the SDP and then determine if the call need to be transcoded we can then send a redirect / 302 to the user and have them contact say a Asterisk box which can then run transcoding for us. Now my question is how do I do that? My thinking is the call hits the dialplan and run regex on the SDP and redirect if needed. I only need to send some calls to the Asterisk box for transcode and most of the carriers I use accept g729, but some do not. Could I take that value in the SDP and then pass it to the diaplan so I can then add that information to my LCR query that I run, I do all this with the curl dialplan not the built in in LCR module. Any help would be great. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090104/b8b4b8ed/attachment.html From mszlazak at aol.com Sun Jan 4 12:14:27 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sun, 04 Jan 2009 15:14:27 -0500 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. Message-ID: <8CB3CC02D83DFD4-12D4-9FB@WEBMAIL-DY39.sysops.aol.com> I believe there is a variable $${domain} that is used by FreeSwitch for it's internal IP address. This address was 10.0.0.2 but changes when I connect or disconnect things to my computer and I wanted to make it "static." The "Getting Started Guide" suggests this is do-able but it seems I have to change things in several places besides vars.xml like \sip_profiles\internal.xml and maybe other places. Is there some general way to set my internal domain? Thanks. ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090104/1507e93d/attachment.html From krice at suspicious.org Sun Jan 4 12:48:04 2009 From: krice at suspicious.org (Ken Rice) Date: Sun, 04 Jan 2009 14:48:04 -0600 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: <8CB3CC02D83DFD4-12D4-9FB@WEBMAIL-DY39.sysops.aol.com> Message-ID: That is the DOMAIN not the IP... Hence the name is domain... If you look thru vars.xml you?ll see other variables for the IP... From: Reply-To: Date: Sun, 04 Jan 2009 15:14:27 -0500 To: Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. I believe there is a variable $${domain} that is used by FreeSwitch for it's internal IP address. This address was 10.0.0.2 but changes when I connect or disconnect things to my computer and I wanted to make it "static." The "Getting Started Guide" suggests this is do-able but it seems I have to change things in several places besides vars.xml like \sip_profiles\internal.xml and maybe other places. Is there some general way to set my internal domain? Thanks. Get a free MP3 every day with the Spinner.com Toolbar. Get it Now . _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090104/2e1c7b5d/attachment.html From mszlazak at aol.com Sun Jan 4 13:05:40 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sun, 04 Jan 2009 16:05:40 -0500 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: Message-ID: <8CB3CC755875C1C-12D4-BE9@WEBMAIL-DY39.sysops.aol.com> Ok, so is it? $${local_ip_v4} that needs changing somewhere? I see stuff in vars.xml for external address changes but not for internal ip address changes?? -----Original Message----- From: Ken Rice To: freeswitch-users at lists.freeswitch.org Sent: Sun, 4 Jan 2009 12:48 pm Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. That is the DOMAIN not the IP... Hence the name is domain... If you look thru vars.xml you?ll see other variables for the IP... From: Reply-To: Date: Sun, 04 Jan 2009 15:14:27 -0500 To: Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. I believe there is a variable $${domain} that is used by FreeSwitch for it's internal IP address. This address was 10.0.0.2 but changes when I connect or disconnect things to my computer and I wanted to make it "static." The "Getting Started Guide" suggests this is do-able but it seems I have to change things in several places besides vars.xml like \sip_profiles\internal.xml and maybe other places. Is there some general way to set my internal domain? Thanks. ?? Get a free MP3 every day with the Spinner.com Toolbar. Get it Now . _______________________________________________ Freeswitch-users mailing li st Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090104/f52ac518/attachment.html From brian at freeswitch.org Sun Jan 4 13:54:27 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 4 Jan 2009 15:54:27 -0600 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: <8CB3CC755875C1C-12D4-BE9@WEBMAIL-DY39.sysops.aol.com> References: <8CB3CC755875C1C-12D4-BE9@WEBMAIL-DY39.sysops.aol.com> Message-ID: <9EE4A945-56BF-467B-81F2-88AE1F08E934@freeswitch.org> or you can open the confit in sip_profiles/*.xml and input the sip-ip and rtp-ip for the sofia profile. /b On Jan 4, 2009, at 3:05 PM, mszlazak at aol.com wrote: > Ok, so is it $${local_ip_v4} that needs changing somewhere? I see > stuff in vars.xml for external address changes but not for internal > ip address changes?? > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090104/0963e905/attachment.html From jason at jasonjgw.net Sun Jan 4 15:15:45 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 5 Jan 2009 10:15:45 +1100 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: <8CB3CC755875C1C-12D4-BE9@WEBMAIL-DY39.sysops.aol.com> References: <8CB3CC755875C1C-12D4-BE9@WEBMAIL-DY39.sysops.aol.com> Message-ID: <20090104231545.GB5849@jdc.jasonjgw.net> mszlazak at aol.com wrote: > > Ok, so is it? $${local_ip_v4} that needs changing somewhere? I see stuff in > vars.xml for external address changes but not for internal ip address > changes?? If you set $${local_ip_v4} in vars.xml it will determine which address FreeSWITCH binds to, at least in the default configuration. Addresses can be configured more flexibly in the SIP profiles. From mszlazak at aol.com Sun Jan 4 16:23:11 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sun, 04 Jan 2009 19:23:11 -0500 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. Message-ID: <8CB3CE2ED37CC9C-12D4-12E9@WEBMAIL-DY39.sysops.aol.com> I see the following in that file: ? and thought that's what might need changing to: ? But your looking at the same variable I was and I'm guessing something else might be in order like: I'll see if either of these work unless you have a different suggestion. You and Brian also suggested a more specific approach in sip_profiles/*.xml and input the sip-ip and rtp-ip for the sofia profile. ??? ????? to???? ??? ?????? to???? What's the advantage to doing it in the sip_profiles\internal.xml file over the general way in vars.xml? ? ? -----Original Message----- From: Jason White To: freeswitch-users at lists.freeswitch.org Sent: Sun, 4 Jan 2009 3:15 pm Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. mszlazak at aol.com wrote: > > Ok, so is it? $${local_ip_v4} that needs changing somewhere? I see stuff in > vars.xml for external address changes but not for internal ip address > changes?? If you set $${local_ip_v4} in vars.xml it will determine which address FreeSWITCH binds to, at least in the default configuration. Addresses can be configured more flexibly in the SIP profiles. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090104/9934d328/attachment-0001.html From jason at jasonjgw.net Sun Jan 4 16:49:24 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 5 Jan 2009 11:49:24 +1100 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: <8CB3CE2ED37CC9C-12D4-12E9@WEBMAIL-DY39.sysops.aol.com> References: <8CB3CE2ED37CC9C-12D4-12E9@WEBMAIL-DY39.sysops.aol.com> Message-ID: <20090105004924.GA6863@jdc.jasonjgw.net> mszlazak at aol.com wrote: > What's the advantage to doing it in the sip_profiles\internal.xml file over > the general way in vars.xml? The obvious advantage is that you can set those RTP and SIP addresses independently of each other in the profile, so presumably they could be different if that's what you wanted. From brian at freeswitch.org Sun Jan 4 17:56:53 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 4 Jan 2009 19:56:53 -0600 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: <8CB3CE2ED37CC9C-12D4-12E9@WEBMAIL-DY39.sysops.aol.com> References: <8CB3CE2ED37CC9C-12D4-12E9@WEBMAIL-DY39.sysops.aol.com> Message-ID: <81FA5575-1864-4305-9DD6-B7477F9A1ECF@freeswitch.org> I'm going to guess you don't have the latest configs. bind_server_ip is for Dingaling ONLY at this point and is noted with the nice HUGE warning above the setting. You can set the local_ip_v4 address but you're better off setting the ip's in the profile. /b On Jan 4, 2009, at 6:23 PM, mszlazak at aol.com wrote: > I see the following in that file: > > > > and thought that's what might need changing to: > > > > But your looking at the same variable I was and I'm guessing > something else might be in order like: > > > > I'll see if either of these work unless you have a different > suggestion. > > You and Brian also suggested a more specific approach in > sip_profiles/*.xml and input the sip-ip and rtp-ip for the sofia > profile. > > to > > > to > > > > What's the advantage to doing it in the sip_profiles\internal.xml > file over the general way in vars.xml? > > > > > > > > > > > -----Original Message----- > From: Jason White > To: freeswitch-users at lists.freeswitch.org > Sent: Sun, 4 Jan 2009 3:15 pm > Subject: Re: [Freeswitch-users] How do I set my FS internal ip > address to a "static" value. > > mszlazak at aol.com wrote: > > > > > > > > > > Ok, so is it $${local_ip_v4} that needs changing somewhere? I > see stuff in > > > > > vars.xml for external address changes but not for internal ip > address > > > > > changes?? > > > > > > > > If you set $${local_ip_v4} in vars.xml it will determine which address > > > > FreeSWITCH binds to, at least in the default configuration. > > > > > > > > Addresses can be configured more flexibly in the SIP profiles. > > > > > > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > Get a free MP3 every day with the Spinner.com Toolbar. Get it Now. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090104/91465a28/attachment.html From mszlazak at aol.com Sun Jan 4 22:06:10 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 05 Jan 2009 01:06:10 -0500 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: <81FA5575-1864-4305-9DD6-B7477F9A1ECF@freeswitch.org> References: <8CB3CE2ED37CC9C-12D4-12E9@WEBMAIL-DY39.sysops.aol.com> <81FA5575-1864-4305-9DD6-B7477F9A1ECF@freeswitch.org> Message-ID: <8CB3D12D6F57033-1358-20A8@WEBMAIL-MZ16.sysops.aol.com> Brian and Jason. I tried changing $${local_ip_v4} to a static IP in the vars.xml and internal and external sip_profiles files. This didn't work. What I want to do is maybe more related to what local ip address my Windows machine binds FreeSwitch to. It's currently associating it to 10.0.0.2 if I change? $${local_ip_v4} to 10.0.0.3 then FS gets errors and sofia status doesn't show any ip address associated with FS. I'm guessing I need to get Windows always associating FS with a static IP instead of a possibly changing value. I did this with a Linksys SPA3103 in it's configuration menu so is there something analogous for FS and how do I do that? Thanks. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Sun, 4 Jan 2009 5:56 pm Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. I'm going to guess you don't have the latest configs. ?bind_server_ip is for Dingaling ONLY at this point and is noted with the nice HUGE warning above the setting. You can set the local_ip_v4 address but you're better off setting the ip's in the profile. /b On Jan 4, 2009, at 6:23 PM, mszlazak at aol.com wrote: I see the following in that file: ? and thought that's what might need changing to: ? But your looking at the same variable I was and I'm guessing something else might be in order like: I'll see if either of these work unless you have a different suggestion. You and Brian also suggested a more specific approach in sip_profiles/*.xml and input the sip-ip and rtp-ip for the sofia profile. ??? ????? to???? ??? ?????? to???? What's the advantage to doing it in the sip_profiles\internal.xml file over the general way in vars.xml? ? ? -----Original Message----- From: Jason White To: freeswitch-users at lists.freeswitch.org Sent: Sun, 4 Jan 2009 3:15 pm Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. mszlazak at aol.com wrote: > > Ok, so is it? $${local_ip_v4} that needs changing somewhere? I see stuff in > vars.xml for external address changes but not for internal ip address > changes?? If you set $${local_ip_v4} in vars.xml it will determine which address FreeSWITCH binds to, at least in the default configuration. Addresses can be configured more flexibly in the SIP profiles. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Get a free MP3 every day with the Spinner.com Toolbar. Get it Now. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/901c0468/attachment.html From brian at freeswitch.org Sun Jan 4 22:13:05 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 5 Jan 2009 00:13:05 -0600 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: <8CB3D12D6F57033-1358-20A8@WEBMAIL-MZ16.sysops.aol.com> References: <8CB3CE2ED37CC9C-12D4-12E9@WEBMAIL-DY39.sysops.aol.com> <81FA5575-1864-4305-9DD6-B7477F9A1ECF@freeswitch.org> <8CB3D12D6F57033-1358-20A8@WEBMAIL-MZ16.sysops.aol.com> Message-ID: Well I did say open up sip_profiles/internal.xml or sip_profiles/ external.xml and every place it has $${local_ip_v4} REPLACE IT with the desired IP. Is 10.0.0.3 bound to your windows machine? /b On Jan 5, 2009, at 12:06 AM, mszlazak at aol.com wrote: > I tried changing $${local_ip_v4} to a static IP in the vars.xml and > internal and external sip_profiles files. This didn't work. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/0aaa51b1/attachment-0001.html From jason at jasonjgw.net Sun Jan 4 22:17:35 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 5 Jan 2009 17:17:35 +1100 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: <8CB3D12D6F57033-1358-20A8@WEBMAIL-MZ16.sysops.aol.com> References: <8CB3CE2ED37CC9C-12D4-12E9@WEBMAIL-DY39.sysops.aol.com> <81FA5575-1864-4305-9DD6-B7477F9A1ECF@freeswitch.org> <8CB3D12D6F57033-1358-20A8@WEBMAIL-MZ16.sysops.aol.com> Message-ID: <20090105061735.GA1376@jdc.jasonjgw.net> mszlazak at aol.com wrote: > > I tried changing $${local_ip_v4} to a static IP in the vars.xml and internal > and external sip_profiles files. This didn't work. Make sure that one of your network interfaces - the one that you want FreeSWITCH to use - is assigned that address first, otherwise there will be nothing for FreeSWITCH to bind to when it starts. I don't know much about Windows, so if you're running FreeSWITCH on Windows, I'm sure there will be others on the list who are in a position to help. If you're using Linux, just make sure that the address you want is associated with an interface, e.g., using ifconfig, then start FreeSWITCH with the pre-processor variable $${local_ip_v4} set appropriately. From mszlazak at aol.com Sun Jan 4 22:37:02 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 05 Jan 2009 01:37:02 -0500 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: References: <8CB3CE2ED37CC9C-12D4-12E9@WEBMAIL-DY39.sysops.aol.com><81FA5575-1864-4305-9DD6-B7477F9A1ECF@freeswitch.org><8CB3D12D6F57033-1358-20A8@WEBMAIL-MZ16.sysops.aol.com> Message-ID: <8CB3D1726CDBE87-1358-2124@WEBMAIL-MZ16.sysops.aol.com> Brian, I did that and know it was every instance since I used Textpad's "replace" function. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Sun, 4 Jan 2009 10:13 pm Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. Well I did say open up sip_profiles/internal.xml or sip_profiles/external.xml and every place it has $${local_ip_v4} REPLACE IT with the desired IP. ?Is?10.0.0.3 bound to your windows machine? /b On Jan 5, 2009, at 12:06 AM, mszlazak at aol.com wrote: I tried changing $${local_ip_v4} to a static IP in the vars.xml and internal and external sip_profiles files. This didn't work. = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/4d3c5b39/attachment.html From mszlazak at aol.com Sun Jan 4 22:51:48 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 05 Jan 2009 01:51:48 -0500 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: <20090105061735.GA1376@jdc.jasonjgw.net> References: <8CB3CE2ED37CC9C-12D4-12E9@WEBMAIL-DY39.sysops.aol.com><81FA5575-1864-4305-9DD6-B7477F9A1ECF@freeswitch.org><8CB3D12D6F57033-1358-20A8@WEBMAIL-MZ16.sysops.aol.com> <20090105061735.GA1376@jdc.jasonjgw.net> Message-ID: <8CB3D1936EE3E33-1358-215E@WEBMAIL-MZ16.sysops.aol.com> Jason, I have no idea either but if I assign an IP address to the network interface, say 10.0.0.3 then I may have another problem. I have a Voxeo Prophecy ASR server being assigned to 10.0.0.2. Actually both FS and Prophecy were being assigned by Windows to 10.0.0.2 but this was causing audio transfer problems when I bridged FS to Prophecy. So I wanted to assign FS to something else but I *believe* that Windows only uses one adaptor. What I need to do next I don't know (create another adaptor) and probably I've never done. -----Original Message----- From: Jason White To: freeswitch-users at lists.freeswitch.org Sent: Sun, 4 Jan 2009 10:17 pm Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. mszlazak at aol.com wrote: > > I tried changing $${local_ip_v4} to a static IP in the vars.xml and internal > and external sip_profiles files. This didn't work. Make sure that one of your network interfaces - the one that you want FreeSWITCH to use - is assigned that address first, otherwise there will be nothing for FreeSWITCH to bind to when it starts. I don't know much about Windows, so if you're running FreeSWITCH on Windows, I'm sure there will be others on the list who are in a position to help. If you're using Linux, just make sure that the address you want is associated with an interface, e.g., using ifconfig, then start FreeSWITCH with the pre-processor variable $${local_ip_v4} set appropriately. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/451880da/attachment.html From yudha2008 at gmail.com Sun Jan 4 23:50:23 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 5 Jan 2009 13:20:23 +0530 Subject: [Freeswitch-users] busy tone detection In-Reply-To: References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com> <51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com> <2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org> <965B5A92-4182-489B-8AA0-6E6173C1931B@jerris.com> Message-ID: *Hi Michael Jerris, I have some questions can you answer me so that it will helpful to me **** 1)The tone detect will work only with openzap. Am i correct?" **2)To detect with IP related media gateway like audiocode can we get the response code from the freeswitch console?* * 3)**In the above mail you have told that busy number and detecting a busy tone are COMPLETELY different things **Can u explain briefly about How to detecting a busy* *tone*. Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/2a377b37/attachment.html From saigop at gmail.com Sun Jan 4 23:53:14 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Mon, 5 Jan 2009 13:23:14 +0530 Subject: [Freeswitch-users] SIP response code in Freeswitch Message-ID: <2ea4d47e0901042353v608a0747r51e577e91e22108c@mail.gmail.com> Hi, Is there any possibilities that Freeswitch may detect the SIP response code from the IP media gateway. -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/971fa9fd/attachment.html From saigop at gmail.com Sun Jan 4 23:54:13 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Mon, 5 Jan 2009 13:24:13 +0530 Subject: [Freeswitch-users] VXML support in Freeswitch Message-ID: <2ea4d47e0901042354r72e21bb7v9388910f0fbb21f9@mail.gmail.com> Does freeswitch support VXML? Is there any separate module for this. -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/b68a369d/attachment.html From fidibus83 at aol.com Mon Jan 5 01:20:53 2009 From: fidibus83 at aol.com (fidibus83) Date: Mon, 5 Jan 2009 10:20:53 +0100 Subject: [Freeswitch-users] (no subject) In-Reply-To: <491E1798-26E7-48C9-950C-8BEBA40BB928@jerris.com> References: <00fd01c96ccd$810ce050$6445310a@Franzi> <491E1798-26E7-48C9-950C-8BEBA40BB928@jerris.com> Message-ID: <012d01c96f16$e8e05770$6445310a@Franzi> Great! Is there an example in the wiki? I don?t find something like this! _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Freitag, 2. Januar 2009 18:56 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] (no subject) Yes, callback agents are now part of mod_fifo as well. On Jan 2, 2009, at 6:30 AM, fidibus83 wrote: Hello! Can FS do something like an ACD? I have configure a Supportqueue like this: And my Agents are in a Call Group. How can I transfer the call to a free agent? I saw the example in mod_fifo. But my Agent can?t wait in the queue all the time. Is it possible that the agents are called back when a call is waiting in the queue? = -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/32c1de6a/attachment-0001.html From oseslija at gmail.com Mon Jan 5 01:30:42 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Mon, 5 Jan 2009 10:30:42 +0100 Subject: [Freeswitch-users] SIP response code in Freeswitch In-Reply-To: <2ea4d47e0901042353v608a0747r51e577e91e22108c@mail.gmail.com> References: <2ea4d47e0901042353v608a0747r51e577e91e22108c@mail.gmail.com> Message-ID: <4468a6770901050130u61ee7602x3b57b6d2495e6fdd@mail.gmail.com> Hi, there is proto_specific_hangup_cause switch variable you can use for the cdr i.e. You can also use SIP messages number for a continue_on_fail action like: Regards, Ognjen On Mon, Jan 5, 2009 at 8:53 AM, Gopalakrishnan A.N wrote: > Hi, > Is there any possibilities that Freeswitch may detect the SIP response > code from the IP media gateway. > > -- > Thank you with regards, > Gopal, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/99083c7e/attachment.html From kristjan.ugrin at gmail.com Mon Jan 5 02:09:51 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Mon, 05 Jan 2009 11:09:51 +0100 Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite Message-ID: Is it possible for dingaling to automatically accept all new friends who sent me an invite? -- kriko From stkn at freeswitch.org Mon Jan 5 03:41:51 2009 From: stkn at freeswitch.org (Stefan Knoblich) Date: Mon, 5 Jan 2009 12:41:51 +0100 Subject: [Freeswitch-users] VXML support in Freeswitch In-Reply-To: <2ea4d47e0901042354r72e21bb7v9388910f0fbb21f9@mail.gmail.com> References: <2ea4d47e0901042354r72e21bb7v9388910f0fbb21f9@mail.gmail.com> Message-ID: <200901051241.51744.stkn@freeswitch.org> Am Monday 05 January 2009 schrieb Gopalakrishnan A.N: > Does freeswitch support VXML? Is there any separate module for this. > > -- > Thank you with regards, > Gopal, > Nope, it doesn't. -- Stefan Knoblich Systemadministrator axsentis GmbH Eupener Strasse 74 50933 K?ln Tel: 0180 - 506 705 521* Fax: 0180 - 506 705 529* E-Mail: s.knoblich at axsentis.de Web: www.axsentis.de Eingetragen beim AG K?ln: HR B 56238 UST-ID: DE244977565 Gesellschafter-Gesch?ftsf?hrer: Yan Lecomte, Eduard Schlein, Apostolos Varsamis *14ct/min aus dem Festnetz der T-Com | dtms From mike at jerris.com Mon Jan 5 04:48:18 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2009 07:48:18 -0500 Subject: [Freeswitch-users] VXML support in Freeswitch In-Reply-To: <2ea4d47e0901042354r72e21bb7v9388910f0fbb21f9@mail.gmail.com> References: <2ea4d47e0901042354r72e21bb7v9388910f0fbb21f9@mail.gmail.com> Message-ID: <7307E2A2-27A8-45B6-801E-040097BCC208@jerris.com> No On Jan 5, 2009, at 2:54 AM, "Gopalakrishnan A.N" wrote: > Does freeswitch support VXML? Is there any separate module for this. > > -- > Thank you with regards, > Gopal, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Mon Jan 5 05:14:34 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2009 08:14:34 -0500 Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite In-Reply-To: References: Message-ID: <34B84E83-5DA6-41A1-BC22-7BC2BC0A13F0@jerris.com> It already does Mike On Jan 5, 2009, at 5:09 AM, kriko wrote: > Is it possible for dingaling to automatically accept all new friends > who > sent me an invite? > > -- > kriko > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pmhshz at gmail.com Mon Jan 5 05:27:58 2009 From: pmhshz at gmail.com (shehzad p) Date: Mon, 5 Jan 2009 05:27:58 -0800 (PST) Subject: [Freeswitch-users] Idea/Suggestion required Message-ID: <21291002.post@talk.nabble.com> Hi All, Is there a way in freeswitch, such that we can play sound and receive a DTMF from the end user without actually answering a call like in session progress stage. Basically, the system should accept the call and play a progress tone during call progress (SIP 183). The system should be able to collect DTMF input. All this should happen while in SIP 183 (Progress) and the call shouldn't change its state to CONNECT, which means, no 200 OK should be issued to the use via the system. This is more like a IVR based system, however it all happens during call progress and not after connect. Can anyone suggest any idea about this situation? Thanks in advance. -- View this message in context: http://www.nabble.com/Idea-Suggestion-required-tp21291002p21291002.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gservat at gmail.com Mon Jan 5 05:30:57 2009 From: gservat at gmail.com (Gonzalo Servat) Date: Mon, 5 Jan 2009 11:30:57 -0200 Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite In-Reply-To: <34B84E83-5DA6-41A1-BC22-7BC2BC0A13F0@jerris.com> References: <34B84E83-5DA6-41A1-BC22-7BC2BC0A13F0@jerris.com> Message-ID: On Mon, Jan 5, 2009 at 11:14 AM, Michael Jerris wrote: > It already does > > Mike > I've noticed it does for normal Jabber servers but not for Google's "Talk" Server. It also has issues sending messages when using said server, but not with other jabber servers I've tried. - Gonzalo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/f5e5f95f/attachment.html From anthony.minessale at gmail.com Mon Jan 5 06:11:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 5 Jan 2009 08:11:23 -0600 Subject: [Freeswitch-users] Idea/Suggestion required In-Reply-To: <21291002.post@talk.nabble.com> References: <21291002.post@talk.nabble.com> Message-ID: <191c3a030901050611r3cc3705fy7691c357903c2cb@mail.gmail.com> if you make a dialplan that plays a file but does not answer then it will happen in the early media stage. DTMF working during the call is dependent on the other side. We certainly support it but some carriers purposely do not allow DTMF during progress since running an ivr in progress stage means you are not paying them any money for the calls. On Mon, Jan 5, 2009 at 7:27 AM, shehzad p wrote: > > Hi All, > Is there a way in freeswitch, such that we can play sound and receive a > DTMF > from the end user without actually answering a call like in session > progress > stage. > Basically, the system should accept the call and play a progress tone > during > call progress (SIP 183). The system should be able to collect DTMF input. > All this should happen while in SIP 183 (Progress) and the call shouldn't > change its state to CONNECT, which means, no 200 OK should be issued to the > use via the system. This is more like a IVR based system, however it all > happens during call progress and not after connect. > Can anyone suggest any idea about this situation? > > Thanks in advance. > -- > View this message in context: > http://www.nabble.com/Idea-Suggestion-required-tp21291002p21291002.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/6946f978/attachment.html From vkobashi at ydeasolutions.com.br Mon Jan 5 06:12:02 2009 From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi) Date: Mon, 05 Jan 2009 12:12:02 -0200 Subject: [Freeswitch-users] LDAP Integration In-Reply-To: <4949025F.9040008@ydeasolutions.com.br> References: <49417123.10709@ydeasolutions.com.br> <49417538.9040203@ydeasolutions.com.br> <200812120842.00808.hads@nice.net.nz> <49418790.60001@ydeasolutions.com.br> <87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com> <494198F3.10806@ydeasolutions.com.br> <4947A61F.6060806@ydeasolutions.com.br> <49485AD1.5070708@skopis.com> <4949025F.9040008@ydeasolutions.com.br> Message-ID: <49621532.5080003@ydeasolutions.com.br> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/e45d89da/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ydea.jpg Type: image/jpeg Size: 3721 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/e45d89da/attachment-0001.jpg From anthony.minessale at gmail.com Mon Jan 5 06:18:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 5 Jan 2009 08:18:44 -0600 Subject: [Freeswitch-users] VXML support in Freeswitch In-Reply-To: <7307E2A2-27A8-45B6-801E-040097BCC208@jerris.com> References: <2ea4d47e0901042354r72e21bb7v9388910f0fbb21f9@mail.gmail.com> <7307E2A2-27A8-45B6-801E-040097BCC208@jerris.com> Message-ID: <191c3a030901050618u51845a99y98fda90e1c7b3144@mail.gmail.com> we do support each of the necessary components so if some day someone were to try making a vxml module it would probably be pretty straightforward. On Mon, Jan 5, 2009 at 6:48 AM, Michael Jerris wrote: > No > > On Jan 5, 2009, at 2:54 AM, "Gopalakrishnan A.N" > wrote: > > > Does freeswitch support VXML? Is there any separate module for this. > > > > -- > > Thank you with regards, > > Gopal, > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/9f52ca7a/attachment.html From mike at jerris.com Mon Jan 5 06:42:57 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2009 09:42:57 -0500 Subject: [Freeswitch-users] busy tone detection In-Reply-To: References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com> <51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com> <2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org> <965B5A92-4182-489B-8AA0-6E6173C1931B@jerris.com> Message-ID: On Jan 5, 2009, at 2:50 AM, Baskar wrote: > Hi Michael Jerris, > > I have some questions can you answer me so that it will helpful to me > > > 1)The tone detect will work only with openzap. Am i correct?" No, it works anywhere to detect tones > > > 2)To detect with IP related media gateway like audiocode can we get > the response code from the freeswitch console? You get the response code on any call regardless of endpoint type in the response from originate. The easiest way to get to this information is in the cdr or events. > > 3)In the above mail you have told that busy number and detecting a > busy tone are COMPLETELY different things Can u explain briefly > about How to detecting a busy tone. You already had it right I think for detecting a tone, for detecting an out of band busy indication you would need to look at the cdr or events for the call. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/2fc081eb/attachment.html From mike at jerris.com Mon Jan 5 06:44:38 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2009 09:44:38 -0500 Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite In-Reply-To: References: <34B84E83-5DA6-41A1-BC22-7BC2BC0A13F0@jerris.com> Message-ID: <9F9E0C78-9AEE-401C-8FBE-A8F3CB0B20E8@jerris.com> There were some issues in earlier revisions that should now be fixed, otherwise you may not have tls support compiled in which is required for talking to googles servers. Mike On Jan 5, 2009, at 8:30 AM, Gonzalo Servat wrote: > On Mon, Jan 5, 2009 at 11:14 AM, Michael Jerris > wrote: > It already does > > Mike > > I've noticed it does for normal Jabber servers but not for Google's > "Talk" Server. It also has issues sending messages when using said > server, but not with other jabber servers I've tried. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/28e8d6da/attachment.html From benholtsclaw at averyschools.net Mon Jan 5 06:50:51 2009 From: benholtsclaw at averyschools.net (Ben Holtsclaw) Date: Mon, 5 Jan 2009 09:50:51 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 31, Issue 12 Message-ID: <200901050950520000@298828409> - Sent from my Windows Mobile Smart Phone - ------- Original Message ------- From: freeswitch-users-request at lists.freeswitch.org To: freeswitch-users at lists.freeswitch.org Sent: 1/5/09, 9:12:15 AM Subject: Freeswitch-users Digest, Vol 31, Issue 12 Send Freeswitch-users mailing list submissions to freeswitch-users at 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 at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at 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 response code in Freeswitch (Ognjen Seslija) 2. [dingaling] Gtalk accept friend invite (kriko) 3. Re: VXML support in Freeswitch (Stefan Knoblich) 4. Re: VXML support in Freeswitch (Michael Jerris) 5. Re: [dingaling] Gtalk accept friend invite (Michael Jerris) 6. Idea/Suggestion required (shehzad p) 7. Re: [dingaling] Gtalk accept friend invite (Gonzalo Servat) 8. Re: Idea/Suggestion required (Anthony Minessale) 9. Re: LDAP Integration (Vinicius Kobashi) ---------------------------------------------------------------------- Message: 1 Date: Mon, 5 Jan 2009 10:30:42 +0100 From: "Ognjen Seslija" Subject: Re: [Freeswitch-users] SIP response code in Freeswitch To: freeswitch-users at lists.freeswitch.org Message-ID: <4468a6770901050130u61ee7602x3b57b6d2495e6fdd at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, there is proto_specific_hangup_cause switch variable you can use for the cdr i.e. You can also use SIP messages number for a continue_on_fail action like: Regards, Ognjen On Mon, Jan 5, 2009 at 8:53 AM, Gopalakrishnan A.N wrote: > Hi, > Is there any possibilities that Freeswitch may detect the SIP response > code from the IP media gateway. > > -- > Thank you with regards, > Gopal, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/99083c7e/attachment-0001.html ------------------------------ Message: 2 Date: Mon, 05 Jan 2009 11:09:51 +0100 From: kriko Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite To: "freeswitch-users at lists.freeswitch.org" Message-ID: Content-Type: text/plain; format=flowed; delsp=yes; charset=utf-8 Is it possible for dingaling to automatically accept all new friends who sent me an invite? -- kriko ------------------------------ Message: 3 Date: Mon, 5 Jan 2009 12:41:51 +0100 From: Stefan Knoblich Subject: Re: [Freeswitch-users] VXML support in Freeswitch To: freeswitch-users at lists.freeswitch.org Message-ID: <200901051241.51744.stkn at freeswitch.org> Content-Type: text/plain; charset="iso-8859-1" Am Monday 05 January 2009 schrieb Gopalakrishnan A.N: > Does freeswitch support VXML? Is there any separate module for this. > > -- > Thank you with regards, > Gopal, > Nope, it doesn't. -- Stefan Knoblich Systemadministrator axsentis GmbH Eupener Strasse 74 50933 K?ln Tel: 0180 - 506 705 521* Fax: 0180 - 506 705 529* E-Mail: s.knoblich at axsentis.de Web: www.axsentis.de Eingetragen beim AG K?ln: HR B 56238 UST-ID: DE244977565 Gesellschafter-Gesch?ftsf?hrer: Yan Lecomte, Eduard Schlein, Apostolos Varsamis *14ct/min aus dem Festnetz der T-Com | dtms ------------------------------ Message: 4 Date: Mon, 5 Jan 2009 07:48:18 -0500 From: Michael Jerris Subject: Re: [Freeswitch-users] VXML support in Freeswitch To: "freeswitch-users at lists.freeswitch.org" Message-ID: <7307E2A2-27A8-45B6-801E-040097BCC208 at jerris.com> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes No On Jan 5, 2009, at 2:54 AM, "Gopalakrishnan A.N" wrote: > Does freeswitch support VXML? Is there any separate module for this. > > -- > Thank you with regards, > Gopal, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ------------------------------ Message: 5 Date: Mon, 5 Jan 2009 08:14:34 -0500 From: Michael Jerris Subject: Re: [Freeswitch-users] [dingaling] Gtalk accept friend invite To: "freeswitch-users at lists.freeswitch.org" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/00c892bf/attachment-0001.html From benholtsclaw at averyschools.net Mon Jan 5 06:51:25 2009 From: benholtsclaw at averyschools.net (Ben Holtsclaw) Date: Mon, 5 Jan 2009 09:51:25 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 31, Issue 12 Message-ID: <200901050951250000@108707933> - Sent from my Windows Mobile Smart Phone - ------- Original Message ------- From: freeswitch-users-request at lists.freeswitch.org To: freeswitch-users at lists.freeswitch.org Sent: 1/5/09, 9:12:15 AM Subject: Freeswitch-users Digest, Vol 31, Issue 12 Send Freeswitch-users mailing list submissions to freeswitch-users at 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 at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at 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 response code in Freeswitch (Ognjen Seslija) 2. [dingaling] Gtalk accept friend invite (kriko) 3. Re: VXML support in Freeswitch (Stefan Knoblich) 4. Re: VXML support in Freeswitch (Michael Jerris) 5. Re: [dingaling] Gtalk accept friend invite (Michael Jerris) 6. Idea/Suggestion required (shehzad p) 7. Re: [dingaling] Gtalk accept friend invite (Gonzalo Servat) 8. Re: Idea/Suggestion required (Anthony Minessale) 9. Re: LDAP Integration (Vinicius Kobashi) ---------------------------------------------------------------------- Message: 1 Date: Mon, 5 Jan 2009 10:30:42 +0100 From: "Ognjen Seslija" Subject: Re: [Freeswitch-users] SIP response code in Freeswitch To: freeswitch-users at lists.freeswitch.org Message-ID: <4468a6770901050130u61ee7602x3b57b6d2495e6fdd at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, there is proto_specific_hangup_cause switch variable you can use for the cdr i.e. You can also use SIP messages number for a continue_on_fail action like: Regards, Ognjen On Mon, Jan 5, 2009 at 8:53 AM, Gopalakrishnan A.N wrote: > Hi, > Is there any possibilities that Freeswitch may detect the SIP response > code from the IP media gateway. > > -- > Thank you with regards, > Gopal, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/99083c7e/attachment-0001.html ------------------------------ Message: 2 Date: Mon, 05 Jan 2009 11:09:51 +0100 From: kriko Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite To: "freeswitch-users at lists.freeswitch.org" Message-ID: Content-Type: text/plain; format=flowed; delsp=yes; charset=utf-8 Is it possible for dingaling to automatically accept all new friends who sent me an invite? -- kriko ------------------------------ Message: 3 Date: Mon, 5 Jan 2009 12:41:51 +0100 From: Stefan Knoblich Subject: Re: [Freeswitch-users] VXML support in Freeswitch To: freeswitch-users at lists.freeswitch.org Message-ID: <200901051241.51744.stkn at freeswitch.org> Content-Type: text/plain; charset="iso-8859-1" Am Monday 05 January 2009 schrieb Gopalakrishnan A.N: > Does freeswitch support VXML? Is there any separate module for this. > > -- > Thank you with regards, > Gopal, > Nope, it doesn't. -- Stefan Knoblich Systemadministrator axsentis GmbH Eupener Strasse 74 50933 K?ln Tel: 0180 - 506 705 521* Fax: 0180 - 506 705 529* E-Mail: s.knoblich at axsentis.de Web: www.axsentis.de Eingetragen beim AG K?ln: HR B 56238 UST-ID: DE244977565 Gesellschafter-Gesch?ftsf?hrer: Yan Lecomte, Eduard Schlein, Apostolos Varsamis *14ct/min aus dem Festnetz der T-Com | dtms ------------------------------ Message: 4 Date: Mon, 5 Jan 2009 07:48:18 -0500 From: Michael Jerris Subject: Re: [Freeswitch-users] VXML support in Freeswitch To: "freeswitch-users at lists.freeswitch.org" Message-ID: <7307E2A2-27A8-45B6-801E-040097BCC208 at jerris.com> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes No On Jan 5, 2009, at 2:54 AM, "Gopalakrishnan A.N" wrote: > Does freeswitch support VXML? Is there any separate module for this. > > -- > Thank you with regards, > Gopal, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ------------------------------ Message: 5 Date: Mon, 5 Jan 2009 08:14:34 -0500 From: Michael Jerris Subject: Re: [Freeswitch-users] [dingaling] Gtalk accept friend invite To: "freeswitch-users at lists.freeswitch.org" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/86d73a66/attachment.html From benholtsclaw at averyschools.net Mon Jan 5 06:52:07 2009 From: benholtsclaw at averyschools.net (Ben Holtsclaw) Date: Mon, 5 Jan 2009 09:52:07 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 31, Issue 12 Message-ID: <200901050952080000@251642128> - Sent from my Windows Mobile Smart Phone - ------- Original Message ------- From: freeswitch-users-request at lists.freeswitch.org To: freeswitch-users at lists.freeswitch.org Sent: 1/5/09, 9:12:15 AM Subject: Freeswitch-users Digest, Vol 31, Issue 12 Send Freeswitch-users mailing list submissions to freeswitch-users at 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 at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at 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 response code in Freeswitch (Ognjen Seslija) 2. [dingaling] Gtalk accept friend invite (kriko) 3. Re: VXML support in Freeswitch (Stefan Knoblich) 4. Re: VXML support in Freeswitch (Michael Jerris) 5. Re: [dingaling] Gtalk accept friend invite (Michael Jerris) 6. Idea/Suggestion required (shehzad p) 7. Re: [dingaling] Gtalk accept friend invite (Gonzalo Servat) 8. Re: Idea/Suggestion required (Anthony Minessale) 9. Re: LDAP Integration (Vinicius Kobashi) ---------------------------------------------------------------------- Message: 1 Date: Mon, 5 Jan 2009 10:30:42 +0100 From: "Ognjen Seslija" Subject: Re: [Freeswitch-users] SIP response code in Freeswitch To: freeswitch-users at lists.freeswitch.org Message-ID: <4468a6770901050130u61ee7602x3b57b6d2495e6fdd at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, there is proto_specific_hangup_cause switch variable you can use for the cdr i.e. You can also use SIP messages number for a continue_on_fail action like: Regards, Ognjen On Mon, Jan 5, 2009 at 8:53 AM, Gopalakrishnan A.N wrote: > Hi, > Is there any possibilities that Freeswitch may detect the SIP response > code from the IP media gateway. > > -- > Thank you with regards, > Gopal, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/99083c7e/attachment-0001.html ------------------------------ Message: 2 Date: Mon, 05 Jan 2009 11:09:51 +0100 From: kriko Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite To: "freeswitch-users at lists.freeswitch.org" Message-ID: Content-Type: text/plain; format=flowed; delsp=yes; charset=utf-8 Is it possible for dingaling to automatically accept all new friends who sent me an invite? -- kriko ------------------------------ Message: 3 Date: Mon, 5 Jan 2009 12:41:51 +0100 From: Stefan Knoblich Subject: Re: [Freeswitch-users] VXML support in Freeswitch To: freeswitch-users at lists.freeswitch.org Message-ID: <200901051241.51744.stkn at freeswitch.org> Content-Type: text/plain; charset="iso-8859-1" Am Monday 05 January 2009 schrieb Gopalakrishnan A.N: > Does freeswitch support VXML? Is there any separate module for this. > > -- > Thank you with regards, > Gopal, > Nope, it doesn't. -- Stefan Knoblich Systemadministrator axsentis GmbH Eupener Strasse 74 50933 K?ln Tel: 0180 - 506 705 521* Fax: 0180 - 506 705 529* E-Mail: s.knoblich at axsentis.de Web: www.axsentis.de Eingetragen beim AG K?ln: HR B 56238 UST-ID: DE244977565 Gesellschafter-Gesch?ftsf?hrer: Yan Lecomte, Eduard Schlein, Apostolos Varsamis *14ct/min aus dem Festnetz der T-Com | dtms ------------------------------ Message: 4 Date: Mon, 5 Jan 2009 07:48:18 -0500 From: Michael Jerris Subject: Re: [Freeswitch-users] VXML support in Freeswitch To: "freeswitch-users at lists.freeswitch.org" Message-ID: <7307E2A2-27A8-45B6-801E-040097BCC208 at jerris.com> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes No On Jan 5, 2009, at 2:54 AM, "Gopalakrishnan A.N" wrote: > Does freeswitch support VXML? Is there any separate module for this. > > -- > Thank you with regards, > Gopal, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ------------------------------ Message: 5 Date: Mon, 5 Jan 2009 08:14:34 -0500 From: Michael Jerris Subject: Re: [Freeswitch-users] [dingaling] Gtalk accept friend invite To: "freeswitch-users at lists.freeswitch.org" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/94aa5bb3/attachment.html From can_man at gmx.de Mon Jan 5 07:16:04 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Mon, 05 Jan 2009 16:16:04 +0100 Subject: [Freeswitch-users] XML lib curl - what is the best practice for directory binding? Message-ID: <20090105151604.264500@gmx.net> Hello, I have been looking into the xml curl directory binding and I would like to update the wiki with the accepted best practice. I have listed the HTTP POST request I am getting and how I respond. If there is a better way please let me know and I will update the wiki accordingly. Btw, what I have done works - so no bug hunting this time ;-) I will make a pylons webserver available in the next few days, starting with dialplan and directory support. Thank you, Phil At boot: HTTP POST request 1 [('hostname', u'voip'), ('section', u'directory'), ('tag_name', u''), ('key_name', u''), ('key_value', u'')] my response:
I have left the response empty as I want to provide the users at runtime. ----------------------------------------------------------------------- At boot: HTTP POST request 2 [('hostname', u'voip'), ('section', u'directory'), ('tag_name', u''), ('key_name', u''), ('key_value', u'')] my response:
----------------------------------------------------------------------- At boot: HTTP POST request 3 [('hostname', u'voip'), ('section', u'directory'), ('tag_name', u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'), ('domain', u'192.168.178.22'), ('purpose', u'network-list')] my response:
What is meant by network list here? If all the users should be loaded at boot time, is this the request which should get a response with the complete list? ---------------------------------------------------------------------- During runtime following this action:
HTTP POST request: ('hostname', u'voip'), ('section', u'directory'), ('tag_name', u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'), ('mailbox', u'315'), ('key', u'id'), ('user', u'315'), ('domain', u'192.168.178.22'), ('ip', u'217.10.79.9') my response:
//change to your domain
-- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger From kristjan.ugrin at gmail.com Mon Jan 5 08:19:51 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Mon, 05 Jan 2009 17:19:51 +0100 Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite In-Reply-To: <9F9E0C78-9AEE-401C-8FBE-A8F3CB0B20E8@jerris.com> References: <34B84E83-5DA6-41A1-BC22-7BC2BC0A13F0@jerris.com> <9F9E0C78-9AEE-401C-8FBE-A8F3CB0B20E8@jerris.com> Message-ID: Probably it has TLS support, since it logs in and I can see chat messages, place calls... I'll recheck from svn and retry, thanks. On Mon, 05 Jan 2009 15:44:38 +0100, Michael Jerris wrote: > There were some issues in earlier revisions that should now be fixed, > otherwise you may not have tls support compiled in which is required > for talking to googles servers. > > Mike > > > On Jan 5, 2009, at 8:30 AM, Gonzalo Servat wrote: > >> On Mon, Jan 5, 2009 at 11:14 AM, Michael Jerris >> wrote: >> It already does >> >> Mike >> >> I've noticed it does for normal Jabber servers but not for Google's >> "Talk" Server. It also has issues sending messages when using said >> server, but not with other jabber servers I've tried. > -- kriko From kristjan.ugrin at gmail.com Mon Jan 5 08:27:13 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Mon, 05 Jan 2009 17:27:13 +0100 Subject: [Freeswitch-users] [gtalk to sip] Ringing tone Message-ID: When placing a call between gtalk and sip, one end first picks up and then ringing starts on the other side. In that time user which answers doesn't hear any tone that could indicate a call in progress and it seems a bit odd. Is it possible to play a track in the meantime, so the caller gets a feeling that something is going on? -- kriko From gservat at gmail.com Mon Jan 5 08:34:18 2009 From: gservat at gmail.com (Gonzalo Servat) Date: Mon, 5 Jan 2009 14:34:18 -0200 Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite In-Reply-To: <9F9E0C78-9AEE-401C-8FBE-A8F3CB0B20E8@jerris.com> References: <34B84E83-5DA6-41A1-BC22-7BC2BC0A13F0@jerris.com> <9F9E0C78-9AEE-401C-8FBE-A8F3CB0B20E8@jerris.com> Message-ID: On Mon, Jan 5, 2009 at 12:44 PM, Michael Jerris wrote: > There were some issues in earlier revisions that should now be fixed, > otherwise you may not have tls support compiled in which is required for > talking to googles servers. > I've just updated and same result. I fired up Google Talk on a Windows machine then added the contact I've setup on FreeSWITCH and even with mod_dingaling in debug mode I don't see the invite-request packet come in at all so it might be Google filtering those invitations and making them so that only a Google Talk client can see and act on them. Possible? - Gonzalo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/bd453a4f/attachment.html From anthony.minessale at gmail.com Mon Jan 5 08:47:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 5 Jan 2009 10:47:42 -0600 Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite In-Reply-To: References: <34B84E83-5DA6-41A1-BC22-7BC2BC0A13F0@jerris.com> <9F9E0C78-9AEE-401C-8FBE-A8F3CB0B20E8@jerris.com> Message-ID: <191c3a030901050847k4a084bcck43baef5bb3685c75@mail.gmail.com> if you do not have a proper jabber SRV record google will not send you the requests. On Mon, Jan 5, 2009 at 10:34 AM, Gonzalo Servat wrote: > On Mon, Jan 5, 2009 at 12:44 PM, Michael Jerris wrote: > >> There were some issues in earlier revisions that should now be fixed, >> otherwise you may not have tls support compiled in which is required for >> talking to googles servers. >> > > I've just updated and same result. I fired up Google Talk on a Windows > machine then added the contact I've setup on FreeSWITCH and even with > mod_dingaling in debug mode I don't see the invite-request packet come in at > all so it might be Google filtering those invitations and making them so > that only a Google Talk client can see and act on them. Possible? > > - Gonzalo > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/df79c67e/attachment.html From mike at jerris.com Mon Jan 5 08:49:35 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jan 2009 11:49:35 -0500 Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite In-Reply-To: References: <34B84E83-5DA6-41A1-BC22-7BC2BC0A13F0@jerris.com> <9F9E0C78-9AEE-401C-8FBE-A8F3CB0B20E8@jerris.com> Message-ID: TLS is not worked into the windows build at this time, if someone would like to work up patches to support this it would be appreciated. Mike On Jan 5, 2009, at 11:34 AM, Gonzalo Servat wrote: > On Mon, Jan 5, 2009 at 12:44 PM, Michael Jerris > wrote: > There were some issues in earlier revisions that should now be > fixed, otherwise you may not have tls support compiled in which is > required for talking to googles servers. > > I've just updated and same result. I fired up Google Talk on a > Windows machine then added the contact I've setup on FreeSWITCH and > even with mod_dingaling in debug mode I don't see the invite-request > packet come in at all so it might be Google filtering those > invitations and making them so that only a Google Talk client can > see and act on them. Possible? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/bc6bcd51/attachment.html From gservat at gmail.com Mon Jan 5 09:03:51 2009 From: gservat at gmail.com (Gonzalo Servat) Date: Mon, 5 Jan 2009 15:03:51 -0200 Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite In-Reply-To: <191c3a030901050847k4a084bcck43baef5bb3685c75@mail.gmail.com> References: <34B84E83-5DA6-41A1-BC22-7BC2BC0A13F0@jerris.com> <9F9E0C78-9AEE-401C-8FBE-A8F3CB0B20E8@jerris.com> <191c3a030901050847k4a084bcck43baef5bb3685c75@mail.gmail.com> Message-ID: On Mon, Jan 5, 2009 at 2:47 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > if you do not have a proper jabber SRV record google will not send you the > requests. > I've got those setup. The jabber account I've setup on FS is a google hosted domain with the right jabber SRV records. The one I was sending a chat-invite *from* is a standard Google email account. - Gonzalo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/6c94e0c8/attachment.html From edpimentl at gmail.com Mon Jan 5 09:05:11 2009 From: edpimentl at gmail.com (EdPimentl) Date: Mon, 5 Jan 2009 12:05:11 -0500 Subject: [Freeswitch-users] [dingaling] Gtalk accept friend invite In-Reply-To: <191c3a030901050847k4a084bcck43baef5bb3685c75@mail.gmail.com> References: <34B84E83-5DA6-41A1-BC22-7BC2BC0A13F0@jerris.com> <9F9E0C78-9AEE-401C-8FBE-A8F3CB0B20E8@jerris.com> <191c3a030901050847k4a084bcck43baef5bb3685c75@mail.gmail.com> Message-ID: <9dc4a1670901050905k6f49fb2au869ef2866dfe4825@mail.gmail.com> FYI: By the end of the month we will release a Twitter Engine BOT that transparently creates public and private groups and seamlessly sends messages to: FaceBook GoogleTalk Jabber Twitter Tumblr LinkedIN Plaxo Last.FM FireEagle (GPS/Location Based Service) iPhone Ready **** BlackBerry Ready *** SmartPhone Ready *** plus many more Next is tying it with FreeSwitch service ... like Rice/Michal :-) You can see what we have thus far here... http://TwiTR.me (No charge service) Best , E http://DatR.ws (Store, Sync, Share, Publish) On Mon, Jan 5, 2009 at 11:47 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > if you do not have a proper jabber SRV record google will not send you the > requests. > > > On Mon, Jan 5, 2009 at 10:34 AM, Gonzalo Servat wrote: > >> On Mon, Jan 5, 2009 at 12:44 PM, Michael Jerris wrote: >> >>> There were some issues in earlier revisions that should now be fixed, >>> otherwise you may not have tls support compiled in which is required for >>> talking to googles servers. >>> >> >> I've just updated and same result. I fired up Google Talk on a Windows >> machine then added the contact I've setup on FreeSWITCH and even with >> mod_dingaling in debug mode I don't see the invite-request packet come in at >> all so it might be Google filtering those invitations and making them so >> that only a Google Talk client can see and act on them. Possible? >> >> - Gonzalo >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/b9fa8711/attachment-0001.html From andy at fabulous4.co.uk Mon Jan 5 09:55:39 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 5 Jan 2009 17:55:39 -0000 Subject: [Freeswitch-users] DTMF and firewall Message-ID: <8778B9B3ED1F40E0B49C812C2C8B1DBF@wsandy> Hi, I'm using freeswitch to receive incoming calls from a sip provider namely AQL. When my freeswitch box is connected directly to the internet everything works fine. When I place a firewall/router inbetween the box and the internet, the software registers with the sip provider ok and answers calls but fails to respond to in call dtmf tones. AQL advised me to make sure I was using RFC2833 which I believe I have done by setting dtmf-type in my sip profile xml to 'RFC2833'. Can anyone advise me as to what other settings I should change to make the dtmf work correctly across the firewall/router? The router is currently set to allow all traffic. Many thanks for any help you can give. regards Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/b69e5e04/attachment.html From stevecrozz at gmail.com Mon Jan 5 11:36:18 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 5 Jan 2009 11:36:18 -0800 Subject: [Freeswitch-users] using spidermonkey to get remote http status codes Message-ID: <11990ade0901051136o51af676cm281792929888ae16@mail.gmail.com> I've been working with mod_spidermonkey and an external api that I've set up. I need to determine the status code of http responses. I've tried fetchUrl() and fetchUrlHash(), but neither seems to get that status code. Any suggestions? --Stephen From jlists at skopis.com Mon Jan 5 19:55:57 2009 From: jlists at skopis.com (John Skopis (Lists)) Date: Mon, 05 Jan 2009 21:55:57 -0600 Subject: [Freeswitch-users] LDAP Integration In-Reply-To: <49621532.5080003@ydeasolutions.com.br> References: <49417123.10709@ydeasolutions.com.br> <49417538.9040203@ydeasolutions.com.br> <200812120842.00808.hads@nice.net.nz> <49418790.60001@ydeasolutions.com.br> <87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com> <494198F3.10806@ydeasolutions.com.br> <4947A61F.6060806@ydeasolutions.com.br> <49485AD1.5070708@skopis.com> <4949025F.9040008@ydeasolutions.com.br> <49621532.5080003@ydeasolutions.com.br> Message-ID: <4962D64D.3080809@skopis.com> Vinicius Kobashi wrote: > hi ppl. > > i tried hard to make it work, but still i couldnt find a complete > openldap scheme that provides these information, and i still could't > find out where to put these configuration... > > can anyone help me? > > thankz! > > vinicius escreveu: >> thankz! >> >> ill set my openldap to provide these information.. >> >> but these about these binding settings... where should i set them? >> >> best regards >> >> John Skopis (Lists) wrote: >>> vinicius wrote: >>> >>>> hi ppl.. i tried to find something at google, but i couldnt manage to find >>>> anything. >>>> i still dont know what to do to make the mod_xml_ldap work. >>>> i couldnt find information about how to build a config file for the >>>> module, and where to store it... >>>> >>>> can anyone give me a help? >>>> >>>> >>> >>> Be advised mod_xml_ldap is probably not production quality and will >>> undoubtedly change, eventually at least. >>> >>> Here is what I used once: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> which should/probably/might work with ldap objects like these: >>> >>> dn: cn=John Skopis,ou=people,dc=example >>> objectClass: person >>> objectClass: inetOrgPerson >>> objectClass: organizationalPerson >>> objectClass: FreeSWITCH-Exten-Object >>> objectClass: top >>> cn: John Skopis >>> sn: Skopis >>> givenName: John >>> FSid: 1001 >>> FSmailbox: 1001 >>> FSpassword: 1234 >>> FSvm-password: 1001 >>> FSemail-addr: john+fs at skopis.com >>> FSvm-email-all-messages: TRUE >>> FSvm-delete-file: TRUE >>> FSvm-attach-file: TRUE >>> >>> dn: SIPIdentityUserName=1001,ou=h350,dc=example >>> objectClass: person >>> objectClass: SIPIdentity >>> objectClass: top >>> cn: 1001 >>> sn: 1001 >>> SIPIdentitySIPURI: sip:1001 at 172.16.75.129 >>> SIPIdentityRegistrarAddress: 172.16.75.128 >>> SIPIdentityProxyAddress: 172.16.75.128 >>> SIPIdentityPassword: 1234 >>> SIPIdentityUserName: 1001 >>> SIPIdentityServiceLevel: premium >>> >>> Again, the module is not production quality. Hopefully I will conjurer the time and know-how to put something decent together eventually. To load configuration for any fs module you need to define the XML configuration element under the section "configuration". A good starting point is the file $PREFIX/conf/freeswitch.xml http://wiki.freeswitch.org/wiki/Freeswitch.xml Also take a look at $PREFIX/logs/freeswitch.xml.fsxml to load mod_xml_ldap you would need to add something like this to modules.conf.xml and create an xml_ldap.conf.xml in $PREFIX/autoload_configs/xml_ldap.conf.xml ... The ITU is doing some work called h.350: http://www.itu.int/ITU-T/studygroups/com16/h350/index.html Here is what I was working with: attributetype ( 1.3.6.1.4.1.65535.2.1.1 NAME 'FSid' DESC 'FreeSWITCH Extension ID' EQUALITY caseIgnoreIA5Match SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) attributetype ( 1.3.6.1.4.1.65535.2.1.2 NAME 'FSmailbox' DESC 'FreeSWITCH Extension Mailbox' EQUALITY caseIgnoreIA5Match SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) attributetype ( 1.3.6.1.4.1.65535.2.1.3 NAME 'FSpassword' DESC 'FreeSWITCH Password' EQUALITY caseExactIA5Match SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 SINGLE-VALUE ) attributetype ( 1.3.6.1.4.1.65535.2.1.4 NAME 'FSa1hash' DESC 'FreeSWITCH Crypted Password' EQUALITY caseExactIA5Match SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 SINGLE-VALUE ) attributetype ( 1.3.6.1.4.1.65535.2.1.5 NAME 'FSvm-password' DESC 'FreeSWITCH VoiceMail Password' EQUALITY integerMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.27 SINGLE-VALUE ) attributetype ( 1.3.6.1.4.1.65535.2.1.6 NAME 'FSemail-addr' DESC 'E-mail address to send voicemail' EQUALITY caseIgnoreIA5Match SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) attributetype ( 1.3.6.1.4.1.65535.2.1.7 NAME 'FSvm-email-all-messages' DESC 'FreeSWITCH Email All Mesages' EQUALITY booleanMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 SINGLE-VALUE ) attributetype ( 1.3.6.1.4.1.65535.2.1.8 NAME 'FSvm-delete-file' DESC 'FreeSWITCH VoiceMail Delete File' EQUALITY booleanMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 SINGLE-VALUE ) attributetype ( 1.3.6.1.4.1.65535.2.1.9 NAME 'FSvm-attach-file' DESC 'FreeSWITCH VoiceMail Attach file' EQUALITY booleanMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 SINGLE-VALUE ) objectclass ( 1.3.6.1.4.1.65535.2.2.1 NAME 'FreeSWITCH-Exten-Object' SUP top AUXILIARY DESC '%obj_desc%' MUST ( FSid $ FSpassword ) MAY ( FSmailbox $ FSa1hash $ FSvm-password $ FSemail-addr $ FSvm-email-all-messages $ FSvm-delete-file $ FSvm-attach-file ) ) hth From jaugenstine at gmail.com Mon Jan 5 20:43:26 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Mon, 5 Jan 2009 20:43:26 -0800 Subject: [Freeswitch-users] sofia deflect issue Message-ID: <207e7a5e0901052043r3b99d7e4v4f722e70efef179a@mail.gmail.com> I have a lua script that plays a prompt and then collects digits from an inbound call. After making a database query the call is either dropped or transferred to another server. The way I transfer the call is via the 'deflect' command. I execute that command as follows: session:execute("deflect", "user at target_server"); When I do a tcpdump, everything seems to progress as expected. I see the correct SIP messages exchanged, but after the last 200 OK is transferred Freeswitch starts sending BYE messages to the client and the client hangs up with an "Abnormal call termination". According to RFC 3515 there are no BYE messages in the protocol exchange. I have set auto hangup to false. Any ideas on what I might be doing wrong? Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090105/8643c634/attachment.html From kristjan.ugrin at gmail.com Tue Jan 6 02:21:31 2009 From: kristjan.ugrin at gmail.com (Kristjan Ugrin) Date: Tue, 6 Jan 2009 11:21:31 +0100 Subject: [Freeswitch-users] [gtalk to sip] Ringing tone In-Reply-To: References: Message-ID: <1dd3c7c0901060221x2c7d007dv3cb9a53b9de65d48@mail.gmail.com> I found out it's called ringback: http://wiki.freeswitch.org/wiki/Mod_commands#originate On Mon, Jan 5, 2009 at 5:27 PM, kriko wrote: > When placing a call between gtalk and sip, one end first picks up and then > ringing starts on the other side. > > In that time user which answers doesn't hear any tone that could indicate a > call in progress and > it seems a bit odd. > Is it possible to play a track in the meantime, so the caller gets a feeling > that something is going on? > > -- > kriko > > > From kristjan.ugrin at gmail.com Tue Jan 6 02:30:06 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Tue, 06 Jan 2009 11:30:06 +0100 Subject: [Freeswitch-users] [ringback] problems with dingaling In-Reply-To: References: Message-ID: I'm trying to entertain caller with some music or audio message so he knows that call is in progress. I tried to setup ringback like this: "api originate {ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003 at 10.99.8.221 &bridge(sofia/default/1000 at 10.99.8.221)" and it works. I can hear continuous playback of this wav file, however when calling a gtalk user: "api originate {ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003 at 10.99.8.221 &bridge(dingaling/gmail.com/my.mail at gmail.com)" there is silence only. Is this dingaling problem? Could it be solved? -- kriko From anthony.minessale at gmail.com Tue Jan 6 05:47:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Jan 2009 07:47:56 -0600 Subject: [Freeswitch-users] sofia deflect issue In-Reply-To: <207e7a5e0901052043r3b99d7e4v4f722e70efef179a@mail.gmail.com> References: <207e7a5e0901052043r3b99d7e4v4f722e70efef179a@mail.gmail.com> Message-ID: <191c3a030901060547oc995ddbkf89971c5c0772e64@mail.gmail.com> did you answer the call because deflect only works on calls that have been answered since it sends a refer. before you answer you can use the "redirect" app instead to reply with a 302 rather than blind xfer. On Mon, Jan 5, 2009 at 10:43 PM, jonathan augenstine wrote: > I have a lua script that plays a prompt and then collects digits from an > inbound call. After making a database query the call is either dropped or > transferred to another server. The way I transfer the call is via the > 'deflect' command. I execute that command as follows: > > session:execute("deflect", "user at target_server"); > > When I do a tcpdump, everything seems to progress as expected. I see the > correct SIP messages exchanged, but after the last 200 OK is transferred > Freeswitch starts sending BYE messages to the client and the client hangs up > with an "Abnormal call termination". According to RFC 3515 there are no BYE > messages in the protocol exchange. I have set auto hangup to false. Any > ideas on what I might be doing wrong? > > Jonathan > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/84f8ec92/attachment-0001.html From fidibus83 at aol.com Tue Jan 6 05:59:23 2009 From: fidibus83 at aol.com (fidibus83) Date: Tue, 6 Jan 2009 14:59:23 +0100 Subject: [Freeswitch-users] Problems with ODBC Message-ID: <007801c97006$fb3e7960$6445310a@Franzi> Hello, I need your help again! I want to use ODBC in javascripts. First I installed unixodbc and I did all the steps what are mentioned in wiki: mod spidermonkey odbc. But when I test my ODBC Setup by running isql I get this error: [ISQL]ERROR: Could not SQLConnect And I can?t start FS. I get this Error: ./freeswitch: error while loading shared libraries: libodbc.so.1: cannot open shared object file: No such file or directory I uncomment: conf/autoload_configs/spidermonkey.conf.xml build/modules.conf languages/mod_spidermonkey_odbc Here my configurations: odbc.ini [fs01_odbc] Driver=MySQL SERVER=localhost PORT=3306 DATABASE=freeswitch Socket= /var/run/mysqld/mysqld.sock odbcinst.ini [MySQL] Description = ODBC for MySQL Driver = /usr/local/lib/libodbc.so UsageCount = 1 Can you help me? What do I wrong? Thanks! Regards, fidibus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/8bd71056/attachment.html From vkobashi at ydeasolutions.com.br Tue Jan 6 06:13:21 2009 From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi) Date: Tue, 06 Jan 2009 12:13:21 -0200 Subject: [Freeswitch-users] LDAP Integration In-Reply-To: <4962D64D.3080809@skopis.com> References: <49417123.10709@ydeasolutions.com.br> <49417538.9040203@ydeasolutions.com.br> <200812120842.00808.hads@nice.net.nz> <49418790.60001@ydeasolutions.com.br> <87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com> <494198F3.10806@ydeasolutions.com.br> <4947A61F.6060806@ydeasolutions.com.br> <49485AD1.5070708@skopis.com> <4949025F.9040008@ydeasolutions.com.br> <49621532.5080003@ydeasolutions.com.br> <4962D64D.3080809@skopis.com> Message-ID: <49636701.2060207@ydeasolutions.com.br> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/d4e53e3a/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ydea.jpg Type: image/jpeg Size: 3721 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/d4e53e3a/attachment.jpg From woof at nortel.com Tue Jan 6 06:26:57 2009 From: woof at nortel.com (Andy Spitzer) Date: Tue, 06 Jan 2009 09:26:57 -0500 Subject: [Freeswitch-users] sofia deflect issue In-Reply-To: <207e7a5e0901052043r3b99d7e4v4f722e70efef179a@mail.gmail.com> References: <207e7a5e0901052043r3b99d7e4v4f722e70efef179a@mail.gmail.com> Message-ID: Woof! On Mon, 05 Jan 2009 23:43:26 -0500, jonathan augenstine wrote: > According to RFC 3515 there are no BYE messages in the protocol exchange. Once the REFER is completed (as determined by a final response returned in the NOTIFY SIPFRAG from the REFER), the original dialog can be torn down, by either side. Thus the BYE from FreeSWITCH, clearing that dialog as it has been sucessfully transfered. --Woof From jaugenstine at gmail.com Tue Jan 6 06:29:34 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Tue, 6 Jan 2009 06:29:34 -0800 Subject: [Freeswitch-users] sofia deflect issue In-Reply-To: <191c3a030901060547oc995ddbkf89971c5c0772e64@mail.gmail.com> References: <207e7a5e0901052043r3b99d7e4v4f722e70efef179a@mail.gmail.com> <191c3a030901060547oc995ddbkf89971c5c0772e64@mail.gmail.com> Message-ID: <207e7a5e0901060629h24d8670an9246d6287d7f1d98@mail.gmail.com> Yes, I do answer the call. The call flow is the lua script answers the call, plays a prompt, collects a pin code via playAndGetDigits, and then after a database lookup transfers the call via deflect. On Tue, Jan 6, 2009 at 5:47 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > did you answer the call because deflect only works on calls that have been > answered since it sends a refer. > before you answer you can use the "redirect" app instead to reply with a > 302 rather than blind xfer. > > > On Mon, Jan 5, 2009 at 10:43 PM, jonathan augenstine < > jaugenstine at gmail.com> wrote: > >> I have a lua script that plays a prompt and then collects digits from an >> inbound call. After making a database query the call is either dropped or >> transferred to another server. The way I transfer the call is via the >> 'deflect' command. I execute that command as follows: >> >> session:execute("deflect", "user at target_server"); >> >> When I do a tcpdump, everything seems to progress as expected. I see the >> correct SIP messages exchanged, but after the last 200 OK is transferred >> Freeswitch starts sending BYE messages to the client and the client hangs up >> with an "Abnormal call termination". According to RFC 3515 there are no BYE >> messages in the protocol exchange. I have set auto hangup to false. Any >> ideas on what I might be doing wrong? >> >> Jonathan >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/12de6d06/attachment.html From jaugenstine at gmail.com Tue Jan 6 06:42:28 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Tue, 6 Jan 2009 06:42:28 -0800 Subject: [Freeswitch-users] sofia deflect issue In-Reply-To: References: <207e7a5e0901052043r3b99d7e4v4f722e70efef179a@mail.gmail.com> Message-ID: <207e7a5e0901060642g620cca2ah1d62ad686ef6f248@mail.gmail.com> Thanks, I missed that. There must be a bug in Ekiga. On Tue, Jan 6, 2009 at 6:26 AM, Andy Spitzer wrote: > Woof! > > On Mon, 05 Jan 2009 23:43:26 -0500, jonathan augenstine > wrote: > > > According to RFC 3515 there are no BYE messages in the protocol exchange. > > Once the REFER is completed (as determined by a final response returned in > the NOTIFY SIPFRAG from the REFER), the original dialog can be torn down, > by either side. Thus the BYE from FreeSWITCH, clearing that dialog as it > has been sucessfully transfered. > > --Woof > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/a18f6fc2/attachment.html From raul at etellicom.com Tue Jan 6 06:29:15 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 06 Jan 2009 12:29:15 -0200 Subject: [Freeswitch-users] Problems with ODBC In-Reply-To: <007801c97006$fb3e7960$6445310a@Franzi> References: <007801c97006$fb3e7960$6445310a@Franzi> Message-ID: <1231252155.5449.1.camel@stargate> Your odbcinst.ini file doesn't seem to have the MySQL ODBC driver installed correctly. Did you install the libmyodbc package ? This is the contents of my /etc/odbcinst.ini file with a working driver for MySQL: [MySQL] Description = MySQL driver Driver = /usr/lib/odbc/libmyodbc.so Setup = /usr/lib/odbc/libodbcmyS.so CPTimeout = CPReuse = On Tue, 2009-01-06 at 14:59 +0100, fidibus83 wrote: > Hello, > > > > I need your help again! > > > > I want to use ODBC in javascripts. > > > > First I installed unixodbc and I did all the steps what are mentioned > in wiki: mod spidermonkey odbc. > > > > But when I test my ODBC Setup by running isql I get this error: > > [ISQL]ERROR: Could not SQLConnect > > > > And I can?t start FS. I get this Error: > > ./freeswitch: error while loading shared libraries: libodbc.so.1: > cannot open shared object file: No such file or directory > > > > I uncomment: > > > > conf/autoload_configs/spidermonkey.conf.xml > > > > build/modules.conf > > languages/mod_spidermonkey_odbc > > > > > > Here my configurations: > > > > odbc.ini > > > > [fs01_odbc] > > Driver=MySQL > > SERVER=localhost > > PORT=3306 > > DATABASE=freeswitch > > Socket= /var/run/mysqld/mysqld.sock > > > > odbcinst.ini > > > > [MySQL] > > Description = ODBC for MySQL > > Driver = /usr/local/lib/libodbc.so > > UsageCount = 1 > > > > Can you help me? What do I wrong? > > > > Thanks! > > > > Regards, fidibus > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Jan 6 06:45:11 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jan 2009 08:45:11 -0600 Subject: [Freeswitch-users] Problems with ODBC In-Reply-To: <1231252155.5449.1.camel@stargate> References: <007801c97006$fb3e7960$6445310a@Franzi> <1231252155.5449.1.camel@stargate> Message-ID: Chances are the paths to the .so's are wrong... I would verify that info... if you're on 64bit it would be in lib64. /b On Jan 6, 2009, at 8:29 AM, Raul Fragoso wrote: > Your odbcinst.ini file doesn't seem to have the MySQL ODBC driver > installed correctly. Did you install the libmyodbc package ? > This is the contents of my /etc/odbcinst.ini file with a working > driver > for MySQL: > > [MySQL] > Description = MySQL driver > Driver = /usr/lib/odbc/libmyodbc.so > Setup = /usr/lib/odbc/libodbcmyS.so > CPTimeout = > CPReuse = From mike at jerris.com Tue Jan 6 06:45:21 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 6 Jan 2009 09:45:21 -0500 Subject: [Freeswitch-users] Problems with ODBC In-Reply-To: <007801c97006$fb3e7960$6445310a@Franzi> References: <007801c97006$fb3e7960$6445310a@Franzi> Message-ID: That error seemsto indicatethat unixodbc is not installed or not correctly installed. Mike On Jan 6, 2009, at 8:59 AM, "fidibus83" wrote: > Hello, > > > > I need your help again! > > > > I want to use ODBC in javascripts. > > > > First I installed unixodbc and I did all the steps what are > mentioned in wiki: mod spidermonkey odbc. > > > > But when I test my ODBC Setup by running isql I get this error: > > [ISQL]ERROR: Could not SQLConnect > > > > And I can?t start FS. I get this Error: > > ./freeswitch: error while loading shared libraries: libodbc.so.1: > cannot open shared object file: No such file or directory > > > > I uncomment: > > > > conf/autoload_configs/spidermonkey.conf.xml > > > > build/modules.conf > languages/mod_spidermonkey_odbc > > > > > > Here my configurations: > > > > odbc.ini > > > > [fs01_odbc] > > Driver=MySQL > > SERVER=localhost > > PORT=3306 > > DATABASE=freeswitch > > Socket= /var/run/mysqld/mysqld.sock > > > > odbcinst.ini > > > > [MySQL] > > Description = ODBC for MySQL > > Driver = /usr/local/lib/libodbc.so > > UsageCount = 1 > > > > Can you help me? What do I wrong? > > > > Thanks! > > > > Regards, fidibus > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/6f54cdcf/attachment-0001.html From brian at freeswitch.org Tue Jan 6 06:48:52 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jan 2009 08:48:52 -0600 Subject: [Freeswitch-users] [ringback] problems with dingaling In-Reply-To: References: Message-ID: On Jan 6, 2009, at 4:30 AM, kriko wrote: > I'm trying to entertain caller with some music or audio message so > he knows that call is in progress. > I tried to setup ringback like this: > "api originate {ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003 at 10.99.8.221 > &bridge(sofia/default/1000 at 10.99.8.221)"' In this case the first leg is already answered you need to use transfer_ringback. > > > and it works. I can hear continuous playback of this wav file, > however when calling a gtalk user: > "api originate {ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003 at 10.99.8.221 > &bridge(dingaling/gmail.com/my.mail at gmail.com)" > > there is silence only. Is this dingaling problem? Could it be solved? > > -- > kriko > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From intralanman at freeswitch.org Tue Jan 6 07:12:49 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 06 Jan 2009 15:12:49 +0000 Subject: [Freeswitch-users] Problems with ODBC In-Reply-To: <007801c97006$fb3e7960$6445310a@Franzi> References: <007801c97006$fb3e7960$6445310a@Franzi> Message-ID: <496374F1.5040503@freeswitch.org> fidibus83 wrote: > > Hello, > > > > I need your help again! > > > > I want to use ODBC in javascripts. > > > > First I installed unixodbc and I did all the steps what are mentioned > in wiki: mod spidermonkey odbc. > > > > But when I test my ODBC Setup by running isql I get this error: > > [ISQL]ERROR: Could not SQLConnect > This isn't really a general purpose ODBC list, but using the -v option of isql tends to yield more verbose output (imagine that!) so you get: isql -vvv freeswitch freeswitch freeswitch [IM002][unixODBC][Driver Manager]Data source name not found, and no default driver specified [ISQL]ERROR: Could not SQLConnect as opposed to: isql freeswitch freeswitch freeswitch [ISQL]ERROR: Could not SQLConnect -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/35482086/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: intralanman.vcf Type: text/x-vcard Size: 209 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/35482086/attachment.vcf From fidibus83 at aol.com Tue Jan 6 07:30:24 2009 From: fidibus83 at aol.com (fidibus83) Date: Tue, 6 Jan 2009 16:30:24 +0100 Subject: [Freeswitch-users] Problems with ODBC In-Reply-To: <496374F1.5040503@freeswitch.org> References: <007801c97006$fb3e7960$6445310a@Franzi> <496374F1.5040503@freeswitch.org> Message-ID: <00b301c97013$b4169c40$6445310a@Franzi> The libmyodbc package is installed. But I see I have a lot of version of libmyodbc: libmyodbc3.so libmyodbc3_r.so libmyodbc3_r-3.51.14.so libmyodbc3-3.51.14.so I changed odbcinst.ini: [MySQL] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc3.so Setup = /usr/lib/libodbcmyS.so I run isgl ?vvv and I get this: [08S01] [unixODBC] [MySQL] [ODBC 3.51 Driver] Can?t connect to local MySQL server through socket ?/var/run/mysqld/mysqld.sock? (2) [ISQL] ERROR: Could not SQLConnect _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Raymond Chandler Gesendet: Dienstag, 6. Januar 2009 16:13 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] Problems with ODBC fidibus83 wrote: Hello, I need your help again! I want to use ODBC in javascripts. First I installed unixodbc and I did all the steps what are mentioned in wiki: mod spidermonkey odbc. But when I test my ODBC Setup by running isql I get this error: [ISQL]ERROR: Could not SQLConnect This isn't really a general purpose ODBC list, but using the -v option of isql tends to yield more verbose output (imagine that!) so you get: isql -vvv freeswitch freeswitch freeswitch [IM002][unixODBC][Driver Manager]Data source name not found, and no default driver specified [ISQL]ERROR: Could not SQLConnect as opposed to: isql freeswitch freeswitch freeswitch [ISQL]ERROR: Could not SQLConnect -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/86b1dc6c/attachment.html From jforman at wcgltd.com Tue Jan 6 07:54:47 2009 From: jforman at wcgltd.com (Josh Forman) Date: Tue, 6 Jan 2009 10:54:47 -0500 Subject: [Freeswitch-users] mod_opal calls and records Message-ID: <4F14448B-6DD4-4B05-96A5-22289C8E5093@wcgltd.com> I started to do some testing with h323 calls using mod_opal and I have been having a number of issues. I've been creating calls that come into freeswitch as SIP and output as h323 and while I have a call open I did not see any entries from fs_cli when using the "show channels" or "show calls" commands. Also no CDR is generated from the call. I was wondering if someone could tell me if these issues are from something that just has not been implemented yet in mod_opal or if this is most likely a configuration issue on my side. Thank you, Josh From Prometheus001 at gmx.net Tue Jan 6 08:58:41 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 06 Jan 2009 17:58:41 +0100 Subject: [Freeswitch-users] different behaviour on uuid_bridge, doesn't really bridge Message-ID: <49638DC1.2000307@gmx.net> I have setup a test machine and a production machine. Since recently the production machine behaves differently in terms of uuid bridge. How it should work (and how it worked before) 1) call A comes in 2) I play some messages to A 3) In the meantime I originate a call to B and transfer to an extension, where also some messages are played 4) Then I bridge A and B, so they are dropped off the current announcements an speak to each other 5) when either A or B hangs up, both legs are terminated New behaviour 1) call A comes in 2) I play some messages to A 3) In the meantime I originate a call to B and transfer to an extension, where also some messages are played 4) Then I bridge A and B, A and B can hear each other for 1/2 sec, then A constinues to hear its messages, B does not hear anything 5) when either A or B hangs up, both legs are terminated 4) is different now! The FS console show some messages about unbridge (SWITCH_MESSAGE_INDICATE_UNBRIDGE), but somehow the calls are still connected until A or B hangs up. Anybody has a clue? Best regards Peter 3) is finished, 4) starts 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] 2009-01-06 17:35:31 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 98 (0x62) 2009-01-06 17:35:34 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1094 switch_ivr_uuid_bridge() (OpenZAP/2:3/49171xxxxxxx) State Change CS_EXECUTE -> CS_RESET 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 switch_core_session_signal_state_change() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1095 switch_ivr_uuid_bridge() (OpenZAP/2:1/216xxxxx) State Change CS_EXECUTE -> CS_RESET 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:35:35 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (OpenZAP/2:1/216xxxxx) State EXECUTE going to sleep 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State Change CS_RESET 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET 4) Start 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 uuid_bridge_on_reset() OpenZAP/2:1/216xxxxx CUSTOM RESET 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reset() OpenZAP/2:1/216xxxxx Standard RESET 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET going to sleep 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State EXECUTE going to sleep 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) Running State Change CS_RESET 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State RESET 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 uuid_bridge_on_reset() OpenZAP/2:3/49171xxxxxxx CUSTOM RESET 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:533 uuid_bridge_on_reset() (OpenZAP/2:3/49171xxxxxxx) State Change CS_RESET -> CS_SOFT_EXECUTE 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 switch_core_session_signal_state_change() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State RESET going to sleep 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) Running State Change CS_SOFT_EXECUTE 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State SOFT_EXECUTE 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 channel_on_soft_execute() CHANNEL SOFT_EXECUTE 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 uuid_bridge_on_soft_execute() OpenZAP/2:3/49171xxxxxxx CUSTOM TRANSMIT 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:571 uuid_bridge_on_soft_execute() (OpenZAP/2:1/216xxxxx) State Change CS_RESET -> CS_SOFT_EXECUTE 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State Change CS_SOFT_EXECUTE 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (OpenZAP/2:1/216xxxxx) State SOFT_EXECUTE 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 channel_on_soft_execute() CHANNEL SOFT_EXECUTE 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 uuid_bridge_on_soft_execute() OpenZAP/2:1/216xxxxx CUSTOM TRANSMIT 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:181 switch_core_standard_on_soft_execute() OpenZAP/2:1/216xxxxx Standard SOFT_EXECUTE 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (OpenZAP/2:1/216xxxxx) State SOFT_EXECUTE going to sleep 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:814 switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) State Change CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:856 switch_ivr_multi_threaded_bridge() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_BRIDGE] 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:863 switch_ivr_multi_threaded_bridge() OpenZAP/2:3/49171xxxxxxx receive message [SWITCH_MESSAGE_INDICATE_BRIDGE] 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:667 switch_core_session_queue_private_event() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:907 switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State Change CS_EXCHANGE_MEDIA 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:436 switch_core_session_run() (OpenZAP/2:1/216xxxxx) State EXCHANGE_MEDIA 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:511 channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:667 switch_core_session_queue_private_event() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_UNBRIDGE] 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() OpenZAP/2:1/216xxxxx Command Execute read(0 1 custom/warteschleife_30.wav interrupt_digit 0 ) 2009-01-06 17:35:36 [DEBUG] switch_ivr_play_say.c:932 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-01-06 17:35:36 [DEBUG] switch_core_io.c:610 switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message [SWITCH_MESSAGE_INDICATE_UNBRIDGE] 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() OpenZAP/2:3/49171xxxxxxx Command Execute stop_dtmf(true) 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:229 audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message [SWITCH_MESSAGE_INDICATE_BRIDGE] 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:231 audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:667 switch_core_session_queue_private_event() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:226 audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message [SWITCH_MESSAGE_INDICATE_UNBRIDGE] 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:229 audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message [SWITCH_MESSAGE_INDICATE_BRIDGE] 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:231 audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 31 -------------------------------------------------------------------------------- [08 02 80 04 62 1c 18 91 a1 15 02 01 7d 02 01 22 30 0d a1 05 30 03 02 01 08 82 01 00 83 01 00] 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] 2009-01-06 17:35:46 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 98 (0x62) 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 31 -------------------------------------------------------------------------------- [08 02 80 04 62 1c 18 91 a1 15 02 01 7e 02 01 22 30 0d a1 05 30 03 02 01 09 82 01 00 83 01 00] 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] 2009-01-06 17:36:02 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 98 (0x62) 2009-01-06 17:36:06 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:229 audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_BRIDGE] 2009-01-06 17:36:07 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:231 audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:667 switch_core_session_queue_private_event() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:08 [DEBUG] switch_ivr_bridge.c:226 audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_UNBRIDGE] 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:08 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() OpenZAP/2:1/216xxxxx Command Execute read(0 1 custom/warteschleife_30.wav interrupt_digit 0 ) 2009-01-06 17:36:08 [DEBUG] switch_ivr_play_say.c:932 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-01-06 17:36:08 [DEBUG] switch_core_io.c:610 switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 31 -------------------------------------------------------------------------------- [08 02 80 04 62 1c 18 91 a1 15 02 01 7f 02 01 22 30 0d a1 05 30 03 02 01 0a 82 01 00 83 01 00] 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] 2009-01-06 17:36:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 98 (0x62) 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 32 -------------------------------------------------------------------------------- [08 02 80 04 62 1c 19 91 a1 16 02 02 00 80 02 01 22 30 0d a1 05 30 03 02 01 0b 82 01 00 83 01 00] 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] 2009-01-06 17:36:33 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 98 (0x62) 2009-01-06 17:36:38 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:229 audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_BRIDGE] 2009-01-06 17:36:39 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:231 audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:667 switch_core_session_queue_private_event() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:40 [DEBUG] switch_ivr_bridge.c:226 audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_UNBRIDGE] 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:40 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() OpenZAP/2:1/216xxxxx Command Execute read(0 1 custom/dieser_berater_ist_leider_zur_zeit_nicht_erreichbar_bitte_waehlen_sie_einen_ande.wav interrupt_digit 0 ) 2009-01-06 17:36:40 [DEBUG] switch_ivr_play_say.c:932 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-01-06 17:36:40 [DEBUG] switch_core_io.c:610 switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:45 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:229 audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_BRIDGE] 2009-01-06 17:36:46 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:231 audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:667 switch_core_session_queue_private_event() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:47 [DEBUG] switch_ivr_bridge.c:226 audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_UNBRIDGE] 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:47 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() OpenZAP/2:1/216xxxxx Command Execute read(0 4 custom/geben_sie_bitte_nach_dem_signalton_die_kennung_des_gewuenschten_beraters_ein_sch2.wav dtmfdtmf 10000 #,*) 2009-01-06 17:36:47 [DEBUG] switch_ivr_play_say.c:932 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-01-06 17:36:47 [DEBUG] switch_core_io.c:610 switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 32 -------------------------------------------------------------------------------- [08 02 80 04 62 1c 19 91 a1 16 02 02 00 81 02 01 22 30 0d a1 05 30 03 02 01 0c 82 01 00 83 01 00] 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] 2009-01-06 17:36:48 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 98 (0x62) 2009-01-06 17:37:03 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 32 -------------------------------------------------------------------------------- [08 02 80 04 62 1c 19 91 a1 16 02 02 00 82 02 01 22 30 0d a1 05 30 03 02 01 0d 82 01 00 83 01 00] 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] 2009-01-06 17:37:04 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 98 (0x62) 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 13 -------------------------------------------------------------------------------- [08 02 00 35 45 08 02 80 90 1e 02 82 88] 5) Hangup 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[45] Size:[115] CRV: 53 (0x35, CTX: Originator) 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 77df70 (2:1) source isdn_data->channels_remote_crv[0x35] 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:455 zap_isdn_931_34() Changing state on 2:1 from UP to TERMINATING 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:813 state_advance() 2:1 STATE [TERMINATING] 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1121 state_advance() Terminating: Direction Inbound 2009-01-06 17:37:12 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() got clear channel sig [STOP] 2009-01-06 17:37:12 [NOTICE] mod_openzap.c:1437 on_clear_channel_signal() Hangup OpenZAP/2:1/216xxxxx [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-01-06 17:37:12 [DEBUG] switch_channel.c:1441 switch_channel_perform_hangup() Send signal OpenZAP/2:1/216xxxxx [KILL] 2009-01-06 17:37:12 [DEBUG] switch_core_session.c:779 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/216xxxxx [BREAK] 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 From intralanman at freeswitch.org Tue Jan 6 09:12:50 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 06 Jan 2009 17:12:50 +0000 Subject: [Freeswitch-users] Problems with ODBC In-Reply-To: <00b301c97013$b4169c40$6445310a@Franzi> References: <007801c97006$fb3e7960$6445310a@Franzi> <496374F1.5040503@freeswitch.org> <00b301c97013$b4169c40$6445310a@Franzi> Message-ID: <49639112.7080902@freeswitch.org> is mysql running on the localhost? if so, is that the correct path to the .sock? you might also try using 127.0.0.1 as the host instead of localhost to force a network connection. -Ray fidibus83 wrote: > > The libmyodbc package is installed. > > > > But I see I have a lot of version of libmyodbc: > > libmyodbc3.so > > libmyodbc3_r.so > > libmyodbc3_r-3.51.14.so > > libmyodbc3-3.51.14.so > > > > I changed odbcinst.ini: > > > > [MySQL] > > Description = ODBC for MySQL > > Driver = /usr/lib/libmyodbc3.so > > Setup = /usr/lib/libodbcmyS.so > > > > I run isgl --vvv and I get this: > > > > [08S01] [unixODBC] [MySQL] [ODBC 3.51 Driver] Can't connect to local > MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) > > [ISQL] ERROR: Could not SQLConnect > > > > ------------------------------------------------------------------------ > > *Von:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Raymond Chandler > *Gesendet:* Dienstag, 6. Januar 2009 16:13 > *An:* freeswitch-users at lists.freeswitch.org > *Betreff:* Re: [Freeswitch-users] Problems with ODBC > > > > fidibus83 wrote: > > Hello, > > > > I need your help again! > > > > I want to use ODBC in javascripts. > > > > First I installed unixodbc and I did all the steps what are mentioned > in wiki: mod spidermonkey odbc. > > > > But when I test my ODBC Setup by running isql I get this error: > > [ISQL]ERROR: Could not SQLConnect > > This isn't really a general purpose ODBC list, but using the -v option > of isql tends to yield more verbose output (imagine that!) > > so you get: > isql -vvv freeswitch freeswitch freeswitch > [IM002][unixODBC][Driver Manager]Data source name not found, and no > default driver specified > [ISQL]ERROR: Could not SQLConnect > > > as opposed to: > isql freeswitch freeswitch freeswitch > [ISQL]ERROR: Could not SQLConnect > > -Ray > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/48df1fca/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: intralanman.vcf Type: text/x-vcard Size: 209 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/48df1fca/attachment.vcf From Prometheus001 at gmx.net Tue Jan 6 10:25:44 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 06 Jan 2009 19:25:44 +0100 Subject: [Freeswitch-users] different behaviour on uuid_bridge, doesn't really bridge In-Reply-To: <49638DC1.2000307@gmx.net> References: <49638DC1.2000307@gmx.net> Message-ID: <4963A228.3000103@gmx.net> One more info: I have updated to the newest SVN version of FS. A and B can actually hear each other (just a bit, some scratching) while the announcement to A is very slow (~50% speed) and very choppy. Best regards peter Peter P GMX schrieb: > I have setup a test machine and a production machine. Since recently the > production machine behaves differently in terms of uuid bridge. > > How it should work (and how it worked before) > 1) call A comes in > 2) I play some messages to A > 3) In the meantime I originate a call to B and transfer to an > extension, where also some messages are played > 4) Then I bridge A and B, so they are dropped off the current > announcements an speak to each other > 5) when either A or B hangs up, both legs are terminated > > New behaviour > 1) call A comes in > 2) I play some messages to A > 3) In the meantime I originate a call to B and transfer to an > extension, where also some messages are played > 4) Then I bridge A and B, A and B can hear each other for 1/2 sec, > then A constinues to hear its messages, B does not hear anything > 5) when either A or B hangs up, both legs are terminated > > 4) is different now! > > The FS console show some messages about unbridge > (SWITCH_MESSAGE_INDICATE_UNBRIDGE), but somehow the calls are still > connected until A or B hangs up. > > Anybody has a clue? > Best regards > Peter > > 3) is finished, 4) starts > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > 2009-01-06 17:35:31 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > unhandled message 98 (0x62) > 2009-01-06 17:35:34 [DEBUG] switch_ivr_play_say.c:1222 > switch_ivr_play_file() done playing file > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1094 > switch_ivr_uuid_bridge() (OpenZAP/2:3/49171xxxxxxx) State Change > CS_EXECUTE -> CS_RESET > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > switch_core_session_signal_state_change() Send signal > OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1095 > switch_ivr_uuid_bridge() (OpenZAP/2:1/216xxxxx) State Change CS_EXECUTE > -> CS_RESET > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:35:35 [DEBUG] switch_ivr_play_say.c:1222 > switch_ivr_play_file() done playing file > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State EXECUTE going to > sleep > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State Change > CS_RESET > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET > 4) Start > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > uuid_bridge_on_reset() OpenZAP/2:1/216xxxxx CUSTOM RESET > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reset() OpenZAP/2:1/216xxxxx Standard RESET > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET going to sleep > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State EXECUTE going > to sleep > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) Running State > Change CS_RESET > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State RESET > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > uuid_bridge_on_reset() OpenZAP/2:3/49171xxxxxxx CUSTOM RESET > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:533 > uuid_bridge_on_reset() (OpenZAP/2:3/49171xxxxxxx) State Change CS_RESET > -> CS_SOFT_EXECUTE > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > switch_core_session_signal_state_change() Send signal > OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State RESET going > to sleep > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) Running State > Change CS_SOFT_EXECUTE > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State SOFT_EXECUTE > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 channel_on_soft_execute() > CHANNEL SOFT_EXECUTE > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > uuid_bridge_on_soft_execute() OpenZAP/2:3/49171xxxxxxx CUSTOM TRANSMIT > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:571 > uuid_bridge_on_soft_execute() (OpenZAP/2:1/216xxxxx) State Change > CS_RESET -> CS_SOFT_EXECUTE > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State Change > CS_SOFT_EXECUTE > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State SOFT_EXECUTE > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 channel_on_soft_execute() > CHANNEL SOFT_EXECUTE > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > uuid_bridge_on_soft_execute() OpenZAP/2:1/216xxxxx CUSTOM TRANSMIT > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:181 > switch_core_standard_on_soft_execute() OpenZAP/2:1/216xxxxx Standard > SOFT_EXECUTE > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State SOFT_EXECUTE > going to sleep > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:814 > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) State Change > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:856 > switch_ivr_multi_threaded_bridge() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_BRIDGE] > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:863 > switch_ivr_multi_threaded_bridge() OpenZAP/2:3/49171xxxxxxx receive > message [SWITCH_MESSAGE_INDICATE_BRIDGE] > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:667 > switch_core_session_queue_private_event() Send signal > OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:907 > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) State Change > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State Change > CS_EXCHANGE_MEDIA > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:436 > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State EXCHANGE_MEDIA > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:511 > channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:667 > switch_core_session_queue_private_event() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > custom/warteschleife_30.wav interrupt_digit 0 ) > 2009-01-06 17:35:36 [DEBUG] switch_ivr_play_say.c:932 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-01-06 17:35:36 [DEBUG] switch_core_io.c:610 > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() > OpenZAP/2:3/49171xxxxxxx Command Execute stop_dtmf(true) > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:229 > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > [SWITCH_MESSAGE_INDICATE_BRIDGE] > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:231 > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:667 > switch_core_session_queue_private_event() Send signal > OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:226 > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:229 > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > [SWITCH_MESSAGE_INDICATE_BRIDGE] > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:231 > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 31 > -------------------------------------------------------------------------------- > [08 02 80 04 62 1c 18 91 a1 15 02 01 7d 02 01 22 30 0d a1 05 30 03 02 01 > 08 82 01 00 83 01 00] > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > 2009-01-06 17:35:46 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > unhandled message 98 (0x62) > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 31 > -------------------------------------------------------------------------------- > [08 02 80 04 62 1c 18 91 a1 15 02 01 7e 02 01 22 30 0d a1 05 30 03 02 01 > 09 82 01 00 83 01 00] > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > 2009-01-06 17:36:02 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > unhandled message 98 (0x62) > 2009-01-06 17:36:06 [DEBUG] switch_ivr_play_say.c:1222 > switch_ivr_play_file() done playing file > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:229 > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_BRIDGE] > 2009-01-06 17:36:07 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:231 > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:667 > switch_core_session_queue_private_event() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:08 [DEBUG] switch_ivr_bridge.c:226 > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:08 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > custom/warteschleife_30.wav interrupt_digit 0 ) > 2009-01-06 17:36:08 [DEBUG] switch_ivr_play_say.c:932 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-01-06 17:36:08 [DEBUG] switch_core_io.c:610 > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 31 > -------------------------------------------------------------------------------- > [08 02 80 04 62 1c 18 91 a1 15 02 01 7f 02 01 22 30 0d a1 05 30 03 02 01 > 0a 82 01 00 83 01 00] > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > 2009-01-06 17:36:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > unhandled message 98 (0x62) > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 32 > -------------------------------------------------------------------------------- > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 80 02 01 22 30 0d a1 05 30 03 02 > 01 0b 82 01 00 83 01 00] > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > 2009-01-06 17:36:33 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > unhandled message 98 (0x62) > 2009-01-06 17:36:38 [DEBUG] switch_ivr_play_say.c:1222 > switch_ivr_play_file() done playing file > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:229 > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_BRIDGE] > 2009-01-06 17:36:39 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:231 > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:667 > switch_core_session_queue_private_event() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:40 [DEBUG] switch_ivr_bridge.c:226 > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:40 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > custom/dieser_berater_ist_leider_zur_zeit_nicht_erreichbar_bitte_waehlen_sie_einen_ande.wav > interrupt_digit 0 ) > 2009-01-06 17:36:40 [DEBUG] switch_ivr_play_say.c:932 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-01-06 17:36:40 [DEBUG] switch_core_io.c:610 > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:45 [DEBUG] switch_ivr_play_say.c:1222 > switch_ivr_play_file() done playing file > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:229 > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_BRIDGE] > 2009-01-06 17:36:46 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:231 > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:667 > switch_core_session_queue_private_event() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:47 [DEBUG] switch_ivr_bridge.c:226 > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:47 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() > OpenZAP/2:1/216xxxxx Command Execute read(0 4 > custom/geben_sie_bitte_nach_dem_signalton_die_kennung_des_gewuenschten_beraters_ein_sch2.wav > dtmfdtmf 10000 #,*) > 2009-01-06 17:36:47 [DEBUG] switch_ivr_play_say.c:932 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-01-06 17:36:47 [DEBUG] switch_core_io.c:610 > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 32 > -------------------------------------------------------------------------------- > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 81 02 01 22 30 0d a1 05 30 03 02 > 01 0c 82 01 00 83 01 00] > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > 2009-01-06 17:36:48 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > unhandled message 98 (0x62) > 2009-01-06 17:37:03 [DEBUG] switch_ivr_play_say.c:1222 > switch_ivr_play_file() done playing file > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 32 > -------------------------------------------------------------------------------- > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 82 02 01 22 30 0d a1 05 30 03 02 > 01 0d 82 01 00 83 01 00] > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > 2009-01-06 17:37:04 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > unhandled message 98 (0x62) > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 13 > -------------------------------------------------------------------------------- > [08 02 00 35 45 08 02 80 90 1e 02 82 88] > > 5) Hangup > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[45] Size:[115] CRV: 53 (0x35, CTX: Originator) > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > 77df70 (2:1) source isdn_data->channels_remote_crv[0x35] > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:455 zap_isdn_931_34() Changing > state on 2:1 from UP to TERMINATING > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:813 state_advance() 2:1 STATE > [TERMINATING] > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1121 state_advance() > Terminating: Direction Inbound > 2009-01-06 17:37:12 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() > got clear channel sig [STOP] > 2009-01-06 17:37:12 [NOTICE] mod_openzap.c:1437 > on_clear_channel_signal() Hangup OpenZAP/2:1/216xxxxx > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-01-06 17:37:12 [DEBUG] switch_channel.c:1441 > switch_channel_perform_hangup() Send signal OpenZAP/2:1/216xxxxx [KILL] > 2009-01-06 17:37:12 [DEBUG] switch_core_session.c:779 > switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 > > > > From msc at freeswitch.org Tue Jan 6 11:54:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Jan 2009 11:54:02 -0800 Subject: [Freeswitch-users] New FreeSWITCH release Message-ID: <87f2f3b90901061154o125d8b13v140ab3440f1980e0@mail.gmail.com> FYI, If you haven't already heard, we've now released v1.0.2 of FreeSWITCH! Please digg the new release story: http://digg.com/software/FreeSWITCH_New_Release_For_The_New_Year The source can be downloaded here: http://files.freeswitch.org/ in both tar.gz and tar.bz2 formats A Windows MSI file can be downloaded here as well: http://files.freeswitch.org/freeswitch-1.0.2.msi Thanks for your support and keep on FreeSWITCHing! -MC (mercutioviz) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/af75d9b5/attachment.html From akm.freeswitchuser at gmail.com Tue Jan 6 06:44:12 2009 From: akm.freeswitchuser at gmail.com (Aadil M) Date: Tue, 6 Jan 2009 20:14:12 +0530 Subject: [Freeswitch-users] Need some info Message-ID: <3cbb52a70901060644l683d4ab5hcd5c01d2621c0b7d@mail.gmail.com> Hi All, I am a new comer to FreeSwitch. The thing is I am planning to setup a call center along with a few of my friends. Hence I was looking around for some IP PBX and so stumbled upon FreeSwitch. This call center would be a 10 to 20 person setup. Here are a few of the questions that I have? 1. Can I have just on PSTN line and configure multiple extensions on FS by using relevent hardware?If not then how many PSTN lines would I require? 2. I did Google for the hardware, it seems I would require some FXO/FXS cards. Am I right? Where can I get them? I am sorry if you find these question too novice or lame. Kindly forgive me. Looking for a positive reply from the community. Regards, Aadil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/d6334563/attachment.html From markgreene at gmail.com Tue Jan 6 11:42:59 2009 From: markgreene at gmail.com (Mark Greene) Date: Tue, 6 Jan 2009 13:42:59 -0600 Subject: [Freeswitch-users] BLF on Aastra 9133i Message-ID: <8ecbc2000901061142s3d5140cel43e6f146b828ba9b@mail.gmail.com> Hey guys, I have a user who is in charge of checking multiple voicemail boxes. They are registered under extension 0650, which will have voicemails as well. So the user has a total of 3 VM boxes to check. The MWI light works great for thier extension (0650), but I can't use it for the other voicemail boxes that the user is in charge of checking without asking the user to call and check each mailbox every time the MWI light comes on. The Aastra 9133i has 7 programable keys. One of the many options for programming the keys is to assign keys to be BLF keys. So I think a good solution is to label two of the programable keys VM Box 2 and VM Box 3 and set them to be BLF keys. Then when a VM is left on one of those boxes I can just have an action that turns the BLF on for the correct key. Can this be done with FS. How could I go about doing it? Thanks in advance, - Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/dc30894a/attachment.html From msc at freeswitch.org Tue Jan 6 12:13:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Jan 2009 12:13:08 -0800 Subject: [Freeswitch-users] Need some info In-Reply-To: <3cbb52a70901060644l683d4ab5hcd5c01d2621c0b7d@mail.gmail.com> References: <3cbb52a70901060644l683d4ab5hcd5c01d2621c0b7d@mail.gmail.com> Message-ID: <87f2f3b90901061213s6b1023eeua993428854c3197b@mail.gmail.com> Welcome to FreeSWITCH! On Tue, Jan 6, 2009 at 6:44 AM, Aadil M wrote: > Hi All, > > I am a new comer to FreeSwitch. The thing is I am planning to setup a call > center along with a few of my friends. Hence I was looking around for some > IP PBX and so stumbled upon FreeSwitch. > This call center would be a 10 to 20 person setup. > Here are a few of the questions that I have? > > 1. Can I have just on PSTN line and configure multiple extensions on FS by > using relevent hardware?If not then how many PSTN lines would I require? > > If you are using a VoIP provider then you won't need PSTN lines. However, if you do need PSTN then it is a matter of analog or digital. You need to know what your call traffic volume is like before you can decide how many and what types of cards. Do you know what kind of call traffic you will have? > 2. I did Google for the hardware, it seems I would require some FXO/FXS > cards. Am I right? Where can I get them? > > Sangoma, Rhino Technologies, pbxhardware.com, etc. - Anything that is "Asterisk" compatible will work with FreeSWITCH. Note: The FreeSWITCH PRI stack is still in development; it works but has a few bugs. > I am sorry if you find these question too novice or lame. Kindly forgive > me. > Looking for a positive reply from the community. > > You may also want to join us on IRC: www.freenode.net, #freeswitch channel. -MC (mercutioviz) > Regards, > Aadil > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/ab128380/attachment.html From kristian.kielhofner at gmail.com Tue Jan 6 12:44:43 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 6 Jan 2009 15:44:43 -0500 Subject: [Freeswitch-users] vmd Message-ID: <2d9149cd0901061244t51a4ae90s3ac5f9db6f90192d@mail.gmail.com> Hello everyone, I know 1.0.2 if fresh (yeah!) but what exactly is the status of vmd? The wiki page says it can't be used as an app yet it shows up in "show application" with start/stop. How might this be used from the dialplan? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From msc at freeswitch.org Tue Jan 6 13:50:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Jan 2009 13:50:39 -0800 Subject: [Freeswitch-users] vmd In-Reply-To: <2d9149cd0901061244t51a4ae90s3ac5f9db6f90192d@mail.gmail.com> References: <2d9149cd0901061244t51a4ae90s3ac5f9db6f90192d@mail.gmail.com> Message-ID: <87f2f3b90901061350q2cb6c331p3dffd51e42b5ac04@mail.gmail.com> Kristian, It can be used from the dialplan (I'm pretty sure) but all is does is set a channel variable. It doesn't actually do anything else. It was originally designed to be used from the event socket. I added the code to set the channel variable as a proof-of-concept... If you need it to do more then we'll have to roll up the sleeves and do some coding. :) -MC On Tue, Jan 6, 2009 at 12:44 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > I know 1.0.2 if fresh (yeah!) but what exactly is the status of vmd? > > The wiki page says it can't be used as an app yet it shows up in > "show application" with start/stop. How might this be used from the > dialplan? > > Thanks! > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/78eb2908/attachment.html From kristian.kielhofner at gmail.com Tue Jan 6 13:58:57 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 6 Jan 2009 16:58:57 -0500 Subject: [Freeswitch-users] vmd In-Reply-To: <87f2f3b90901061350q2cb6c331p3dffd51e42b5ac04@mail.gmail.com> References: <2d9149cd0901061244t51a4ae90s3ac5f9db6f90192d@mail.gmail.com> <87f2f3b90901061350q2cb6c331p3dffd51e42b5ac04@mail.gmail.com> Message-ID: <2d9149cd0901061358j8fff6ecvbd3d217968aa265e@mail.gmail.com> On 1/6/09, Michael Collins wrote: > Kristian, > > It can be used from the dialplan (I'm pretty sure) but all is does is set a > channel variable. It doesn't actually do anything else. It was originally > designed to be used from the event socket. I added the code to set the > channel variable as a proof-of-concept... > > If you need it to do more then we'll have to roll up the sleeves and do some > coding. :) > > -MC > Michael, That might be enough for what I'm trying to do... What is this variable? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From msc at freeswitch.org Tue Jan 6 14:52:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Jan 2009 14:52:43 -0800 Subject: [Freeswitch-users] vmd In-Reply-To: <2d9149cd0901061358j8fff6ecvbd3d217968aa265e@mail.gmail.com> References: <2d9149cd0901061244t51a4ae90s3ac5f9db6f90192d@mail.gmail.com> <87f2f3b90901061350q2cb6c331p3dffd51e42b5ac04@mail.gmail.com> <2d9149cd0901061358j8fff6ecvbd3d217968aa265e@mail.gmail.com> Message-ID: <87f2f3b90901061452y16b5f4ado68503fd59a9eaa70@mail.gmail.com> See my other email and you can glean it from there... -MC On Tue, Jan 6, 2009 at 1:58 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > On 1/6/09, Michael Collins wrote: > > Kristian, > > > > It can be used from the dialplan (I'm pretty sure) but all is does is set > a > > channel variable. It doesn't actually do anything else. It was originally > > designed to be used from the event socket. I added the code to set the > > channel variable as a proof-of-concept... > > > > If you need it to do more then we'll have to roll up the sleeves and do > some > > coding. :) > > > > -MC > > > > Michael, > > That might be enough for what I'm trying to do... What is this variable? > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/cf7a50f9/attachment-0001.html From anthony.minessale at gmail.com Tue Jan 6 16:09:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Jan 2009 18:09:42 -0600 Subject: [Freeswitch-users] different behaviour on uuid_bridge, doesn't really bridge In-Reply-To: <4963A228.3000103@gmx.net> References: <49638DC1.2000307@gmx.net> <4963A228.3000103@gmx.net> Message-ID: <191c3a030901061609m19d91c15o8fad4b54efba017b@mail.gmail.com> update one more time and see how that is On Tue, Jan 6, 2009 at 12:25 PM, Peter P GMX wrote: > One more info: > I have updated to the newest SVN version of FS. > A and B can actually hear each other (just a bit, some scratching) while > the announcement to A is very slow (~50% speed) and very choppy. > > Best regards > peter > > Peter P GMX schrieb: > > I have setup a test machine and a production machine. Since recently the > > production machine behaves differently in terms of uuid bridge. > > > > How it should work (and how it worked before) > > 1) call A comes in > > 2) I play some messages to A > > 3) In the meantime I originate a call to B and transfer to an > > extension, where also some messages are played > > 4) Then I bridge A and B, so they are dropped off the current > > announcements an speak to each other > > 5) when either A or B hangs up, both legs are terminated > > > > New behaviour > > 1) call A comes in > > 2) I play some messages to A > > 3) In the meantime I originate a call to B and transfer to an > > extension, where also some messages are played > > 4) Then I bridge A and B, A and B can hear each other for 1/2 sec, > > then A constinues to hear its messages, B does not hear anything > > 5) when either A or B hangs up, both legs are terminated > > > > 4) is different now! > > > > The FS console show some messages about unbridge > > (SWITCH_MESSAGE_INDICATE_UNBRIDGE), but somehow the calls are still > > connected until A or B hangs up. > > > > Anybody has a clue? > > Best regards > > Peter > > > > 3) is finished, 4) starts > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:35:31 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > > unhandled message 98 (0x62) > > 2009-01-06 17:35:34 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1094 > > switch_ivr_uuid_bridge() (OpenZAP/2:3/49171xxxxxxx) State Change > > CS_EXECUTE -> CS_RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1095 > > switch_ivr_uuid_bridge() (OpenZAP/2:1/216xxxxx) State Change CS_EXECUTE > > -> CS_RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State EXECUTE going to > > sleep > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State Change > > CS_RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET > > 4) Start > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > > uuid_bridge_on_reset() OpenZAP/2:1/216xxxxx CUSTOM RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:53 > > switch_core_standard_on_reset() OpenZAP/2:1/216xxxxx Standard RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET going to > sleep > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State EXECUTE going > > to sleep > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) Running State > > Change CS_RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State RESET > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > > uuid_bridge_on_reset() OpenZAP/2:3/49171xxxxxxx CUSTOM RESET > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:533 > > uuid_bridge_on_reset() (OpenZAP/2:3/49171xxxxxxx) State Change CS_RESET > > -> CS_SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State RESET going > > to sleep > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) Running State > > Change CS_SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 channel_on_soft_execute() > > CHANNEL SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > > uuid_bridge_on_soft_execute() OpenZAP/2:3/49171xxxxxxx CUSTOM TRANSMIT > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:571 > > uuid_bridge_on_soft_execute() (OpenZAP/2:1/216xxxxx) State Change > > CS_RESET -> CS_SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State Change > > CS_SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 channel_on_soft_execute() > > CHANNEL SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > > uuid_bridge_on_soft_execute() OpenZAP/2:1/216xxxxx CUSTOM TRANSMIT > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:181 > > switch_core_standard_on_soft_execute() OpenZAP/2:1/216xxxxx Standard > > SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State SOFT_EXECUTE > > going to sleep > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:814 > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) State Change > > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:856 > > switch_ivr_multi_threaded_bridge() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:863 > > switch_ivr_multi_threaded_bridge() OpenZAP/2:3/49171xxxxxxx receive > > message [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:907 > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) State Change > > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State Change > > CS_EXCHANGE_MEDIA > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:436 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State EXCHANGE_MEDIA > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:511 > > channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > custom/warteschleife_30.wav interrupt_digit 0 ) > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_play_say.c:932 > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > 2009-01-06 17:35:36 [DEBUG] switch_core_io.c:610 > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() > > OpenZAP/2:3/49171xxxxxxx Command Execute stop_dtmf(true) > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:229 > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:231 > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:229 > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:231 > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 31 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7d 02 01 22 30 0d a1 05 30 03 02 01 > > 08 82 01 00 83 01 00] > > > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:35:46 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > > unhandled message 98 (0x62) > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 31 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7e 02 01 22 30 0d a1 05 30 03 02 01 > > 09 82 01 00 83 01 00] > > > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:36:02 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > > unhandled message 98 (0x62) > > 2009-01-06 17:36:06 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:229 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:36:07 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:231 > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:08 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > custom/warteschleife_30.wav interrupt_digit 0 ) > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_play_say.c:932 > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > 2009-01-06 17:36:08 [DEBUG] switch_core_io.c:610 > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 31 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7f 02 01 22 30 0d a1 05 30 03 02 01 > > 0a 82 01 00 83 01 00] > > > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:36:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > > unhandled message 98 (0x62) > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 32 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 80 02 01 22 30 0d a1 05 30 03 02 > > 01 0b 82 01 00 83 01 00] > > > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:36:33 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > > unhandled message 98 (0x62) > > 2009-01-06 17:36:38 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:229 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:36:39 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:231 > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:40 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > > custom/dieser_berater_ist_leider_zur_zeit_nicht_erreichbar_bitte_waehlen_sie_einen_ande.wav > > interrupt_digit 0 ) > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_play_say.c:932 > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > 2009-01-06 17:36:40 [DEBUG] switch_core_io.c:610 > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:45 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:229 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:36:46 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:231 > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:47 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() > > OpenZAP/2:1/216xxxxx Command Execute read(0 4 > > > custom/geben_sie_bitte_nach_dem_signalton_die_kennung_des_gewuenschten_beraters_ein_sch2.wav > > dtmfdtmf 10000 #,*) > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_play_say.c:932 > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > 2009-01-06 17:36:47 [DEBUG] switch_core_io.c:610 > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 32 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 81 02 01 22 30 0d a1 05 30 03 02 > > 01 0c 82 01 00 83 01 00] > > > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:36:48 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > > unhandled message 98 (0x62) > > 2009-01-06 17:37:03 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 32 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 82 02 01 22 30 0d a1 05 30 03 02 > > 01 0d 82 01 00 83 01 00] > > > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:37:04 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > > unhandled message 98 (0x62) > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 13 > > > -------------------------------------------------------------------------------- > > [08 02 00 35 45 08 02 80 90 1e 02 82 88] > > > > 5) Hangup > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > > an event! Type:[45] Size:[115] CRV: 53 (0x35, CTX: Originator) > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 77df70 (2:1) source isdn_data->channels_remote_crv[0x35] > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:455 zap_isdn_931_34() Changing > > state on 2:1 from UP to TERMINATING > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:813 state_advance() 2:1 STATE > > [TERMINATING] > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1121 state_advance() > > Terminating: Direction Inbound > > 2009-01-06 17:37:12 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() > > got clear channel sig [STOP] > > 2009-01-06 17:37:12 [NOTICE] mod_openzap.c:1437 > > on_clear_channel_signal() Hangup OpenZAP/2:1/216xxxxx > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 2009-01-06 17:37:12 [DEBUG] switch_channel.c:1441 > > switch_channel_perform_hangup() Send signal OpenZAP/2:1/216xxxxx [KILL] > > 2009-01-06 17:37:12 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090106/e6e378cb/attachment-0001.html From carole.olivier at enst.fr Wed Jan 7 00:16:34 2009 From: carole.olivier at enst.fr (Carole O.) Date: Wed, 7 Jan 2009 00:16:34 -0800 (PST) Subject: [Freeswitch-users] close channels properly In-Reply-To: References: <21127913.post@talk.nabble.com> <2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org> <21140461.post@talk.nabble.com> <490132B3-A767-4441-BB98-A57F3CBFE51D@freeswitch.org> <97D97C56-AECF-4F43-8432-8DE12A652D0C@freeswitch.org> Message-ID: <21326619.post@talk.nabble.com> Hello, Happy New Year!!!! Yes, I am calling a group of people but those are actually speakers so unfortunately they can not decide to hang up alone at the end of a speech or an announcement and somebody has to kick them out. The only thing the speakers can do is stop transmitting after a long silence but they don't send a "BYE" to freeswitch so the channels between them and freeswitch are not closed (the api command "show channels" keeps listing them). I have seen the application sched_hangup but if I use it in the dialplan it seems to concern just the caller of the extension. Do you have an idea to use it for all the members of a conference? Just as you know, I am aware there is the application api_hangup_hook but I don't want the speakers to get kicked when the caller hangs up. (There are cases where the caller just wants a file to be played through the speakers and does not need to wait for the end before it hangs up). I am going to try to see if there is not a solution on the speakers' side but if someone has a any other ideas that I missed... Thanks, Carole Brian West-3 wrote: > > I think Carole is calling a group of people into a conference.. > leaving and expecting everyone to get kicked. > > /b > > On Dec 23, 2008, at 9:53 AM, Michael S Collins wrote: > >> Carole, >> >> Are you calling the hangup app from the Dialplan? >> >> -MC >> >> Sent from my iPhone > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/close-channels-properly-tp21127913p21326619.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Wed Jan 7 00:29:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 7 Jan 2009 00:29:02 -0800 Subject: [Freeswitch-users] close channels properly In-Reply-To: <21326619.post@talk.nabble.com> References: <21127913.post@talk.nabble.com> <2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org> <21140461.post@talk.nabble.com> <490132B3-A767-4441-BB98-A57F3CBFE51D@freeswitch.org> <97D97C56-AECF-4F43-8432-8DE12A652D0C@freeswitch.org> <21326619.post@talk.nabble.com> Message-ID: <87f2f3b90901070029ta12eb3eid7809db0857e66f0@mail.gmail.com> On Wed, Jan 7, 2009 at 12:16 AM, Carole O. wrote: > > Hello, > > Happy New Year!!!! > > Yes, I am calling a group of people but those are actually speakers so > unfortunately they can not decide to hang up alone at the end of a speech > or > an announcement and somebody has to kick them out. The only thing the > speakers can do is stop transmitting after a long silence but they don't > send a "BYE" to freeswitch so the channels between them and freeswitch are > not closed (the api command "show channels" keeps listing them). So what are the conditions under which you want the channels to be hung up? > > > I have seen the application sched_hangup but if I use it in the dialplan it > seems to concern just the caller of the extension. Do you have an idea to > use it for all the members of a conference? > Just as you know, I am aware there is the application api_hangup_hook but I > don't want the speakers to get kicked when the caller hangs up. (There are > cases where the caller just wants a file to be played through the speakers > and does not need to wait for the end before it hangs up). > > I am going to try to see if there is not a solution on the speakers' side > but if someone has a any other ideas that I missed... > > Thanks, > Carole > > > > Brian West-3 wrote: > > > > I think Carole is calling a group of people into a conference.. > > leaving and expecting everyone to get kicked. > > > > /b > > > > On Dec 23, 2008, at 9:53 AM, Michael S Collins wrote: > > > >> Carole, > >> > >> Are you calling the hangup app from the Dialplan? > >> > >> -MC > >> > >> Sent from my iPhone > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/close-channels-properly-tp21127913p21326619.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/d4228aa2/attachment.html From carole.olivier at enst.fr Wed Jan 7 00:54:53 2009 From: carole.olivier at enst.fr (Carole O.) Date: Wed, 7 Jan 2009 00:54:53 -0800 (PST) Subject: [Freeswitch-users] close channels properly In-Reply-To: <87f2f3b90901070029ta12eb3eid7809db0857e66f0@mail.gmail.com> References: <21127913.post@talk.nabble.com> <2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org> <21140461.post@talk.nabble.com> <490132B3-A767-4441-BB98-A57F3CBFE51D@freeswitch.org> <97D97C56-AECF-4F43-8432-8DE12A652D0C@freeswitch.org> <21326619.post@talk.nabble.com> <87f2f3b90901070029ta12eb3eid7809db0857e66f0@mail.gmail.com> Message-ID: <21327016.post@talk.nabble.com> At the end of the announcement whose I know the length. I would like to force freeswitch to send a BYE to the speakers. Michael Collins-11 wrote: > > On Wed, Jan 7, 2009 at 12:16 AM, Carole O. wrote: > >> >> Hello, >> >> Happy New Year!!!! >> >> Yes, I am calling a group of people but those are actually speakers so >> unfortunately they can not decide to hang up alone at the end of a speech >> or >> an announcement and somebody has to kick them out. The only thing the >> speakers can do is stop transmitting after a long silence but they don't >> send a "BYE" to freeswitch so the channels between them and freeswitch >> are >> not closed (the api command "show channels" keeps listing them). > > > So what are the conditions under which you want the channels to be hung > up? > > >> >> >> I have seen the application sched_hangup but if I use it in the dialplan >> it >> seems to concern just the caller of the extension. Do you have an idea to >> use it for all the members of a conference? >> Just as you know, I am aware there is the application api_hangup_hook but >> I >> don't want the speakers to get kicked when the caller hangs up. (There >> are >> cases where the caller just wants a file to be played through the >> speakers >> and does not need to wait for the end before it hangs up). >> >> I am going to try to see if there is not a solution on the speakers' side >> but if someone has a any other ideas that I missed... >> >> Thanks, >> Carole >> >> >> >> Brian West-3 wrote: >> > >> > I think Carole is calling a group of people into a conference.. >> > leaving and expecting everyone to get kicked. >> > >> > /b >> > >> > On Dec 23, 2008, at 9:53 AM, Michael S Collins wrote: >> > >> >> Carole, >> >> >> >> Are you calling the hangup app from the Dialplan? >> >> >> >> -MC >> >> >> >> Sent from my iPhone >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/close-channels-properly-tp21127913p21326619.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/close-channels-properly-tp21127913p21327016.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kristjan.ugrin at gmail.com Wed Jan 7 01:33:06 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Wed, 07 Jan 2009 10:33:06 +0100 Subject: [Freeswitch-users] firewall and nat Message-ID: Hello! Yesterday I've successfully placed a call between two different domains: originate sofia/default/1003 at 10.99.8.221 &bridge(sofia/gateway/212.235.180.41/1001) I didn't hear any audio, but it was kinda working. Today I investigated this more deep and found some issues. FS with 212.235.180.41 is a public computer with firewall, but open TCP and UDP 5060, 5080 ports. Freeswitch on this machine uses default configuration. FS with 10.99.8.221 is a lan computer in a different place, this is where I would like to start a call, the other way would be probably too much difficult for now. I've added a gateway entry to this one: http://pastebin.com/m2174ead Calling from 10.99.8.221 (for e.g. using softphone at ext. 1003) to 212.235.180.41 (ext. 1001 for e.g.) works. Both end answers, however I cannot hear audio coming trough. When testing I'm at the computer which is behind a lan, so I'm capturing music as audio source on the other side. Are there any other ports I should open on public computer? With wireshark on the computer behind a lan, I can see RTP going away to 212.235.180.41, but not the other way. There are also issues when e.g. terminating a call on public computer, fs on the other end will never terminate the call since SIP messages cannot reach the computer behind lan I guess, but this is second problem. -- kriko From saeedahmad1981 at gmail.com Wed Jan 7 01:48:10 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Wed, 7 Jan 2009 10:48:10 +0100 Subject: [Freeswitch-users] Need some info In-Reply-To: <87f2f3b90901061213s6b1023eeua993428854c3197b@mail.gmail.com> References: <3cbb52a70901060644l683d4ab5hcd5c01d2621c0b7d@mail.gmail.com> <87f2f3b90901061213s6b1023eeua993428854c3197b@mail.gmail.com> Message-ID: Dear All, I am also planning the same setup. Do we have a call center graphical solution which works with FS? Same as VICIDIAL which work with asterisk. Kind Regards Saeed Ahmed Tariq _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, January 06, 2009 9:13 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Need some info Welcome to FreeSWITCH! On Tue, Jan 6, 2009 at 6:44 AM, Aadil M wrote: Hi All, I am a new comer to FreeSwitch. The thing is I am planning to setup a call center along with a few of my friends. Hence I was looking around for some IP PBX and so stumbled upon FreeSwitch. This call center would be a 10 to 20 person setup. Here are a few of the questions that I have? 1. Can I have just on PSTN line and configure multiple extensions on FS by using relevent hardware?If not then how many PSTN lines would I require? If you are using a VoIP provider then you won't need PSTN lines. However, if you do need PSTN then it is a matter of analog or digital. You need to know what your call traffic volume is like before you can decide how many and what types of cards. Do you know what kind of call traffic you will have? 2. I did Google for the hardware, it seems I would require some FXO/FXS cards. Am I right? Where can I get them? Sangoma, Rhino Technologies, pbxhardware.com, etc. - Anything that is "Asterisk" compatible will work with FreeSWITCH. Note: The FreeSWITCH PRI stack is still in development; it works but has a few bugs. I am sorry if you find these question too novice or lame. Kindly forgive me. Looking for a positive reply from the community. You may also want to join us on IRC: www.freenode.net, #freeswitch channel. -MC (mercutioviz) Regards, Aadil _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/3d3fbcf0/attachment-0001.html From jason at jasonjgw.net Wed Jan 7 01:52:17 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 7 Jan 2009 20:52:17 +1100 Subject: [Freeswitch-users] firewall and nat In-Reply-To: References: Message-ID: <20090107095217.GA13365@jdc.jasonjgw.net> kriko wrote: > Are there any other ports I should open on public computer? With wireshark > on the computer behind a lan, I can see RTP going away to 212.235.180.41, > but not the other way. Maybe the NAT device between the two machines is blocking the rtp traffic. Can you configure the NAT device to forward incoming rtp to the correct destination on the LAN? If you capture packets on the machine with the public IP address and it shows the RTP traffic being sent, this is evidence that the NAT device in between is causing your problems Have a look also at the wiki pages related to NAT. > > There are also issues when e.g. terminating a call on public computer, fs on > the other end will never terminate the call since SIP messages cannot reach > the computer behind lan I guess, but this is second problem. This is fixed by having SIP packets (port 6080 in the default external profile) forwarded properly by the NAT device to the machine on the LAN. In my router's configuration: ip nat source static udp 192.168.0.2 5080 interface Dialer1 5080 I don't need to worry about the RTP ports because IP inspection is enabled, and it seems to handle everything. From wasim at convergence.pk Wed Jan 7 02:51:06 2009 From: wasim at convergence.pk (Wasim Baig) Date: Wed, 7 Jan 2009 15:51:06 +0500 Subject: [Freeswitch-users] Need some info In-Reply-To: References: <3cbb52a70901060644l683d4ab5hcd5c01d2621c0b7d@mail.gmail.com> <87f2f3b90901061213s6b1023eeua993428854c3197b@mail.gmail.com> Message-ID: On Wed, Jan 7, 2009 at 2:48 PM, Saeed Ahmed wrote > Do we have a call center graphical solution which works with FS? Same as > VICIDIAL which work with asterisk. > No, but I've got a pot going to bribe outtolunc in porting gnudialer to fs ... anyone else interested in anteing up will help get us a PD for FS quicker -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/a4559a6d/attachment.html From Prometheus001 at gmx.net Wed Jan 7 03:00:28 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 07 Jan 2009 12:00:28 +0100 Subject: [Freeswitch-users] different behaviour on uuid_bridge, doesn't really bridge In-Reply-To: <191c3a030901061609m19d91c15o8fad4b54efba017b@mail.gmail.com> References: <49638DC1.2000307@gmx.net> <4963A228.3000103@gmx.net> <191c3a030901061609m19d91c15o8fad4b54efba017b@mail.gmail.com> Message-ID: <49648B4C.8050505@gmx.net> Hello Anthony, I updated to SNV=11084 and still have the problem. The behaviour is slightly changed now. Step 4+5 (a s below in my mail) 4) When I bridge A and B, A and B canNOT hear each other (e.g. for 1/2 sec). A continues to hear its messages, B does not hear anything 5) When I hangup B then A is still active and does not recognize hangup of B. At former times, when the call was bridged, I had an "unbind" for each call on the event_socket interface, so event_socket was out of the loop after uuid_bridge. Now I have an "unbind" for each party only at the time when the party hangs up. Best regards Peter Anthony Minessale schrieb: > update one more time and see how that is > > > On Tue, Jan 6, 2009 at 12:25 PM, Peter P GMX > wrote: > > One more info: > I have updated to the newest SVN version of FS. > A and B can actually hear each other (just a bit, some scratching) > while > the announcement to A is very slow (~50% speed) and very choppy. > > Best regards > peter > > Peter P GMX schrieb: > > I have setup a test machine and a production machine. Since > recently the > > production machine behaves differently in terms of uuid bridge. > > > > How it should work (and how it worked before) > > 1) call A comes in > > 2) I play some messages to A > > 3) In the meantime I originate a call to B and transfer to an > > extension, where also some messages are played > > 4) Then I bridge A and B, so they are dropped off the current > > announcements an speak to each other > > 5) when either A or B hangs up, both legs are terminated > > > > New behaviour > > 1) call A comes in > > 2) I play some messages to A > > 3) In the meantime I originate a call to B and transfer to an > > extension, where also some messages are played > > 4) Then I bridge A and B, A and B can hear each other for 1/2 sec, > > then A constinues to hear its messages, B does not hear anything > > 5) when either A or B hangs up, both legs are terminated > > > > 4) is different now! > > > > The FS console show some messages about unbridge > > (SWITCH_MESSAGE_INDICATE_UNBRIDGE), but somehow the calls are still > > connected until A or B hangs up. > > > > Anybody has a clue? > > Best regards > > Peter > > > > 3) is finished, 4) starts > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:35:31 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > Received > > unhandled message 98 (0x62) > > 2009-01-06 17:35:34 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1094 > > switch_ivr_uuid_bridge() (OpenZAP/2:3/49171xxxxxxx) State Change > > CS_EXECUTE -> CS_RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1095 > > switch_ivr_uuid_bridge() (OpenZAP/2:1/216xxxxx) State Change > CS_EXECUTE > > -> CS_RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State EXECUTE > going to > > sleep > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State > Change > > CS_RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET > > 4) Start > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > > uuid_bridge_on_reset() OpenZAP/2:1/216xxxxx CUSTOM RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:53 > > switch_core_standard_on_reset() OpenZAP/2:1/216xxxxx Standard RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET > going to sleep > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > EXECUTE going > > to sleep > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) Running State > > Change CS_RESET > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State RESET > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > > uuid_bridge_on_reset() OpenZAP/2:3/49171xxxxxxx CUSTOM RESET > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:533 > > uuid_bridge_on_reset() (OpenZAP/2:3/49171xxxxxxx) State Change > CS_RESET > > -> CS_SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State RESET > going > > to sleep > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) Running State > > Change CS_SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > channel_on_soft_execute() > > CHANNEL SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > > uuid_bridge_on_soft_execute() OpenZAP/2:3/49171xxxxxxx CUSTOM > TRANSMIT > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:571 > > uuid_bridge_on_soft_execute() (OpenZAP/2:1/216xxxxx) State Change > > CS_RESET -> CS_SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State > Change > > CS_SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > channel_on_soft_execute() > > CHANNEL SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > > uuid_bridge_on_soft_execute() OpenZAP/2:1/216xxxxx CUSTOM TRANSMIT > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:181 > > switch_core_standard_on_soft_execute() OpenZAP/2:1/216xxxxx Standard > > SOFT_EXECUTE > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State SOFT_EXECUTE > > going to sleep > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:814 > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) State > Change > > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:856 > > switch_ivr_multi_threaded_bridge() OpenZAP/2:1/216xxxxx receive > message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:863 > > switch_ivr_multi_threaded_bridge() OpenZAP/2:3/49171xxxxxxx receive > > message [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:907 > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) State > Change > > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State > Change > > CS_EXCHANGE_MEDIA > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:436 > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > EXCHANGE_MEDIA > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:511 > > channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > switch_ivr_parse_event() > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > custom/warteschleife_30.wav interrupt_digit 0 ) > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_play_say.c:932 > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > 2009-01-06 17:35:36 [DEBUG] switch_core_io.c:610 > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > message > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > switch_ivr_parse_event() > > OpenZAP/2:3/49171xxxxxxx Command Execute stop_dtmf(true) > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:229 > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:231 > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:229 > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:231 > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > READ 31 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7d 02 01 22 30 0d a1 05 30 > 03 02 01 > > 08 82 01 00 83 01 00] > > > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:35:46 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > Received > > unhandled message 98 (0x62) > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > READ 31 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7e 02 01 22 30 0d a1 05 30 > 03 02 01 > > 09 82 01 00 83 01 00] > > > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:36:02 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > Received > > unhandled message 98 (0x62) > > 2009-01-06 17:36:06 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:229 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:36:07 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:231 > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:08 [DEBUG] switch_ivr.c:391 > switch_ivr_parse_event() > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > custom/warteschleife_30.wav interrupt_digit 0 ) > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_play_say.c:932 > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > 2009-01-06 17:36:08 [DEBUG] switch_core_io.c:610 > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > message > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > READ 31 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7f 02 01 22 30 0d a1 05 30 > 03 02 01 > > 0a 82 01 00 83 01 00] > > > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:36:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > Received > > unhandled message 98 (0x62) > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > READ 32 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 80 02 01 22 30 0d a1 05 > 30 03 02 > > 01 0b 82 01 00 83 01 00] > > > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:36:33 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > Received > > unhandled message 98 (0x62) > > 2009-01-06 17:36:38 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:229 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:36:39 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:231 > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:40 [DEBUG] switch_ivr.c:391 > switch_ivr_parse_event() > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > > custom/dieser_berater_ist_leider_zur_zeit_nicht_erreichbar_bitte_waehlen_sie_einen_ande.wav > > interrupt_digit 0 ) > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_play_say.c:932 > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > 2009-01-06 17:36:40 [DEBUG] switch_core_io.c:610 > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > message > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:45 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:229 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > 2009-01-06 17:36:46 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:231 > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:667 > > switch_core_session_queue_private_event() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_bridge.c:226 > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:47 [DEBUG] switch_ivr.c:391 > switch_ivr_parse_event() > > OpenZAP/2:1/216xxxxx Command Execute read(0 4 > > > custom/geben_sie_bitte_nach_dem_signalton_die_kennung_des_gewuenschten_beraters_ein_sch2.wav > > dtmfdtmf 10000 #,*) > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_play_say.c:932 > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > 2009-01-06 17:36:47 [DEBUG] switch_core_io.c:610 > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > message > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > > switch_core_session_perform_receive_message() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > READ 32 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 81 02 01 22 30 0d a1 05 > 30 03 02 > > 01 0c 82 01 00 83 01 00] > > > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:36:48 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > Received > > unhandled message 98 (0x62) > > 2009-01-06 17:37:03 [DEBUG] switch_ivr_play_say.c:1222 > > switch_ivr_play_file() done playing file > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > READ 32 > > > -------------------------------------------------------------------------------- > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 82 02 01 22 30 0d a1 05 > 30 03 02 > > 01 0d 82 01 00 83 01 00] > > > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > Yay I got > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > 2009-01-06 17:37:04 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > Received > > unhandled message 98 (0x62) > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > READ 13 > > > -------------------------------------------------------------------------------- > > [08 02 00 35 45 08 02 80 90 1e 02 82 88] > > > > 5) Hangup > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > Yay I got > > an event! Type:[45] Size:[115] CRV: 53 (0x35, CTX: Originator) > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > > 77df70 (2:1) source isdn_data->channels_remote_crv[0x35] > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:455 zap_isdn_931_34() > Changing > > state on 2:1 from UP to TERMINATING > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:813 state_advance() 2:1 > STATE > > [TERMINATING] > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1121 state_advance() > > Terminating: Direction Inbound > > 2009-01-06 17:37:12 [DEBUG] mod_openzap.c:1418 > on_clear_channel_signal() > > got clear channel sig [STOP] > > 2009-01-06 17:37:12 [NOTICE] mod_openzap.c:1437 > > on_clear_channel_signal() Hangup OpenZAP/2:1/216xxxxx > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 2009-01-06 17:37:12 [DEBUG] switch_channel.c:1441 > > switch_channel_perform_hangup() Send signal OpenZAP/2:1/216xxxxx > [KILL] > > 2009-01-06 17:37:12 [DEBUG] switch_core_session.c:779 > > switch_core_session_signal_state_change() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jonas.gauffin at gmail.com Wed Jan 7 03:18:30 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 7 Jan 2009 12:18:30 +0100 Subject: [Freeswitch-users] Building FS on win Message-ID: I'm trying to build latest trunk on win (vs2008) and get the following errors: 12>mod_spidermonkey.obj : error LNK2001: unresolved external symbol _switch_dso_open 12>mod_spidermonkey.obj : error LNK2001: unresolved external symbol _switch_dso_data_sym any thoughts? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/8913b13e/attachment.html From Prometheus001 at gmx.net Wed Jan 7 03:39:45 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 07 Jan 2009 12:39:45 +0100 Subject: [Freeswitch-users] firewall and nat In-Reply-To: References: Message-ID: <49649481.7000804@gmx.net> Generally speaking you will need to open an UPD port range for the RTP stream. This can be configured on FS. Eg. we use 12000-13000 on our system. Then If you do not hear any sound you may put in your external and internal profile, if FS is natted. Best regards Peter kriko schrieb: > Hello! > > Yesterday I've successfully placed a call between two different domains: > originate sofia/default/1003 at 10.99.8.221 &bridge(sofia/gateway/212.235.180.41/1001) > > I didn't hear any audio, but it was kinda working. Today I investigated this more deep and found some issues. > FS with 212.235.180.41 is a public computer with firewall, but open TCP and UDP 5060, 5080 ports. Freeswitch on this machine > uses default configuration. > > FS with 10.99.8.221 is a lan computer in a different place, this is where I would like to start a call, the other way would > be probably too much difficult for now. I've added a gateway entry to this one: > http://pastebin.com/m2174ead > > Calling from 10.99.8.221 (for e.g. using softphone at ext. 1003) to 212.235.180.41 (ext. 1001 for e.g.) works. Both end > answers, however I cannot hear audio coming trough. When testing I'm at the computer which is behind a lan, so I'm > capturing music as audio source on the other side. > > Are there any other ports I should open on public computer? > With wireshark on the computer behind a lan, I can see RTP going away to 212.235.180.41, but not the other way. > > There are also issues when e.g. terminating a call on public computer, fs on the other end will never terminate the call since > SIP messages cannot reach the computer behind lan I guess, but this is second problem. > > > From fidibus83 at aol.com Wed Jan 7 05:08:05 2009 From: fidibus83 at aol.com (fidibus83) Date: Wed, 7 Jan 2009 14:08:05 +0100 Subject: [Freeswitch-users] Problems with ODBC In-Reply-To: <49639112.7080902@freeswitch.org> References: <007801c97006$fb3e7960$6445310a@Franzi> <496374F1.5040503@freeswitch.org><00b301c97013$b4169c40$6445310a@Franzi> <49639112.7080902@freeswitch.org> Message-ID: <00cf01c970c8$fb233b10$6445310a@Franzi> Thanks for your help. It works. _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Raymond Chandler Gesendet: Dienstag, 6. Januar 2009 18:13 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] Problems with ODBC is mysql running on the localhost? if so, is that the correct path to the .sock? you might also try using 127.0.0.1 as the host instead of localhost to force a network connection. -Ray fidibus83 wrote: The libmyodbc package is installed. But I see I have a lot of version of libmyodbc: libmyodbc3.so libmyodbc3_r.so libmyodbc3_r-3.51.14.so libmyodbc3-3.51.14.so I changed odbcinst.ini: [MySQL] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc3.so Setup = /usr/lib/libodbcmyS.so I run isgl ?vvv and I get this: [08S01] [unixODBC] [MySQL] [ODBC 3.51 Driver] Can?t connect to local MySQL server through socket ?/var/run/mysqld/mysqld.sock? (2) [ISQL] ERROR: Could not SQLConnect _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Raymond Chandler Gesendet: Dienstag, 6. Januar 2009 16:13 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] Problems with ODBC fidibus83 wrote: Hello, I need your help again! I want to use ODBC in javascripts. First I installed unixodbc and I did all the steps what are mentioned in wiki: mod spidermonkey odbc. But when I test my ODBC Setup by running isql I get this error: [ISQL]ERROR: Could not SQLConnect This isn't really a general purpose ODBC list, but using the -v option of isql tends to yield more verbose output (imagine that!) so you get: isql -vvv freeswitch freeswitch freeswitch [IM002][unixODBC][Driver Manager]Data source name not found, and no default driver specified [ISQL]ERROR: Could not SQLConnect as opposed to: isql freeswitch freeswitch freeswitch [ISQL]ERROR: Could not SQLConnect -Ray _____ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/8945361b/attachment.html From jocke29 at gmail.com Wed Jan 7 05:09:25 2009 From: jocke29 at gmail.com (jocke eriksson) Date: Wed, 7 Jan 2009 14:09:25 +0100 Subject: [Freeswitch-users] bootstrap error Message-ID: Happy New Year to you all ! I'm doing an install and ran into some trouble, any ide why ? bootstrap: libtool version 2.2.4 found. You need libtool version 1.5.14 or newer installed to build FreeSWITCH from SVN. Regards Jocke -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/e37f141f/attachment.html From fidibus83 at aol.com Wed Jan 7 05:14:28 2009 From: fidibus83 at aol.com (fidibus83) Date: Wed, 7 Jan 2009 14:14:28 +0100 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so Message-ID: <00ef01c970c9$dfbd91d0$6445310a@Franzi> Hello, I have problems with mod_spidermonkey_odbc. Mod_spidermonkey_odbc.so does not exist. Following the instructions, I have created the ODBC environment and compiled it after running ./configure --without-libcurl --enable-core-odbc-support make make install I have to configure FS without libcurl because I use Fedora Core 8 and I want to use mod_xml_cdr. I uncomment: conf/autoload_configs/spidermonkey.conf.xml build/modules.conf languages/mod_spidermonkey_odbc Any idea of what's wrong? Thanks, fidibus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/425295db/attachment-0001.html From stkn at freeswitch.org Wed Jan 7 05:16:09 2009 From: stkn at freeswitch.org (Stefan Knoblich) Date: Wed, 7 Jan 2009 14:16:09 +0100 Subject: [Freeswitch-users] bootstrap error In-Reply-To: References: Message-ID: <200901071416.09441.stkn@freeswitch.org> Am Wednesday 07 January 2009 schrieb jocke eriksson: > Happy New Year to you all ! > > I'm doing an install and ran into some trouble, any ide why ? > > bootstrap: libtool version 2.2.4 found. > You need libtool version 1.5.14 or newer installed to build > FreeSWITCH from SVN. > > Regards Jocke > See http://jira.freeswitch.org/browse/FSBUILD-82 -- Stefan Knoblich Systemadministrator axsentis GmbH Eupener Strasse 74 50933 K?ln Tel: 0180 - 506 705 521* Fax: 0180 - 506 705 529* E-Mail: s.knoblich at axsentis.de Web: www.axsentis.de Eingetragen beim AG K?ln: HR B 56238 UST-ID: DE244977565 Gesellschafter-Gesch?ftsf?hrer: Yan Lecomte, Eduard Schlein, Apostolos Varsamis *14ct/min aus dem Festnetz der T-Com | dtms From kristjan.ugrin at gmail.com Wed Jan 7 05:51:41 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Wed, 07 Jan 2009 14:51:41 +0100 Subject: [Freeswitch-users] [ringback] problems with dingaling In-Reply-To: References: Message-ID: originate {transfer_ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003 at 10.99.8.221 &bridge(sofia/default/1000 at 10.99.8.221) This works, but still when calling gtalk user: originate {transfer_ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}user/1003 at 10.99.8.221 &bridge(dingaling/gmail.com/atomic.devterium at gmail.com) there is only dead silence. I would be glad even with the default ringing tone. Log: http://pastebin.com/m7a0d6a9e Maybe it is because of dingaling? When originating a call between softphones, ringing tone or ringback is working. On Tue, 06 Jan 2009 15:48:52 +0100, Brian West wrote: > > On Jan 6, 2009, at 4:30 AM, kriko wrote: > >> I'm trying to entertain caller with some music or audio message so >> he knows that call is in progress. >> I tried to setup ringback like this: >> "api originate >> {ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003 at 10.99.8.221 >> &bridge(sofia/default/1000 at 10.99.8.221)"' > > In this case the first leg is already answered you need to use > transfer_ringback. > > -- kriko From matthew at matthew.at Tue Jan 6 22:22:26 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Tue, 06 Jan 2009 22:22:26 -0800 Subject: [Freeswitch-users] polycom one-way audio problem Message-ID: <49644A22.8050802@matthew.at> I have two Polycom phones (one 550 and one 650) successfully registered to the switch. If I call from either extension to the other and answer, audio flows from the calling party to the called party, but audio does not flow from the called party back to the calling party. Even more strange, the called party does not answer, then when the call is sent to voicemail, the calling party *also* does not hear any of the voicemail greeting, though they are recorded successfully. Calling the voicemail box directly from either phone *does* work, and the calling party can hear the prompts just fine in that case. There is no NAT between the phones and FreeSWITCH. Suggestions? Matthew Kaufman matthew at matthew.at From mike at jerris.com Wed Jan 7 06:15:01 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 7 Jan 2009 09:15:01 -0500 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so In-Reply-To: <00ef01c970c9$dfbd91d0$6445310a@Franzi> References: <00ef01c970c9$dfbd91d0$6445310a@Franzi> Message-ID: <91324EF2-354D-4ECA-95AF-9905DD5EB1DC@jerris.com> you would need to modify the modules.conf in the root of the source dir, not in the build dir. On Jan 7, 2009, at 8:14 AM, fidibus83 wrote: > Hello, > > I have problems with mod_spidermonkey_odbc. Mod_spidermonkey_odbc.so > does not exist. > > Following the instructions, I have created the ODBC environment and > compiled it after running > ./configure --without-libcurl --enable-core-odbc-support > make > make install > > I have to configure FS without libcurl because I use Fedora Core 8 > and I want to use mod_xml_cdr. > > I uncomment: > > conf/autoload_configs/spidermonkey.conf.xml > > > build/modules.conf > languages/mod_spidermonkey_odbc > > Any idea of what's wrong? > > Thanks, fidibus > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/31c2ca72/attachment.html From mike at jerris.com Wed Jan 7 06:26:06 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 7 Jan 2009 09:26:06 -0500 Subject: [Freeswitch-users] Building FS on win In-Reply-To: References: Message-ID: <1675B229-6F41-4F98-B4D2-48EDCDD3F037@jerris.com> New Revision: 11085 Log: fix windows build breakage from svn rev 11084 On Jan 7, 2009, at 6:18 AM, Jonas Gauffin wrote: > I'm trying to build latest trunk on win (vs2008) and get the > following errors: > > 12>mod_spidermonkey.obj : error LNK2001: unresolved external symbol > _switch_dso_open > 12>mod_spidermonkey.obj : error LNK2001: unresolved external symbol > _switch_dso_data_sym > > any thoughts? > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fidibus83 at aol.com Wed Jan 7 06:53:54 2009 From: fidibus83 at aol.com (fidibus83) Date: Wed, 7 Jan 2009 15:53:54 +0100 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so In-Reply-To: <91324EF2-354D-4ECA-95AF-9905DD5EB1DC@jerris.com> References: <00ef01c970c9$dfbd91d0$6445310a@Franzi> <91324EF2-354D-4ECA-95AF-9905DD5EB1DC@jerris.com> Message-ID: <011801c970d7$c33b6920$6445310a@Franzi> Oh, I?m sorry. Should I comment mod_spidermonkey_odbc in root/freeswitch/build/modules.conf again? _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 7. Januar 2009 15:15 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so you would need to modify the modules.conf in the root of the source dir, not in the build dir. On Jan 7, 2009, at 8:14 AM, fidibus83 wrote: Hello, I have problems with mod_spidermonkey_odbc. Mod_spidermonkey_odbc.so does not exist. Following the instructions, I have created the ODBC environment and compiled it after running ./configure --without-libcurl --enable-core-odbc-support make make install I have to configure FS without libcurl because I use Fedora Core 8 and I want to use mod_xml_cdr. I uncomment: conf/autoload_configs/spidermonkey.conf.xml build/modules.conf languages/mod_spidermonkey_odbc Any idea of what's wrong? Thanks, fidibus _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org = -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/47e94dd8/attachment-0001.html From chris at maxpowersoft.com Wed Jan 7 07:16:50 2009 From: chris at maxpowersoft.com (Chris Danielson) Date: Wed, 07 Jan 2009 07:16:50 -0800 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so In-Reply-To: <011801c970d7$c33b6920$6445310a@Franzi> References: <00ef01c970c9$dfbd91d0$6445310a@Franzi> <91324EF2-354D-4ECA-95AF-9905DD5EB1DC@jerris.com> <011801c970d7$c33b6920$6445310a@Franzi> Message-ID: <4964C762.5060807@maxpowersoft.com> Fidibus, Make sure that you also have installed unixodbc. As shown here: http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#unixodbc Kind Regards, Chris fidibus83 wrote: > > Oh, I'm sorry. > > > > Should I comment mod_spidermonkey_odbc in > root/freeswitch/build/modules.conf again? > > > > ------------------------------------------------------------------------ > > *Von:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Michael Jerris > *Gesendet:* Mittwoch, 7. Januar 2009 15:15 > *An:* freeswitch-users at lists.freeswitch.org > *Betreff:* Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so > > > > you would need to modify the modules.conf in the root of the source > dir, not in the build dir. > > > > > > On Jan 7, 2009, at 8:14 AM, fidibus83 wrote: > > > > Hello, > > > > I have problems with mod_spidermonkey_odbc. Mod_spidermonkey_odbc.so does not exist. > > Following the instructions, I have created the ODBC environment and compiled it after running > ./configure --without-libcurl --enable-core-odbc-support > make > make install > > I have to configure FS without libcurl because I use Fedora Core 8 and I want to use mod_xml_cdr. > > > I uncomment: > > > > /conf/autoload_configs/spidermonkey.conf.xml/ > > > > |/build/modules.conf/| > > |languages/mod_spidermonkey_odbc| > > | | > > Any idea of what's wrong? > > > > Thanks, fidibus > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/48161594/attachment.html From fidibus83 at aol.com Wed Jan 7 07:34:30 2009 From: fidibus83 at aol.com (fidibus83) Date: Wed, 7 Jan 2009 16:34:30 +0100 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so In-Reply-To: <4964C762.5060807@maxpowersoft.com> References: <00ef01c970c9$dfbd91d0$6445310a@Franzi> <91324EF2-354D-4ECA-95AF-9905DD5EB1DC@jerris.com><011801c970d7$c33b6920$6445310a@Franzi> <4964C762.5060807@maxpowersoft.com> Message-ID: <012901c970dd$6f8bb540$6445310a@Franzi> Sorry Chris, my mistake. Mod_spidermonkey_odbc.so does exist now. I wanted to said that I?m sorry for my stupid mistake that I uncomment the wrong thing. But thank you very much for your help! _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris Danielson Gesendet: Mittwoch, 7. Januar 2009 16:17 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Fidibus, Make sure that you also have installed unixodbc. As shown here: http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#unixodbc Kind Regards, Chris fidibus83 wrote: Oh, I?m sorry. Should I comment mod_spidermonkey_odbc in root/freeswitch/build/modules.conf again? _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 7. Januar 2009 15:15 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so you would need to modify the modules.conf in the root of the source dir, not in the build dir. On Jan 7, 2009, at 8:14 AM, fidibus83 wrote: Hello, I have problems with mod_spidermonkey_odbc. Mod_spidermonkey_odbc.so does not exist. Following the instructions, I have created the ODBC environment and compiled it after running ./configure --without-libcurl --enable-core-odbc-support make make install I have to configure FS without libcurl because I use Fedora Core 8 and I want to use mod_xml_cdr. I uncomment: conf/autoload_configs/spidermonkey.conf.xml build/modules.conf languages/mod_spidermonkey_odbc Any idea of what's wrong? Thanks, fidibus _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/2ae862db/attachment-0001.html From chris at maxpowersoft.com Wed Jan 7 07:39:06 2009 From: chris at maxpowersoft.com (Chris Danielson) Date: Wed, 07 Jan 2009 07:39:06 -0800 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so In-Reply-To: <012901c970dd$6f8bb540$6445310a@Franzi> References: <00ef01c970c9$dfbd91d0$6445310a@Franzi> <91324EF2-354D-4ECA-95AF-9905DD5EB1DC@jerris.com><011801c970d7$c33b6920$6445310a@Franzi> <4964C762.5060807@maxpowersoft.com> <012901c970dd$6f8bb540$6445310a@Franzi> Message-ID: <4964CC9A.2090800@maxpowersoft.com> Anytime. Good luck. fidibus83 wrote: > > Sorry Chris, my mistake. Mod_spidermonkey_odbc.so does exist now. > > > > I wanted to said that I'm sorry for my stupid mistake that I uncomment > the wrong thing. > > > > But thank you very much for your help! > > > > ------------------------------------------------------------------------ > > *Von:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Chris Danielson > *Gesendet:* Mittwoch, 7. Januar 2009 16:17 > *An:* freeswitch-users at lists.freeswitch.org > *Betreff:* Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so > > > > Fidibus, > Make sure that you also have installed unixodbc. As shown here: > http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#unixodbc > > Kind Regards, > Chris > > fidibus83 wrote: > > Oh, I'm sorry. > > > > Should I comment mod_spidermonkey_odbc in > root/freeswitch/build/modules.conf again? > > > > ------------------------------------------------------------------------ > > *Von:* freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Michael Jerris > *Gesendet:* Mittwoch, 7. Januar 2009 15:15 > *An:* freeswitch-users at lists.freeswitch.org > > *Betreff:* Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so > > > > you would need to modify the modules.conf in the root of the source > dir, not in the build dir. > > > > > > On Jan 7, 2009, at 8:14 AM, fidibus83 wrote: > > > > > Hello, > > > > I have problems with mod_spidermonkey_odbc. Mod_spidermonkey_odbc.so does not exist. > > Following the instructions, I have created the ODBC environment and compiled it after running > ./configure --without-libcurl --enable-core-odbc-support > make > make install > > I have to configure FS without libcurl because I use Fedora Core 8 and I want to use mod_xml_cdr. > > > I uncomment: > > > > /conf/autoload_configs/spidermonkey.conf.xml/ > > > > |/build/modules.conf/| > > |languages/mod_spidermonkey_odbc| > > | | > > Any idea of what's wrong? > > > > Thanks, fidibus > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/71213749/attachment.html From msc at freeswitch.org Wed Jan 7 08:14:25 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 7 Jan 2009 08:14:25 -0800 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <49644A22.8050802@matthew.at> References: <49644A22.8050802@matthew.at> Message-ID: What FS revision and is this a default Dialplan? Please pastebin the output of the CLI while making test calls. Be sure to press F8 to enable debug messages. Also, if you can do so turn on SIP messages by launching FreeSWITCH with TPORT_LOG=1. -MC Sent from my iPhone On Jan 6, 2009, at 10:22 PM, Matthew Kaufman wrote: > I have two Polycom phones (one 550 and one 650) successfully > registered > to the switch. If I call from either extension to the other and > answer, > audio flows from the calling party to the called party, but audio does > not flow from the called party back to the calling party. Even more > strange, the called party does not answer, then when the call is > sent to > voicemail, the calling party *also* does not hear any of the voicemail > greeting, though they are recorded successfully. Calling the voicemail > box directly from either phone *does* work, and the calling party can > hear the prompts just fine in that case. > > There is no NAT between the phones and FreeSWITCH. > > Suggestions? > > Matthew Kaufman > matthew at matthew.at > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fidibus83 at aol.com Wed Jan 7 08:31:54 2009 From: fidibus83 at aol.com (fidibus83) Date: Wed, 7 Jan 2009 17:31:54 +0100 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so In-Reply-To: <4964CC9A.2090800@maxpowersoft.com> References: <00ef01c970c9$dfbd91d0$6445310a@Franzi> <91324EF2-354D-4ECA-95AF-9905DD5EB1DC@jerris.com><011801c970d7$c33b6920$6445310a@Franzi> <4964C762.5060807@maxpowersoft.com><012901c970dd$6f8bb540$6445310a@Franzi> <4964CC9A.2090800@maxpowersoft.com> Message-ID: <014301c970e5$740cd6f0$6445310a@Franzi> I have another problem! I want to connect to a mysql database through javascript. I get this error: [ERR] switch_odbc.c:160 switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC] [Driver Manager]Data source name not found, and no default driver specified [ERR] mod_spidermonkey_odbc.c:233 odbc_exec() Database is not connected! [ERR] mod_spidermonkey_odbc.c:289 odbc_next_row() database is not connected! But when I test by running the utility isql the database is connected! What?s going wrong? _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris Danielson Gesendet: Mittwoch, 7. Januar 2009 16:39 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Anytime. Good luck. fidibus83 wrote: Sorry Chris, my mistake. Mod_spidermonkey_odbc.so does exist now. I wanted to said that I?m sorry for my stupid mistake that I uncomment the wrong thing. But thank you very much for your help! _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris Danielson Gesendet: Mittwoch, 7. Januar 2009 16:17 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Fidibus, Make sure that you also have installed unixodbc. As shown here: http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#unixodbc Kind Regards, Chris fidibus83 wrote: Oh, I?m sorry. Should I comment mod_spidermonkey_odbc in root/freeswitch/build/modules.conf again? _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 7. Januar 2009 15:15 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so you would need to modify the modules.conf in the root of the source dir, not in the build dir. On Jan 7, 2009, at 8:14 AM, fidibus83 wrote: Hello, I have problems with mod_spidermonkey_odbc. Mod_spidermonkey_odbc.so does not exist. Following the instructions, I have created the ODBC environment and compiled it after running ./configure --without-libcurl --enable-core-odbc-support make make install I have to configure FS without libcurl because I use Fedora Core 8 and I want to use mod_xml_cdr. I uncomment: conf/autoload_configs/spidermonkey.conf.xml build/modules.conf languages/mod_spidermonkey_odbc Any idea of what's wrong? Thanks, fidibus _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/16af8a74/attachment-0001.html From krice at suspicious.org Wed Jan 7 08:41:48 2009 From: krice at suspicious.org (Ken Rice) Date: Wed, 07 Jan 2009 10:41:48 -0600 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so In-Reply-To: <014301c970e5$740cd6f0$6445310a@Franzi> Message-ID: Well your error tells you explicity DSN not found... From: fidibus83 Reply-To: Date: Wed, 7 Jan 2009 17:31:54 +0100 To: Subject: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so I have another problem! I want to connect to a mysql database through javascript. I get this error: [ERR] switch_odbc.c:160 switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC] [Driver Manager]Data source name not found, and no default driver specified [ERR] mod_spidermonkey_odbc.c:233 odbc_exec() Database is not connected! [ERR] mod_spidermonkey_odbc.c:289 odbc_next_row() database is not connected! But when I test by running the utility isql the database is connected! What?s going wrong? Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris Danielson Gesendet: Mittwoch, 7. Januar 2009 16:39 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Anytime. Good luck. fidibus83 wrote: Sorry Chris, my mistake. Mod_spidermonkey_odbc.so does exist now. I wanted to said that I?m sorry for my stupid mistake that I uncomment the wrong thing. But thank you very much for your help! Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris Danielson Gesendet: Mittwoch, 7. Januar 2009 16:17 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Fidibus, Make sure that you also have installed unixodbc. As shown here: http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#unixodbc Kind Regards, Chris fidibus83 wrote: Oh, I?m sorry. Should I comment mod_spidermonkey_odbc in root/freeswitch/build/modules.conf again? Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 7. Januar 2009 15:15 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so you would need to modify the modules.conf in the root of the source dir, not in the build dir. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/ce02792c/attachment.html From fidibus83 at aol.com Wed Jan 7 08:53:42 2009 From: fidibus83 at aol.com (fidibus83) Date: Wed, 7 Jan 2009 17:53:42 +0100 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so In-Reply-To: References: <014301c970e5$740cd6f0$6445310a@Franzi> Message-ID: <015a01c970e8$7fd66390$6445310a@Franzi> Yes, but why isn?t it found? My odbc.ini configuration: [fs01_odbc] Driver=MySQL SERVER=localhost PORT=3306 DATABASE=freeswitch Socket = /var/lib/mysql/mysql.sock Javascript: use("ODBC"); var DSN="fs01_odbc"; var DB_USER="XXXXX"; var DB_PASS="YYYYY"; var db = new ODBC(DSN, DB_USER, DB_PASS); db.connect(); .. _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Ken Rice Gesendet: Mittwoch, 7. Januar 2009 17:42 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Well your error tells you explicity DSN not found... _____ From: fidibus83 Reply-To: Date: Wed, 7 Jan 2009 17:31:54 +0100 To: Subject: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so I have another problem! I want to connect to a mysql database through javascript. I get this error: [ERR] switch_odbc.c:160 switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC] [Driver Manager]Data source name not found, and no default driver specified [ERR] mod_spidermonkey_odbc.c:233 odbc_exec() Database is not connected! [ERR] mod_spidermonkey_odbc.c:289 odbc_next_row() database is not connected! But when I test by running the utility isql the database is connected! What?s going wrong? _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris Danielson Gesendet: Mittwoch, 7. Januar 2009 16:39 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Anytime. Good luck. fidibus83 wrote: Sorry Chris, my mistake. Mod_spidermonkey_odbc.so does exist now. I wanted to said that I?m sorry for my stupid mistake that I uncomment the wrong thing. But thank you very much for your help! _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris Danielson Gesendet: Mittwoch, 7. Januar 2009 16:17 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Fidibus, Make sure that you also have installed unixodbc. As shown here: http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#unixodbc Kind Regards, Chris fidibus83 wrote: Oh, I?m sorry. Should I comment mod_spidermonkey_odbc in root/freeswitch/build/modules.conf again? _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 7. Januar 2009 15:15 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so you would need to modify the modules.conf in the root of the source dir, not in the build dir. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/31921234/attachment.html From stevecrozz at gmail.com Wed Jan 7 09:38:29 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 7 Jan 2009 09:38:29 -0800 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so In-Reply-To: <015a01c970e8$7fd66390$6445310a@Franzi> References: <014301c970e5$740cd6f0$6445310a@Franzi> <015a01c970e8$7fd66390$6445310a@Franzi> Message-ID: <11990ade0901070938y14f15ba7le95e1c6edfaad70f@mail.gmail.com> I'm assuming you have your driver set up properly since you left that part out, but I got mine working a few weeks ago and I'll show you what I've got if it helps: odbc.ini: [freeswitch_ph] Driver = /usr/lib/odbc/libmyodbc.so Description = MyODBC 3.51 Driver DSN SERVER = localhost PORT = USER = freeswitch_ph Password = Database = freeswitch_ph OPTION = 3 SOCKET = somescript.js use("ODBC"); var DSN = 'freeswitch_ph'; var DB_USER = 'freeswitch_ph'; var DB_PASS = ''; var db = new ODBC(DSN, DB_USER, DB_PASS); db.connect(); On Wed, Jan 7, 2009 at 8:53 AM, fidibus83 wrote: > Yes, but why isn't it found? > > > > My odbc.ini configuration: > > > > [fs01_odbc] > > Driver=MySQL > > SERVER=localhost > > PORT=3306 > > DATABASE=freeswitch > > Socket = /var/lib/mysql/mysql.sock > > > > > > > > Javascript: > > > > ? > > use("ODBC"); > > var DSN="fs01_odbc"; > > var DB_USER="XXXXX"; > > var DB_PASS="YYYYY"; > > var db = new ODBC(DSN, DB_USER, DB_PASS); > > db.connect(); > > ?.. > > > > ________________________________ > > Von: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Ken > Rice > Gesendet: Mittwoch, 7. Januar 2009 17:42 > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so > > > > Well your error tells you explicity DSN not found... > > ________________________________ > > From: fidibus83 > Reply-To: > Date: Wed, 7 Jan 2009 17:31:54 +0100 > To: > Subject: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so > > I have another problem! > > I want to connect to a mysql database through javascript. > > I get this error: > [ERR] switch_odbc.c:160 switch_odbc_handle_connect() STATE: IM002 CODE 0 > ERROR: [unixODBC] [Driver Manager]Data source name not found, and no default > driver specified > [ERR] mod_spidermonkey_odbc.c:233 odbc_exec() Database is not connected! > [ERR] mod_spidermonkey_odbc.c:289 odbc_next_row() database is not connected! > > But when I test by running the utility isql the database is connected! > > What's going wrong? > > > ________________________________ > > Von: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris > Danielson > Gesendet: Mittwoch, 7. Januar 2009 16:39 > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so > > Anytime. Good luck. > > fidibus83 wrote: > Sorry Chris, my mistake. Mod_spidermonkey_odbc.so does exist now. > > I wanted to said that I'm sorry for my stupid mistake that I uncomment the > wrong thing. > > But thank you very much for your help! > > > ________________________________ > > Von: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris > Danielson > Gesendet: Mittwoch, 7. Januar 2009 16:17 > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so > > Fidibus, > Make sure that you also have installed unixodbc. As shown here: > http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#unixodbc > > Kind Regards, > Chris > > fidibus83 wrote: > Oh, I'm sorry. > > Should I comment mod_spidermonkey_odbc in root/freeswitch/build/modules.conf > again? > > > ________________________________ > > Von: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von > Michael Jerris > Gesendet: Mittwoch, 7. Januar 2009 15:15 > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so > > you would need to modify the modules.conf in the root of the source dir, not > in the build dir. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From vhatz at kinetix.gr Wed Jan 7 10:14:09 2009 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Wed, 07 Jan 2009 20:14:09 +0200 Subject: [Freeswitch-users] mod_opal calls and records In-Reply-To: <4F14448B-6DD4-4B05-96A5-22289C8E5093@wcgltd.com> References: <4F14448B-6DD4-4B05-96A5-22289C8E5093@wcgltd.com> Message-ID: <4964F0F1.6000803@kinetix.gr> Hello, Josh Forman wrote: > I started to do some testing with h323 calls using mod_opal and I have > been having a number of issues. This is interesting, as I wasn't aware that there was a working mod_opal with FS, yet. Which FS version are you using? Best regards, Vlasis Hatzistavrou Kinetix Tele.com International Inc. 306 Victoria House, Victoria, Mahe, Seychelles Tel.: +302310556134 Fax: +302310556134 (ext. 0) GSM: +306977835653 e-mail: vhatz at kinetixtele.com http://www.kinetixtele.com Postal address: Monastiriou 9 & Enotikon 54627 Thessaloniki Greece > > I've been creating calls that come into freeswitch as SIP and output > as h323 and while I have a call open I did not see any entries from > fs_cli when using the "show channels" or "show calls" commands. Also > no CDR is generated from the call. > > I was wondering if someone could tell me if these issues are from > something that just has not been implemented yet in mod_opal or if > this is most likely a configuration issue on my side. > > Thank you, > > Josh > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Wed Jan 7 10:26:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jan 2009 12:26:52 -0600 Subject: [Freeswitch-users] different behaviour on uuid_bridge, doesn't really bridge In-Reply-To: <49648B4C.8050505@gmx.net> References: <49638DC1.2000307@gmx.net> <4963A228.3000103@gmx.net> <191c3a030901061609m19d91c15o8fad4b54efba017b@mail.gmail.com> <49648B4C.8050505@gmx.net> Message-ID: <191c3a030901071026r3ec5866cy5b1cf2938573e0a0@mail.gmail.com> how are you exactly doing this? with event socket? can you give a more detailed account of exactly what commands you are sending extensions used etc. On Wed, Jan 7, 2009 at 5:00 AM, Peter P GMX wrote: > Hello Anthony, > > I updated to SNV=11084 and still have the problem. The behaviour is > slightly changed now. > Step 4+5 (a s below in my mail) > 4) When I bridge A and B, A and B canNOT hear each other (e.g. for 1/2 > sec). A continues to hear its messages, B does not hear anything > 5) When I hangup B then A is still active and does not recognize hangup > of B. > > At former times, when the call was bridged, I had an "unbind" for each > call on the event_socket interface, so event_socket was out of the loop > after uuid_bridge. Now I have an "unbind" for each party only at the > time when the party hangs up. > > Best regards > Peter > > > Anthony Minessale schrieb: > > update one more time and see how that is > > > > > > On Tue, Jan 6, 2009 at 12:25 PM, Peter P GMX > > wrote: > > > > One more info: > > I have updated to the newest SVN version of FS. > > A and B can actually hear each other (just a bit, some scratching) > > while > > the announcement to A is very slow (~50% speed) and very choppy. > > > > Best regards > > peter > > > > Peter P GMX schrieb: > > > I have setup a test machine and a production machine. Since > > recently the > > > production machine behaves differently in terms of uuid bridge. > > > > > > How it should work (and how it worked before) > > > 1) call A comes in > > > 2) I play some messages to A > > > 3) In the meantime I originate a call to B and transfer to an > > > extension, where also some messages are played > > > 4) Then I bridge A and B, so they are dropped off the current > > > announcements an speak to each other > > > 5) when either A or B hangs up, both legs are terminated > > > > > > New behaviour > > > 1) call A comes in > > > 2) I play some messages to A > > > 3) In the meantime I originate a call to B and transfer to an > > > extension, where also some messages are played > > > 4) Then I bridge A and B, A and B can hear each other for 1/2 > sec, > > > then A constinues to hear its messages, B does not hear anything > > > 5) when either A or B hangs up, both legs are terminated > > > > > > 4) is different now! > > > > > > The FS console show some messages about unbridge > > > (SWITCH_MESSAGE_INDICATE_UNBRIDGE), but somehow the calls are still > > > connected until A or B hangs up. > > > > > > Anybody has a clue? > > > Best regards > > > Peter > > > > > > 3) is finished, 4) starts > > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() > zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:35:31 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:35:34 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1094 > > > switch_ivr_uuid_bridge() (OpenZAP/2:3/49171xxxxxxx) State Change > > > CS_EXECUTE -> CS_RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1095 > > > switch_ivr_uuid_bridge() (OpenZAP/2:1/216xxxxx) State Change > > CS_EXECUTE > > > -> CS_RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State EXECUTE > > going to > > > sleep > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State > > Change > > > CS_RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET > > > 4) Start > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > > > uuid_bridge_on_reset() OpenZAP/2:1/216xxxxx CUSTOM RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:53 > > > switch_core_standard_on_reset() OpenZAP/2:1/216xxxxx Standard RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET > > going to sleep > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > > EXECUTE going > > > to sleep > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) Running State > > > Change CS_RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > > > uuid_bridge_on_reset() OpenZAP/2:3/49171xxxxxxx CUSTOM RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:533 > > > uuid_bridge_on_reset() (OpenZAP/2:3/49171xxxxxxx) State Change > > CS_RESET > > > -> CS_SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State RESET > > going > > > to sleep > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) Running State > > > Change CS_SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > > SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > > channel_on_soft_execute() > > > CHANNEL SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > > > uuid_bridge_on_soft_execute() OpenZAP/2:3/49171xxxxxxx CUSTOM > > TRANSMIT > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:571 > > > uuid_bridge_on_soft_execute() (OpenZAP/2:1/216xxxxx) State Change > > > CS_RESET -> CS_SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State > > Change > > > CS_SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > > channel_on_soft_execute() > > > CHANNEL SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > > > uuid_bridge_on_soft_execute() OpenZAP/2:1/216xxxxx CUSTOM TRANSMIT > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:181 > > > switch_core_standard_on_soft_execute() OpenZAP/2:1/216xxxxx > Standard > > > SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State SOFT_EXECUTE > > > going to sleep > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:814 > > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) State > > Change > > > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:856 > > > switch_ivr_multi_threaded_bridge() OpenZAP/2:1/216xxxxx receive > > message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:863 > > > switch_ivr_multi_threaded_bridge() OpenZAP/2:3/49171xxxxxxx receive > > > message [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:907 > > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) State > > Change > > > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State > > Change > > > CS_EXCHANGE_MEDIA > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:436 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > > EXCHANGE_MEDIA > > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:511 > > > channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA > > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > > switch_ivr_parse_event() > > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > > custom/warteschleife_30.wav interrupt_digit 0 ) > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_play_say.c:932 > > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > > 2009-01-06 17:35:36 [DEBUG] switch_core_io.c:610 > > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > > message > > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > > switch_ivr_parse_event() > > > OpenZAP/2:3/49171xxxxxxx Command Execute stop_dtmf(true) > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:229 > > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:231 > > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:229 > > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:231 > > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 31 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7d 02 01 22 30 0d a1 05 30 > > 03 02 01 > > > 08 82 01 00 83 01 00] > > > > > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() > zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:35:46 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 31 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7e 02 01 22 30 0d a1 05 30 > > 03 02 01 > > > 09 82 01 00 83 01 00] > > > > > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() > zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:36:02 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:36:06 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:229 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:36:07 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:231 > > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:08 [DEBUG] switch_ivr.c:391 > > switch_ivr_parse_event() > > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > > custom/warteschleife_30.wav interrupt_digit 0 ) > > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_play_say.c:932 > > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > > 2009-01-06 17:36:08 [DEBUG] switch_core_io.c:610 > > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > > message > > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 31 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7f 02 01 22 30 0d a1 05 30 > > 03 02 01 > > > 0a 82 01 00 83 01 00] > > > > > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() > zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:36:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 32 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 80 02 01 22 30 0d a1 05 > > 30 03 02 > > > 01 0b 82 01 00 83 01 00] > > > > > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() > zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:36:33 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:36:38 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:229 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:36:39 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:231 > > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:40 [DEBUG] switch_ivr.c:391 > > switch_ivr_parse_event() > > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > > > > > custom/dieser_berater_ist_leider_zur_zeit_nicht_erreichbar_bitte_waehlen_sie_einen_ande.wav > > > interrupt_digit 0 ) > > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_play_say.c:932 > > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > > 2009-01-06 17:36:40 [DEBUG] switch_core_io.c:610 > > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > > message > > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:45 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:229 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:36:46 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:231 > > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:47 [DEBUG] switch_ivr.c:391 > > switch_ivr_parse_event() > > > OpenZAP/2:1/216xxxxx Command Execute read(0 4 > > > > > > custom/geben_sie_bitte_nach_dem_signalton_die_kennung_des_gewuenschten_beraters_ein_sch2.wav > > > dtmfdtmf 10000 #,*) > > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_play_say.c:932 > > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > > > 2009-01-06 17:36:47 [DEBUG] switch_core_io.c:610 > > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > > message > > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 32 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 81 02 01 22 30 0d a1 05 > > 30 03 02 > > > 01 0c 82 01 00 83 01 00] > > > > > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() > zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:36:48 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:37:03 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 32 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 82 02 01 22 30 0d a1 05 > > 30 03 02 > > > 01 0d 82 01 00 83 01 00] > > > > > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() > zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:37:04 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 13 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 00 35 45 08 02 80 90 1e 02 82 88] > > > > > > 5) Hangup > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[45] Size:[115] CRV: 53 (0x35, CTX: Originator) > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() > zchan > > > 77df70 (2:1) source isdn_data->channels_remote_crv[0x35] > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:455 zap_isdn_931_34() > > Changing > > > state on 2:1 from UP to TERMINATING > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:813 state_advance() 2:1 > > STATE > > > [TERMINATING] > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1121 state_advance() > > > Terminating: Direction Inbound > > > 2009-01-06 17:37:12 [DEBUG] mod_openzap.c:1418 > > on_clear_channel_signal() > > > got clear channel sig [STOP] > > > 2009-01-06 17:37:12 [NOTICE] mod_openzap.c:1437 > > > on_clear_channel_signal() Hangup OpenZAP/2:1/216xxxxx > > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > > 2009-01-06 17:37:12 [DEBUG] switch_channel.c:1441 > > > switch_channel_perform_hangup() Send signal OpenZAP/2:1/216xxxxx > > [KILL] > > > 2009-01-06 17:37:12 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 > > > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 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 > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/1c646736/attachment-0001.html From msc at freeswitch.org Wed Jan 7 10:30:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 7 Jan 2009 10:30:34 -0800 Subject: [Freeswitch-users] mod_opal calls and records In-Reply-To: <4964F0F1.6000803@kinetix.gr> References: <4F14448B-6DD4-4B05-96A5-22289C8E5093@wcgltd.com> <4964F0F1.6000803@kinetix.gr> Message-ID: <87f2f3b90901071030j2406e448qa291ebde435d4390@mail.gmail.com> Please look very carefully at Tony's comments in the commit: http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=10734 ;) -MC On Wed, Jan 7, 2009 at 10:14 AM, Vlasis Hatzistavrou (KTI) wrote: > Hello, > > Josh Forman wrote: > > I started to do some testing with h323 calls using mod_opal and I have > > been having a number of issues. > > > This is interesting, as I wasn't aware that there was a working mod_opal > with FS, yet. > > Which FS version are you using? > > Best regards, > Vlasis Hatzistavrou > Kinetix Tele.com International Inc. > 306 Victoria House, > Victoria, Mahe, > Seychelles > Tel.: +302310556134 > Fax: +302310556134 (ext. 0) > GSM: +306977835653 > e-mail: vhatz at kinetixtele.com > http://www.kinetixtele.com > > Postal address: > Monastiriou 9 & Enotikon > 54627 > Thessaloniki > Greece > > > > > I've been creating calls that come into freeswitch as SIP and output > > as h323 and while I have a call open I did not see any entries from > > fs_cli when using the "show channels" or "show calls" commands. Also > > no CDR is generated from the call. > > > > I was wondering if someone could tell me if these issues are from > > something that just has not been implemented yet in mod_opal or if > > this is most likely a configuration issue on my side. > > > > Thank you, > > > > Josh > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/447c7990/attachment.html From chris at maxpowersoft.com Wed Jan 7 10:32:09 2009 From: chris at maxpowersoft.com (Chris Danielson) Date: Wed, 07 Jan 2009 10:32:09 -0800 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so In-Reply-To: <11990ade0901070938y14f15ba7le95e1c6edfaad70f@mail.gmail.com> References: <014301c970e5$740cd6f0$6445310a@Franzi> <015a01c970e8$7fd66390$6445310a@Franzi> <11990ade0901070938y14f15ba7le95e1c6edfaad70f@mail.gmail.com> Message-ID: <4964F529.8060404@maxpowersoft.com> Make sure to follow this example: http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#General_Configuration Remember to use the isql client to test your DSN connection. When that passes you'll be home free. Regards, Chris Stephen Crosby wrote: > I'm assuming you have your driver set up properly since you left that > part out, but I got mine working a few weeks ago and I'll show you > what I've got if it helps: > > odbc.ini: > [freeswitch_ph] > Driver = /usr/lib/odbc/libmyodbc.so > Description = MyODBC 3.51 Driver DSN > SERVER = localhost > PORT = > USER = freeswitch_ph > Password = > Database = freeswitch_ph > OPTION = 3 > SOCKET = > > somescript.js > use("ODBC"); > var DSN = 'freeswitch_ph'; > var DB_USER = 'freeswitch_ph'; > var DB_PASS = ''; > var db = new ODBC(DSN, DB_USER, DB_PASS); > db.connect(); > > > > On Wed, Jan 7, 2009 at 8:53 AM, fidibus83 wrote: > >> Yes, but why isn't it found? >> >> >> >> My odbc.ini configuration: >> >> >> >> [fs01_odbc] >> >> Driver=MySQL >> >> SERVER=localhost >> >> PORT=3306 >> >> DATABASE=freeswitch >> >> Socket = /var/lib/mysql/mysql.sock >> >> >> >> >> >> >> >> Javascript: >> >> >> >> ? >> >> use("ODBC"); >> >> var DSN="fs01_odbc"; >> >> var DB_USER="XXXXX"; >> >> var DB_PASS="YYYYY"; >> >> var db = new ODBC(DSN, DB_USER, DB_PASS); >> >> db.connect(); >> >> ?.. >> >> >> >> ________________________________ >> >> Von: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Ken >> Rice >> Gesendet: Mittwoch, 7. Januar 2009 17:42 >> An: freeswitch-users at lists.freeswitch.org >> Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so >> >> >> >> Well your error tells you explicity DSN not found... >> >> ________________________________ >> >> From: fidibus83 >> Reply-To: >> Date: Wed, 7 Jan 2009 17:31:54 +0100 >> To: >> Subject: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so >> >> I have another problem! >> >> I want to connect to a mysql database through javascript. >> >> I get this error: >> [ERR] switch_odbc.c:160 switch_odbc_handle_connect() STATE: IM002 CODE 0 >> ERROR: [unixODBC] [Driver Manager]Data source name not found, and no default >> driver specified >> [ERR] mod_spidermonkey_odbc.c:233 odbc_exec() Database is not connected! >> [ERR] mod_spidermonkey_odbc.c:289 odbc_next_row() database is not connected! >> >> But when I test by running the utility isql the database is connected! >> >> What's going wrong? >> >> >> ________________________________ >> >> Von: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris >> Danielson >> Gesendet: Mittwoch, 7. Januar 2009 16:39 >> An: freeswitch-users at lists.freeswitch.org >> Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so >> >> Anytime. Good luck. >> >> fidibus83 wrote: >> Sorry Chris, my mistake. Mod_spidermonkey_odbc.so does exist now. >> >> I wanted to said that I'm sorry for my stupid mistake that I uncomment the >> wrong thing. >> >> But thank you very much for your help! >> >> >> ________________________________ >> >> Von: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris >> Danielson >> Gesendet: Mittwoch, 7. Januar 2009 16:17 >> An: freeswitch-users at lists.freeswitch.org >> Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so >> >> Fidibus, >> Make sure that you also have installed unixodbc. As shown here: >> http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#unixodbc >> >> Kind Regards, >> Chris >> >> fidibus83 wrote: >> Oh, I'm sorry. >> >> Should I comment mod_spidermonkey_odbc in root/freeswitch/build/modules.conf >> again? >> >> >> ________________________________ >> >> Von: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von >> Michael Jerris >> Gesendet: Mittwoch, 7. Januar 2009 15:15 >> An: freeswitch-users at lists.freeswitch.org >> Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so >> >> you would need to modify the modules.conf in the root of the source dir, not >> in the build dir. >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/fb214769/attachment.html From matthew at matthew.at Wed Jan 7 10:33:23 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Wed, 07 Jan 2009 10:33:23 -0800 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: References: <49644A22.8050802@matthew.at> Message-ID: <4964F573.2050307@matthew.at> Updated to current a couple days ago, default dialplan, extensions are registered as 1001 and 1002. tcpdump on the freeswitch host shows RTP going both ways to/from both phones. This is running on CentOS, so your log and debug instructions don't make much sense, but I do have SIP tracing enabled... see http://pastebin.freeswitch.org/6694 The phones are on the 10.10.155.0/24 subnet, the switch is on 198.202.199.1, but there's no NAT between, just a router... from the POV of the switch, it can see all the phones on the 10.10.155.0/24 network directly. Matthew Kaufman Michael S Collins wrote: > What FS revision and is this a default Dialplan? Please pastebin the > output of the CLI while making test calls. Be sure to press F8 to > enable debug messages. Also, if you can do so turn on SIP messages by > launching FreeSWITCH with TPORT_LOG=1. > > -MC > > Sent from my iPhone > > On Jan 6, 2009, at 10:22 PM, Matthew Kaufman wrote: > > >> I have two Polycom phones (one 550 and one 650) successfully >> registered >> to the switch. If I call from either extension to the other and >> answer, >> audio flows from the calling party to the called party, but audio does >> not flow from the called party back to the calling party. Even more >> strange, the called party does not answer, then when the call is >> sent to >> voicemail, the calling party *also* does not hear any of the voicemail >> greeting, though they are recorded successfully. Calling the voicemail >> box directly from either phone *does* work, and the calling party can >> hear the prompts just fine in that case. >> >> There is no NAT between the phones and FreeSWITCH. >> >> Suggestions? >> >> Matthew Kaufman >> matthew at matthew.at >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Jan 7 10:37:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 7 Jan 2009 12:37:03 -0600 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <4964F573.2050307@matthew.at> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> Message-ID: <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> What svn rev and what firmware rev on the phone? On Jan 7, 2009, at 12:33 PM, Matthew Kaufman wrote: > Updated to current a couple days ago, default dialplan, extensions are > registered as 1001 and 1002. > > tcpdump on the freeswitch host shows RTP going both ways to/from both > phones. > > This is running on CentOS, so your log and debug instructions don't > make > much sense, but I do have SIP tracing enabled... see > http://pastebin.freeswitch.org/6694 > > The phones are on the 10.10.155.0/24 subnet, the switch is on > 198.202.199.1, but there's no NAT between, just a router... from the > POV > of the switch, it can see all the phones on the 10.10.155.0/24 network > directly. > > Matthew Kaufman -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/23e11c5c/attachment-0001.html From matthew at matthew.at Wed Jan 7 10:46:43 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Wed, 07 Jan 2009 10:46:43 -0800 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> Message-ID: <4964F893.6020302@matthew.at> Brian West wrote: > What svn rev and what firmware rev on the phone? Looks like 1.0 trunk 11009M for the switch, 2.1.2.0078 SIP on the phone (what it shipped with, haven't updated to latest firmware) Matthew Kaufman From vhatz at kinetix.gr Wed Jan 7 10:49:07 2009 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Wed, 07 Jan 2009 20:49:07 +0200 Subject: [Freeswitch-users] mod_opal calls and records In-Reply-To: <87f2f3b90901071030j2406e448qa291ebde435d4390@mail.gmail.com> References: <4F14448B-6DD4-4B05-96A5-22289C8E5093@wcgltd.com> <4964F0F1.6000803@kinetix.gr> <87f2f3b90901071030j2406e448qa291ebde435d4390@mail.gmail.com> Message-ID: <4964F923.3060100@kinetix.gr> Hello Michael, I went through a quick view of the files of mod_opal. Is it a continuation of Tuyan Ozipek's work on mod_opal? I was under the impression that that mod_opal was to be discarded in order to be replaced by a new one... Best regards, Vlasis Hatzistavrou. Michael Collins wrote: > Please look very carefully at Tony's comments in the commit: > http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=10734 > > ;) > -MC > > On Wed, Jan 7, 2009 at 10:14 AM, Vlasis Hatzistavrou (KTI) > > wrote: > > Hello, > > Josh Forman wrote: > > I started to do some testing with h323 calls using mod_opal and I > have > > been having a number of issues. > > > This is interesting, as I wasn't aware that there was a working mod_opal > with FS, yet. > > Which FS version are you using? > > Best regards, > Vlasis Hatzistavrou > Kinetix Tele.com International Inc. > 306 Victoria House, > Victoria, Mahe, > Seychelles > Tel.: +302310556134 > Fax: +302310556134 (ext. 0) > GSM: +306977835653 > e-mail: vhatz at kinetixtele.com > http://www.kinetixtele.com > > Postal address: > Monastiriou 9 & Enotikon > 54627 > Thessaloniki > Greece > > > > > I've been creating calls that come into freeswitch as SIP and output > > as h323 and while I have a call open I did not see any entries from > > fs_cli when using the "show channels" or "show calls" commands. Also > > no CDR is generated from the call. > > > > I was wondering if someone could tell me if these issues are from > > something that just has not been implemented yet in mod_opal or if > > this is most likely a configuration issue on my side. > > > > Thank you, > > > > Josh > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Jan 7 10:55:09 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 7 Jan 2009 12:55:09 -0600 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <4964F893.6020302@matthew.at> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> <4964F893.6020302@matthew.at> Message-ID: <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> Please update the phone to 3.1.1.0137 http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_3_1_1_release_sig.zip We can't support anything but the latest firmware... the same applies to most phones since we routinely report, fix and work around issues. 2.1.2 is very old. /b On Jan 7, 2009, at 12:46 PM, Matthew Kaufman wrote: > Brian West wrote: >> What svn rev and what firmware rev on the phone? > Looks like 1.0 trunk 11009M for the switch, 2.1.2.0078 SIP on the > phone > (what it shipped with, haven't updated to latest firmware) > > Matthew Kaufman > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From matthew at matthew.at Wed Jan 7 11:00:30 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Wed, 07 Jan 2009 11:00:30 -0800 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> <4964F893.6020302@matthew.at> <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> Message-ID: <4964FBCE.8090008@matthew.at> Brian West wrote: > Please update the phone to 3.1.1.0137 > > http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_3_1_1_release_sig.zip > > We can't support anything but the latest firmware... the same applies > to most phones since we routinely report, fix and work around issues. > 2.1.2 is very old. > > /b > I'll do that. I wanted to change as few things at a time as possible, so I'll make that the next test and let you know. Do note that 2.1.2, while old, is still the current ship version from Polycom, per their matrix. Matthew Kaufman From anthony.minessale at gmail.com Wed Jan 7 11:14:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jan 2009 13:14:39 -0600 Subject: [Freeswitch-users] mod_opal calls and records In-Reply-To: <4964F923.3060100@kinetix.gr> References: <4F14448B-6DD4-4B05-96A5-22289C8E5093@wcgltd.com> <4964F0F1.6000803@kinetix.gr> <87f2f3b90901071030j2406e448qa291ebde435d4390@mail.gmail.com> <4964F923.3060100@kinetix.gr> Message-ID: <191c3a030901071114g194ae214o9ee6e78f08aa0d6c@mail.gmail.com> yes, The author of opal himself has put a lot of effort into making this possible. I announced it several times and asked people to test it and nobody seemed to notice. We had a fundraiser and only a small handful of people contributed so it took a long time to get it in tree. On Wed, Jan 7, 2009 at 12:49 PM, Vlasis Hatzistavrou (KTI) wrote: > Hello Michael, > > I went through a quick view of the files of mod_opal. Is it a > continuation of Tuyan Ozipek's work on mod_opal? I was under the > impression that that mod_opal was to be discarded in order to be > replaced by a new one... > > Best regards, > Vlasis Hatzistavrou. > > Michael Collins wrote: > > Please look very carefully at Tony's comments in the commit: > > http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=10734 > > > > ;) > > -MC > > > > On Wed, Jan 7, 2009 at 10:14 AM, Vlasis Hatzistavrou (KTI) > > > wrote: > > > > Hello, > > > > Josh Forman wrote: > > > I started to do some testing with h323 calls using mod_opal and I > > have > > > been having a number of issues. > > > > > > This is interesting, as I wasn't aware that there was a working > mod_opal > > with FS, yet. > > > > Which FS version are you using? > > > > Best regards, > > Vlasis Hatzistavrou > > Kinetix Tele.com International Inc. > > 306 Victoria House, > > Victoria, Mahe, > > Seychelles > > Tel.: +302310556134 > > Fax: +302310556134 (ext. 0) > > GSM: +306977835653 > > e-mail: vhatz at kinetixtele.com > > http://www.kinetixtele.com > > > > Postal address: > > Monastiriou 9 & Enotikon > > 54627 > > Thessaloniki > > Greece > > > > > > > > I've been creating calls that come into freeswitch as SIP and > output > > > as h323 and while I have a call open I did not see any entries > from > > > fs_cli when using the "show channels" or "show calls" commands. > Also > > > no CDR is generated from the call. > > > > > > I was wondering if someone could tell me if these issues are from > > > something that just has not been implemented yet in mod_opal or if > > > this is most likely a configuration issue on my side. > > > > > > Thank you, > > > > > > Josh > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at 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 at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/49e49088/attachment.html From woof at nortel.com Wed Jan 7 11:20:14 2009 From: woof at nortel.com (Andy Spitzer) Date: Wed, 07 Jan 2009 14:20:14 -0500 Subject: [Freeswitch-users] Global limit on the number of conference legs Message-ID: Woof! Does FS have a way of limiting the total number of conference legs on a box? I am aware that each individual conference profile can have a "max-members" param, but what I'm looking for would span multiple conferences, with a maximum leg limit per server, regardless of the per conference limit. Something like switch.conf "max-sessions", but only for conference legs, with an appropriate message played to the caller if it is exceeded. --Woof From anthony.minessale at gmail.com Wed Jan 7 11:31:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jan 2009 13:31:54 -0600 Subject: [Freeswitch-users] Global limit on the number of conference legs In-Reply-To: References: Message-ID: <191c3a030901071131g59447eb4rb706a5e24269d13@mail.gmail.com> we don't currently have anything like that. It would probably entail a global counter and a new section in the config files for global params. On Wed, Jan 7, 2009 at 1:20 PM, Andy Spitzer wrote: > Woof! > > Does FS have a way of limiting the total number of conference legs on a > box? I am aware that each individual conference profile can have a > "max-members" param, but what I'm looking for would span multiple > conferences, with a maximum leg limit per server, regardless of the per > conference limit. > > Something like switch.conf "max-sessions", but only for conference legs, > with an appropriate message played to the caller if it is exceeded. > > --Woof > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/c78d393d/attachment-0001.html From woof at nortel.com Wed Jan 7 11:36:04 2009 From: woof at nortel.com (Andy Spitzer) Date: Wed, 07 Jan 2009 14:36:04 -0500 Subject: [Freeswitch-users] Global limit on the number of conference legs In-Reply-To: <191c3a030901071131g59447eb4rb706a5e24269d13@mail.gmail.com> References: <191c3a030901071131g59447eb4rb706a5e24269d13@mail.gmail.com> Message-ID: Woof! On Wed, 07 Jan 2009 14:31:54 -0500, Anthony Minessale wrote: > we don't currently have anything like that. Okay. Just wanted to make sure I wasn't missing something obvious. --Woof From matthew at matthew.at Wed Jan 7 11:44:55 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Wed, 07 Jan 2009 11:44:55 -0800 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <4964FBCE.8090008@matthew.at> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> <4964F893.6020302@matthew.at> <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> <4964FBCE.8090008@matthew.at> Message-ID: <49650637.7040703@matthew.at> Matthew Kaufman wrote: > Brian West wrote: > >> Please update the phone to 3.1.1.0137 >> >> http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_3_1_1_release_sig.zip >> >> We can't support anything but the latest firmware... the same applies >> to most phones since we routinely report, fix and work around issues. >> 2.1.2 is very old. >> >> /b >> >> > I'll do that. I wanted to change as few things at a time as possible, so > I'll make that the next test and let you know. > > I have upgraded to the latest boot roms and SIP firmware, 3.1.1.0137. I get the same one-way audio between the polycom and x-lite, and now on a polycom-polycom call I get no audio in *either* direction. (Not much of an improvement, but different) Trace of the call using the newest firmware is at http://pastebin.freeswitch.org/6695 Matthew Kaufman From anthony.minessale at gmail.com Wed Jan 7 12:54:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jan 2009 14:54:40 -0600 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <49650637.7040703@matthew.at> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> <4964F893.6020302@matthew.at> <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> <4964FBCE.8090008@matthew.at> <49650637.7040703@matthew.at> Message-ID: <191c3a030901071254t756b7da9r7e070e8909497a9a@mail.gmail.com> could you use a pcap util like tcpdump or wireshark to capture the traffic so we can see the rtp too that may help to figure it out. On Wed, Jan 7, 2009 at 1:44 PM, Matthew Kaufman wrote: > Matthew Kaufman wrote: > > Brian West wrote: > > > >> Please update the phone to 3.1.1.0137 > >> > >> > http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_3_1_1_release_sig.zip > >> > >> We can't support anything but the latest firmware... the same applies > >> to most phones since we routinely report, fix and work around issues. > >> 2.1.2 is very old. > >> > >> /b > >> > >> > > I'll do that. I wanted to change as few things at a time as possible, so > > I'll make that the next test and let you know. > > > > > I have upgraded to the latest boot roms and SIP firmware, 3.1.1.0137. I > get the same one-way audio between the polycom and x-lite, and now on a > polycom-polycom call I get no audio in *either* direction. (Not much of > an improvement, but different) > > Trace of the call using the newest firmware is at > http://pastebin.freeswitch.org/6695 > > Matthew Kaufman > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/fa554b43/attachment.html From matthew at matthew.at Wed Jan 7 13:21:57 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Wed, 07 Jan 2009 13:21:57 -0800 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <191c3a030901071254t756b7da9r7e070e8909497a9a@mail.gmail.com> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> <4964F893.6020302@matthew.at> <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> <4964FBCE.8090008@matthew.at> <49650637.7040703@matthew.at> <191c3a030901071254t756b7da9r7e070e8909497a9a@mail.gmail.com> Message-ID: <49651CF5.6090501@matthew.at> Anthony Minessale wrote: > could you use a pcap util like tcpdump or wireshark to capture the traffic > so we can see the rtp too that may help to figure it out. Dump at the switch side of all traffic to/from the originating Polycom is at http://pastebin.freeswitch.org/6696 I kept the call short to keep the total amount of RTP to a reasonable size. Matthew Kaufman From anthony.minessale at gmail.com Wed Jan 7 14:18:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jan 2009 16:18:31 -0600 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <49651CF5.6090501@matthew.at> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> <4964F893.6020302@matthew.at> <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> <4964FBCE.8090008@matthew.at> <49650637.7040703@matthew.at> <191c3a030901071254t756b7da9r7e070e8909497a9a@mail.gmail.com> <49651CF5.6090501@matthew.at> Message-ID: <191c3a030901071418w63ad1f07p66851e27e1d1f13e@mail.gmail.com> is it possible to get a binary pcap? That way we can look at it in wireshark. you can email it direct to me and brian On Wed, Jan 7, 2009 at 3:21 PM, Matthew Kaufman wrote: > Anthony Minessale wrote: > > could you use a pcap util like tcpdump or wireshark to capture the > traffic > > so we can see the rtp too that may help to figure it out. > Dump at the switch side of all traffic to/from the originating Polycom > is at http://pastebin.freeswitch.org/6696 > > I kept the call short to keep the total amount of RTP to a reasonable size. > > Matthew Kaufman > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/9d670ce7/attachment.html From testeador01 at gmail.com Wed Jan 7 12:39:04 2009 From: testeador01 at gmail.com (tester tato) Date: Wed, 7 Jan 2009 15:39:04 -0500 Subject: [Freeswitch-users] fs doesn't realize caller hanged up when a call from PSTN is on hold Message-ID: Hello everyone, Thanks in advance if anyone wants to help me in a couple of questions; I set up my fs plant and included a grandstream fxo gateway gxw4104, the gateway redirects all incomming calls from the PSTN to the extension 1005 (with its default sample configuration). This is what happens: 1. I get a call from the PSTN, pick it up and i decide to put it on hold while i do something else, the caller hears the moh. 2. If the caller gets tired or bored of waiting and hangs up, my FS plant doesnt realize that the caller hung up, so the channel stays busy, so does the gateway line and my phone still acts as if i had a call on hold. 3. When i pick up the phonecall that was on hold all i hear (obviously) is the busy tone from the PSTN. My questions are: Is this behavior only from this particular gateway? or does it happen with other FXO devices. Is there a way to specify that i want the call to hang up itself when the caller hangs up so the channel gets released? I know that the call actually gets hanged up and realeases the channel if the caller was another SIP extension on my plant, it just doesn't work with calls from the pstn under my configuration. I highly appreciate any light you can throw to me on this matter. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/24733050/attachment.html From mike at jerris.com Wed Jan 7 15:41:31 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 7 Jan 2009 18:41:31 -0500 Subject: [Freeswitch-users] Global limit on the number of conference legs In-Reply-To: References: <191c3a030901071131g59447eb4rb706a5e24269d13@mail.gmail.com> Message-ID: <651102C0-5D38-41DE-987A-71A9CC223B7D@jerris.com> you can however use mod_limit to implement this yourself with dialplan logic as long as it is used before all calls to the conference (it wouldn't work for outbound calls from the conference without a little bit of thought) Mike On Jan 7, 2009, at 2:36 PM, Andy Spitzer wrote: > Woof! > > On Wed, 07 Jan 2009 14:31:54 -0500, Anthony Minessale > wrote: > >> we don't currently have anything like that. > > Okay. Just wanted to make sure I wasn't missing something obvious. > From msc at freeswitch.org Wed Jan 7 15:55:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 7 Jan 2009 15:55:07 -0800 Subject: [Freeswitch-users] fs doesn't realize caller hanged up when a call from PSTN is on hold In-Reply-To: References: Message-ID: <87f2f3b90901071555n6b6d6d11x54c88a3ce920d21b@mail.gmail.com> What version of FreeSWITCH are you on? Also, can you get a debug trace from when the far end hangs up? We definitely need to know if the gxw is alerting FS to the fact that the far end actually hung up, which is in doubt right now. Thanks, MC On Wed, Jan 7, 2009 at 12:39 PM, tester tato wrote: > Hello everyone, > > Thanks in advance if anyone wants to help me in a couple of questions; I > set up my fs plant and included a grandstream fxo gateway gxw4104, the > gateway redirects all incomming calls from the PSTN to the extension 1005 > (with its default sample configuration). > This is what happens: > > 1. I get a call from the PSTN, pick it up and i decide to put it on hold > while i do something else, the caller hears the moh. > 2. If the caller gets tired or bored of waiting and hangs up, my FS > plant doesnt realize that the caller hung up, so the channel stays busy, so > does the gateway line and my phone still acts as if i had a call on hold. > 3. When i pick up the phonecall that was on hold all i hear (obviously) > is the busy tone from the PSTN. > > My questions are: > Is this behavior only from this particular gateway? or does it happen with > other FXO devices. > Is there a way to specify that i want the call to hang up itself when the > caller hangs up so the channel gets released? > > I know that the call actually gets hanged up and realeases the channel if > the caller was another SIP extension on my plant, it just doesn't work with > calls from the pstn under my configuration. > > I highly appreciate any light you can throw to me on this matter. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/fbb84f65/attachment-0001.html From woof at nortel.com Wed Jan 7 17:14:15 2009 From: woof at nortel.com (Andy Spitzer) Date: Wed, 07 Jan 2009 20:14:15 -0500 Subject: [Freeswitch-users] Global limit on the number of conference legs In-Reply-To: <651102C0-5D38-41DE-987A-71A9CC223B7D@jerris.com> References: <191c3a030901071131g59447eb4rb706a5e24269d13@mail.gmail.com> <651102C0-5D38-41DE-987A-71A9CC223B7D@jerris.com> Message-ID: Woof! On Wed, 07 Jan 2009 18:41:31 -0500, Michael Jerris wrote: > you can however use mod_limit to implement this yourself with dialplan > logic as long as it is used before all calls to the conference (it > wouldn't work for outbound calls from the conference without a little > bit of thought) Good point. Thanks for the info. --Woof From brian at freeswitch.org Wed Jan 7 17:24:38 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 7 Jan 2009 19:24:38 -0600 Subject: [Freeswitch-users] Global limit on the number of conference legs In-Reply-To: References: <191c3a030901071131g59447eb4rb706a5e24269d13@mail.gmail.com> <651102C0-5D38-41DE-987A-71A9CC223B7D@jerris.com> Message-ID: <9679CC8A-4C67-4EB0-8BC9-14EAD4FAADEB@freeswitch.org> make outbound calls use loopback.. same realm.. done. ;) /b On Jan 7, 2009, at 7:14 PM, Andy Spitzer wrote: > Woof! > > On Wed, 07 Jan 2009 18:41:31 -0500, Michael Jerris > wrote: > >> you can however use mod_limit to implement this yourself with >> dialplan >> logic as long as it is used before all calls to the conference (it >> wouldn't work for outbound calls from the conference without a little >> bit of thought) > > Good point. Thanks for the info. > > --Woof -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/42964c38/attachment.html From gservat at gmail.com Wed Jan 7 18:14:40 2009 From: gservat at gmail.com (Gonzalo Servat) Date: Thu, 8 Jan 2009 00:14:40 -0200 Subject: [Freeswitch-users] fs doesn't realize caller hanged up when a call from PSTN is on hold In-Reply-To: References: Message-ID: On Wed, Jan 7, 2009 at 6:39 PM, tester tato wrote: > Hello everyone, > > Thanks in advance if anyone wants to help me in a couple of questions; I > set up my fs plant and included a grandstream fxo gateway gxw4104, the > gateway redirects all incomming calls from the PSTN to the extension 1005 > (with its default sample configuration). > This is what happens: > > 1. I get a call from the PSTN, pick it up and i decide to put it on hold > while i do something else, the caller hears the moh. > 2. If the caller gets tired or bored of waiting and hangs up, my FS > plant doesnt realize that the caller hung up, so the channel stays busy, so > does the gateway line and my phone still acts as if i had a call on hold. > 3. When i pick up the phonecall that was on hold all i hear (obviously) > is the busy tone from the PSTN. > > My questions are: > Is this behavior only from this particular gateway? or does it happen with > other FXO devices. > Is there a way to specify that i want the call to hang up itself when the > caller hangs up so the channel gets released? > > I know that the call actually gets hanged up and realeases the channel if > the caller was another SIP extension on my plant, it just doesn't work with > calls from the pstn under my configuration. > > I highly appreciate any light you can throw to me on this matter. > You might need to use tone_detect. See the example at the bottom (this is how I use it to detect a hang up as my telco doesn't vary the voltage levels on hangup) - Gonzalo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/8d6ac5c7/attachment.html From matthew at matthew.at Wed Jan 7 21:02:54 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Wed, 07 Jan 2009 21:02:54 -0800 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <49650637.7040703@matthew.at> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> <4964F893.6020302@matthew.at> <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> <4964FBCE.8090008@matthew.at> <49650637.7040703@matthew.at> Message-ID: <496588FE.60409@matthew.at> Matthew Kaufman wrote: > I have upgraded to the latest boot roms and SIP firmware, 3.1.1.0137. I > get the same one-way audio between the polycom and x-lite, and now on a > polycom-polycom call I get no audio in *either* direction. (Not much of > an improvement, but different) > For those following on the list, a successful workaround is to set "inbound-proxy-media" to true. Why that should be necessary, and why it behaves the way it does when that is set to false (the strangest part being that calls that go directly to VM have good audio, but calls that ring the far end for a time and then go to VM have no audio *even* when they've gone over to VM), I still don't understand. Matthew Kaufman From kristian.kielhofner at gmail.com Wed Jan 7 22:50:22 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 8 Jan 2009 01:50:22 -0500 Subject: [Freeswitch-users] Export variables from originate command Message-ID: <2d9149cd0901072250o4d08be02m1f4a3e6276d750e4@mail.gmail.com> Hello everyone, I'd like to give some variables to the originate command that pass (export) to the application I am calling. So, if I am calling the transfer app (from originate) to go back into the dialplan I'd like to use the variables in the dialplan that were passed to my original originate command (via the transfer application). Clear as mud, right? ;) Is this possible? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From matthew at matthew.at Wed Jan 7 22:51:13 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Wed, 07 Jan 2009 22:51:13 -0800 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <496588FE.60409@matthew.at> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> <4964F893.6020302@matthew.at> <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> <4964FBCE.8090008@matthew.at> <49650637.7040703@matthew.at> <496588FE.60409@matthew.at> Message-ID: <4965A261.9030708@matthew.at> Matthew Kaufman wrote: > Matthew Kaufman wrote: > >> I have upgraded to the latest boot roms and SIP firmware, 3.1.1.0137. I >> get the same one-way audio between the polycom and x-lite, and now on a >> polycom-polycom call I get no audio in *either* direction. (Not much of >> an improvement, but different) >> >> > For those following on the list, a successful workaround is to set > "inbound-proxy-media" to true. Why that should be necessary, and why it > behaves the way it does when that is set to false (the strangest part > being that calls that go directly to VM have good audio, but calls that > ring the far end for a time and then go to VM have no audio *even* when > they've gone over to VM), I still don't understand. > > I spoke too soon. If I turn on "inbound-proxy-media", then it goes back to "called party can hear calling party, but calling party calling party cannot hear called party" (the same as it was before upgrading to the latest Polycom firmware), and additionally the calling party now gets ringback (didn't before), but if the called party doesn't answer instead of dropping to voicemail it goes to fast busy. The last is probably related to: "[ERR] sofia_glue.c:1608 sofia_glue_tech_set_codec() No audio codec available" which then fires INCOMPATIBLE_DESTINATION. I also picked up version 11089 *and* threw out all the conf directory and regenerated it from the sample source, so this is 100% today-build, default-settings (except for the adjustments to inbound-proxy-media). Matthew Kaufman From jonas.gauffin at gmail.com Wed Jan 7 23:07:09 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Thu, 8 Jan 2009 08:07:09 +0100 Subject: [Freeswitch-users] Building FS on win In-Reply-To: <1675B229-6F41-4F98-B4D2-48EDCDD3F037@jerris.com> References: <1675B229-6F41-4F98-B4D2-48EDCDD3F037@jerris.com> Message-ID: thanks mate. On Wed, Jan 7, 2009 at 3:26 PM, Michael Jerris wrote: > New Revision: 11085 > Log: fix windows build breakage from svn rev 11084 > > On Jan 7, 2009, at 6:18 AM, Jonas Gauffin wrote: > > > I'm trying to build latest trunk on win (vs2008) and get the > > following errors: > > > > 12>mod_spidermonkey.obj : error LNK2001: unresolved external symbol > > _switch_dso_open > > 12>mod_spidermonkey.obj : error LNK2001: unresolved external symbol > > _switch_dso_data_sym > > > > any thoughts? > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/dea88116/attachment.html From fidibus83 at aol.com Thu Jan 8 02:29:50 2009 From: fidibus83 at aol.com (fidibus83) Date: Thu, 8 Jan 2009 11:29:50 +0100 Subject: [Freeswitch-users] question about queue which calls extension Message-ID: <009c01c9717c$0ade3d30$6445310a@Franzi> Hello! I have question about http://wiki.freeswitch.org/wiki/Queue_which_calls_extensions . The javascript connectqueue.js access on tables in the database. Do I have to create them myself? How do the agents register to the queue? Thanks! fidibus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/ba423108/attachment-0001.html From fidibus83 at aol.com Thu Jan 8 02:33:40 2009 From: fidibus83 at aol.com (fidibus83) Date: Thu, 8 Jan 2009 11:33:40 +0100 Subject: [Freeswitch-users] no file mod_spidermonkey_odbc.so In-Reply-To: <4964F529.8060404@maxpowersoft.com> References: <014301c970e5$740cd6f0$6445310a@Franzi> <015a01c970e8$7fd66390$6445310a@Franzi><11990ade0901070938y14f15ba7le95e1c6edfaad70f@mail.gmail.com> <4964F529.8060404@maxpowersoft.com> Message-ID: <00a401c9717c$9362f830$6445310a@Franzi> Thanks for your help! It?s working! I insert the user and the password in odbc.ini. _____ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris Danielson Gesendet: Mittwoch, 7. Januar 2009 19:32 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Make sure to follow this example: http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#General_Configuration Remember to use the isql client to test your DSN connection. When that passes you'll be home free. Regards, Chris Stephen Crosby wrote: I'm assuming you have your driver set up properly since you left that part out, but I got mine working a few weeks ago and I'll show you what I've got if it helps: odbc.ini: [freeswitch_ph] Driver = /usr/lib/odbc/libmyodbc.so Description = MyODBC 3.51 Driver DSN SERVER = localhost PORT = USER = freeswitch_ph Password = Database = freeswitch_ph OPTION = 3 SOCKET = somescript.js use("ODBC"); var DSN = 'freeswitch_ph'; var DB_USER = 'freeswitch_ph'; var DB_PASS = ''; var db = new ODBC(DSN, DB_USER, DB_PASS); db.connect(); On Wed, Jan 7, 2009 at 8:53 AM, fidibus83 wrote: Yes, but why isn't it found? My odbc.ini configuration: [fs01_odbc] Driver=MySQL SERVER=localhost PORT=3306 DATABASE=freeswitch Socket = /var/lib/mysql/mysql.sock Javascript: use("ODBC"); var DSN="fs01_odbc"; var DB_USER="XXXXX"; var DB_PASS="YYYYY"; var db = new ODBC(DSN, DB_USER, DB_PASS); db.connect(); .. ________________________________ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Ken Rice Gesendet: Mittwoch, 7. Januar 2009 17:42 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Well your error tells you explicity DSN not found... ________________________________ From: fidibus83 Reply-To: Date: Wed, 7 Jan 2009 17:31:54 +0100 To: Subject: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so I have another problem! I want to connect to a mysql database through javascript. I get this error: [ERR] switch_odbc.c:160 switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC] [Driver Manager]Data source name not found, and no default driver specified [ERR] mod_spidermonkey_odbc.c:233 odbc_exec() Database is not connected! [ERR] mod_spidermonkey_odbc.c:289 odbc_next_row() database is not connected! But when I test by running the utility isql the database is connected! What's going wrong? ________________________________ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris Danielson Gesendet: Mittwoch, 7. Januar 2009 16:39 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Anytime. Good luck. fidibus83 wrote: Sorry Chris, my mistake. Mod_spidermonkey_odbc.so does exist now. I wanted to said that I'm sorry for my stupid mistake that I uncomment the wrong thing. But thank you very much for your help! ________________________________ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Chris Danielson Gesendet: Mittwoch, 7. Januar 2009 16:17 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so Fidibus, Make sure that you also have installed unixodbc. As shown here: http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#unixodbc Kind Regards, Chris fidibus83 wrote: Oh, I'm sorry. Should I comment mod_spidermonkey_odbc in root/freeswitch/build/modules.conf again? ________________________________ Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 7. Januar 2009 15:15 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] no file mod_spidermonkey_odbc.so you would need to modify the modules.conf in the root of the source dir, not in the build dir. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/17c6c49e/attachment-0001.html From Prometheus001 at gmx.net Thu Jan 8 02:45:46 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 08 Jan 2009 11:45:46 +0100 Subject: [Freeswitch-users] different behaviour on uuid_bridge, doesn't really bridge In-Reply-To: <191c3a030901071026r3ec5866cy5b1cf2938573e0a0@mail.gmail.com> References: <49638DC1.2000307@gmx.net> <4963A228.3000103@gmx.net> <191c3a030901061609m19d91c15o8fad4b54efba017b@mail.gmail.com> <49648B4C.8050505@gmx.net> <191c3a030901071026r3ec5866cy5b1cf2938573e0a0@mail.gmail.com> Message-ID: <4965D95A.10905@gmx.net> Yes, I am doing it via event socket. On a system with pure SIP it works, but on a system where on both logs openzap is used, it doesn't work. So maybe it's an openzap problem? I do the following via event socket: Leg B is doing the bridge via "uuid_bridge " while leg A is playing a soundfile via uuid_playback I can see on the FS side that FS tries to bridge and then then does an unbridge: (see also at the end of this mail history) > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] I recognized that my application detects the bridge approach and initiates the playback of another sound file. This causes, that the soundfile is played, and so the parties cannot hear each other. I changed the behaviour of my application, so that it doesn't try to play another sound file. Now both legs can hear each other. So the call is bridged, although FS indicates a different message. The most important part for my application is though, that for me FS doesn't really complete the bridge so that event socket is still in the loop. This keeps 2 connections for event socket open and blocks resources (I am working with Ruby and eventmachine, there is a limit of 20 simultaneous connections). Best regards Peter Anthony Minessale schrieb: > how are you exactly doing this? > with event socket? > > can you give a more detailed account of exactly what commands you are > sending extensions used etc. > > > On Wed, Jan 7, 2009 at 5:00 AM, Peter P GMX > wrote: > > Hello Anthony, > > I updated to SNV=11084 and still have the problem. The behaviour is > slightly changed now. > Step 4+5 (a s below in my mail) > 4) When I bridge A and B, A and B canNOT hear each other (e.g. for 1/2 > sec). A continues to hear its messages, B does not hear anything > 5) When I hangup B then A is still active and does not recognize > hangup > of B. > > At former times, when the call was bridged, I had an "unbind" for each > call on the event_socket interface, so event_socket was out of the > loop > after uuid_bridge. Now I have an "unbind" for each party only at the > time when the party hangs up. > > Best regards > Peter > > > Anthony Minessale schrieb: > > update one more time and see how that is > > > > > > On Tue, Jan 6, 2009 at 12:25 PM, Peter P GMX > > > >> > wrote: > > > > One more info: > > I have updated to the newest SVN version of FS. > > A and B can actually hear each other (just a bit, some > scratching) > > while > > the announcement to A is very slow (~50% speed) and very choppy. > > > > Best regards > > peter > > > > Peter P GMX schrieb: > > > I have setup a test machine and a production machine. Since > > recently the > > > production machine behaves differently in terms of uuid > bridge. > > > > > > How it should work (and how it worked before) > > > 1) call A comes in > > > 2) I play some messages to A > > > 3) In the meantime I originate a call to B and transfer > to an > > > extension, where also some messages are played > > > 4) Then I bridge A and B, so they are dropped off the > current > > > announcements an speak to each other > > > 5) when either A or B hangs up, both legs are terminated > > > > > > New behaviour > > > 1) call A comes in > > > 2) I play some messages to A > > > 3) In the meantime I originate a call to B and transfer > to an > > > extension, where also some messages are played > > > 4) Then I bridge A and B, A and B can hear each other > for 1/2 sec, > > > then A constinues to hear its messages, B does not hear > anything > > > 5) when either A or B hangs up, both legs are terminated > > > > > > 4) is different now! > > > > > > The FS console show some messages about unbridge > > > (SWITCH_MESSAGE_INDICATE_UNBRIDGE), but somehow the calls > are still > > > connected until A or B hangs up. > > > > > > Anybody has a clue? > > > Best regards > > > Peter > > > > > > 3) is finished, 4) starts > > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:352 > zap_isdn_931_34() zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:35:31 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:35:34 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1094 > > > switch_ivr_uuid_bridge() (OpenZAP/2:3/49171xxxxxxx) State > Change > > > CS_EXECUTE -> CS_RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1095 > > > switch_ivr_uuid_bridge() (OpenZAP/2:1/216xxxxx) State Change > > CS_EXECUTE > > > -> CS_RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State EXECUTE > > going to > > > sleep > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State > > Change > > > CS_RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET > > > 4) Start > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > > > uuid_bridge_on_reset() OpenZAP/2:1/216xxxxx CUSTOM RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:53 > > > switch_core_standard_on_reset() OpenZAP/2:1/216xxxxx > Standard RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET > > going to sleep > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > > EXECUTE going > > > to sleep > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) > Running State > > > Change CS_RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > > > uuid_bridge_on_reset() OpenZAP/2:3/49171xxxxxxx CUSTOM RESET > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:533 > > > uuid_bridge_on_reset() (OpenZAP/2:3/49171xxxxxxx) State Change > > CS_RESET > > > -> CS_SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > RESET > > going > > > to sleep > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) > Running State > > > Change CS_SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > > SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > > channel_on_soft_execute() > > > CHANNEL SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > > > uuid_bridge_on_soft_execute() OpenZAP/2:3/49171xxxxxxx CUSTOM > > TRANSMIT > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:571 > > > uuid_bridge_on_soft_execute() (OpenZAP/2:1/216xxxxx) State > Change > > > CS_RESET -> CS_SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State > > Change > > > CS_SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > > channel_on_soft_execute() > > > CHANNEL SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > > > uuid_bridge_on_soft_execute() OpenZAP/2:1/216xxxxx CUSTOM > TRANSMIT > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:181 > > > switch_core_standard_on_soft_execute() > OpenZAP/2:1/216xxxxx Standard > > > SOFT_EXECUTE > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > SOFT_EXECUTE > > > going to sleep > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:814 > > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) > State > > Change > > > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:856 > > > switch_ivr_multi_threaded_bridge() OpenZAP/2:1/216xxxxx > receive > > message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:863 > > > switch_ivr_multi_threaded_bridge() > OpenZAP/2:3/49171xxxxxxx receive > > > message [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:907 > > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) > State > > Change > > > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State > > Change > > > CS_EXCHANGE_MEDIA > > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:436 > > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > > EXCHANGE_MEDIA > > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:511 > > > channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA > > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > > switch_ivr_parse_event() > > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > > custom/warteschleife_30.wav interrupt_digit 0 ) > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_play_say.c:932 > > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > channels 20ms > > > 2009-01-06 17:35:36 [DEBUG] switch_core_io.c:610 > > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > > message > > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > > switch_ivr_parse_event() > > > OpenZAP/2:3/49171xxxxxxx Command Execute stop_dtmf(true) > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:229 > > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:231 > > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:229 > > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:231 > > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 31 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7d 02 01 22 30 0d a1 > 05 30 > > 03 02 01 > > > 08 82 01 00 83 01 00] > > > > > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:352 > zap_isdn_931_34() zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:35:46 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 31 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7e 02 01 22 30 0d a1 > 05 30 > > 03 02 01 > > > 09 82 01 00 83 01 00] > > > > > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:352 > zap_isdn_931_34() zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:36:02 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:36:06 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:229 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:36:07 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:231 > > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx > [BREAK] > > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:08 [DEBUG] switch_ivr.c:391 > > switch_ivr_parse_event() > > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > > custom/warteschleife_30.wav interrupt_digit 0 ) > > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_play_say.c:932 > > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > channels 20ms > > > 2009-01-06 17:36:08 [DEBUG] switch_core_io.c:610 > > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > > message > > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 31 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7f 02 01 22 30 0d a1 > 05 30 > > 03 02 01 > > > 0a 82 01 00 83 01 00] > > > > > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:352 > zap_isdn_931_34() zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:36:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 32 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 80 02 01 22 30 0d > a1 05 > > 30 03 02 > > > 01 0b 82 01 00 83 01 00] > > > > > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:352 > zap_isdn_931_34() zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:36:33 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:36:38 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:229 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:36:39 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:231 > > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx > [BREAK] > > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:40 [DEBUG] switch_ivr.c:391 > > switch_ivr_parse_event() > > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > > > > > custom/dieser_berater_ist_leider_zur_zeit_nicht_erreichbar_bitte_waehlen_sie_einen_ande.wav > > > interrupt_digit 0 ) > > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_play_say.c:932 > > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > channels 20ms > > > 2009-01-06 17:36:40 [DEBUG] switch_core_io.c:610 > > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > > message > > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:45 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:229 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > > 2009-01-06 17:36:46 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:231 > > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx > [BREAK] > > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:667 > > > switch_core_session_queue_private_event() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_bridge.c:226 > > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:47 [DEBUG] switch_ivr.c:391 > > switch_ivr_parse_event() > > > OpenZAP/2:1/216xxxxx Command Execute read(0 4 > > > > > > custom/geben_sie_bitte_nach_dem_signalton_die_kennung_des_gewuenschten_beraters_ein_sch2.wav > > > dtmfdtmf 10000 #,*) > > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_play_say.c:932 > > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > channels 20ms > > > 2009-01-06 17:36:47 [DEBUG] switch_core_io.c:610 > > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive > > message > > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > > > switch_core_session_perform_receive_message() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 32 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 81 02 01 22 30 0d > a1 05 > > 30 03 02 > > > 01 0c 82 01 00 83 01 00] > > > > > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:352 > zap_isdn_931_34() zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:36:48 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:37:03 [DEBUG] switch_ivr_play_say.c:1222 > > > switch_ivr_play_file() done playing file > > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 32 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 82 02 01 22 30 0d > a1 05 > > 30 03 02 > > > 01 0d 82 01 00 83 01 00] > > > > > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:352 > zap_isdn_931_34() zchan > > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > > 2009-01-06 17:37:04 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() > > Received > > > unhandled message 98 (0x62) > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() > > READ 13 > > > > > > -------------------------------------------------------------------------------- > > > [08 02 00 35 45 08 02 80 90 1e 02 82 88] > > > > > > 5) Hangup > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() > > Yay I got > > > an event! Type:[45] Size:[115] CRV: 53 (0x35, CTX: Originator) > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:352 > zap_isdn_931_34() zchan > > > 77df70 (2:1) source isdn_data->channels_remote_crv[0x35] > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:455 zap_isdn_931_34() > > Changing > > > state on 2:1 from UP to TERMINATING > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:813 > state_advance() 2:1 > > STATE > > > [TERMINATING] > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1121 state_advance() > > > Terminating: Direction Inbound > > > 2009-01-06 17:37:12 [DEBUG] mod_openzap.c:1418 > > on_clear_channel_signal() > > > got clear channel sig [STOP] > > > 2009-01-06 17:37:12 [NOTICE] mod_openzap.c:1437 > > > on_clear_channel_signal() Hangup OpenZAP/2:1/216xxxxx > > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > > 2009-01-06 17:37:12 [DEBUG] switch_channel.c:1441 > > > switch_channel_perform_hangup() Send signal > OpenZAP/2:1/216xxxxx > > [KILL] > > > 2009-01-06 17:37:12 [DEBUG] switch_core_session.c:779 > > > switch_core_session_signal_state_change() Send signal > > > OpenZAP/2:1/216xxxxx [BREAK] > > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() > WRITE 5 > > > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 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 > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kristjan.ugrin at gmail.com Thu Jan 8 02:58:33 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Thu, 08 Jan 2009 11:58:33 +0100 Subject: [Freeswitch-users] firewall and nat In-Reply-To: <49649481.7000804@gmx.net> References: <49649481.7000804@gmx.net> Message-ID: Thanks for all suggestions. Ufortunately I cannot get it working. Seems like packets are not coming to phone behind nat (freeswitch is on public ip). When registering I can see multiple notify retries like this: send 802 bytes to udp/[10.99.10.6]:5060 at 10:49:31.762605: ------------------------------------------------------------------------ NOTIFY sip:1003 at 10.99.10.6;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 212.235.180.41:5080;rport;branch=z9hG4bKtNStS2gtr8DNr Max-Forwards: 70 From: ;tag=veSr4DmgmFHjr To: Call-ID: cec2b00b-5814-122c-f981-000fea488302 CSeq: 109587536 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10924M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, refer Subscription-State: terminated;timeout Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: yes Message-Account: sip:1003 at 212.235.180.41 Voice-Message: 3/0 (0/0) I've opened necessary ports and I've defined custom rtp port range (which goes trough). Does nat should really just work if you register on external profile via port 5080? This is what I'm doing now. The phone on lan is a nokia N95 configured like described here (using port 5080): http://wiki.freeswitch.org/wiki/Nokia_N95 Phone shows registered message, but it takes like a half minute to register, when I'm on home network this happens in a second. On Wed, 07 Jan 2009 12:39:45 +0100, Peter P GMX wrote: > Generally speaking you will need to open an UPD port range for the RTP > stream. This can be configured on FS. Eg. we use 12000-13000 on our > system. > Then If you do not hear any sound you may put > > > > in your external and internal profile, if FS is natted. > > Best regards > Peter > > > kriko schrieb: >> Hello! >> >> Yesterday I've successfully placed a call between two different domains: >> originate sofia/default/1003 at 10.99.8.221 >> &bridge(sofia/gateway/212.235.180.41/1001) >> >> I didn't hear any audio, but it was kinda working. Today I investigated >> this more deep and found some issues. >> FS with 212.235.180.41 is a public computer with firewall, but open TCP >> and UDP 5060, 5080 ports. Freeswitch on this machine >> uses default configuration. >> >> FS with 10.99.8.221 is a lan computer in a different place, this is >> where I would like to start a call, the other way would >> be probably too much difficult for now. I've added a gateway entry to >> this one: >> http://pastebin.com/m2174ead >> >> Calling from 10.99.8.221 (for e.g. using softphone at ext. 1003) to >> 212.235.180.41 (ext. 1001 for e.g.) works. Both end >> answers, however I cannot hear audio coming trough. When testing I'm at >> the computer which is behind a lan, so I'm >> capturing music as audio source on the other side. >> >> Are there any other ports I should open on public computer? >> With wireshark on the computer behind a lan, I can see RTP going away >> to 212.235.180.41, but not the other way. >> >> There are also issues when e.g. terminating a call on public computer, >> fs on the other end will never terminate the call since >> SIP messages cannot reach the computer behind lan I guess, but this is >> second problem. >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- kriko From helmut.kuper at ewetel.de Thu Jan 8 03:46:12 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 08 Jan 2009 12:46:12 +0100 Subject: [Freeswitch-users] Hint for DTMF handling in sofia.c Message-ID: <4965E784.5030803@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today we found a DTMF problem in an older FS build. We weren't able to pass "*" and "#" to PSTN. FS sents those chars as "0". I fixed that on my own. After that I searched in build 11055M to if it was already fixed in trunk. I found it was fixed in parts. I found that in INFO messages of application type "dtmf-relay" the rfc2833 DTMF events "*#ABCD" are now handled correct, but "EF" are still ignored. In INFO messages of application type "dtmf" EVENTS 0-9 are considered. The others are still ignored resp. changed to "0". So I would recommend to replace the code beginning in line 3535 <<<<<<<<<<<<<<<<<< if (*signal_ptr && (*signal_ptr == '*' || *signal_ptr == '#' || *signal_ptr == 'A' || *signal_ptr == 'B' || *signal_ptr == 'C' || *signal_ptr == 'D')) { dtmf.digit = *signal_ptr; } else { tmp = atoi(signal_ptr); dtmf.digit = switch_rfc2833_to_char(tmp); } <<<<<<<<<<<<<<<<<< with this two lines: >>>>>>>>>>>>>> tmp=switch_char_to_rfc2833(*signal_ptr); dtmf.digit = switch_rfc2833_to_char(tmp); >>>>>>>>>>>>>> This will handle all rfc2833 dtmf events in dtmf-relay INFO messages correctly. This is tested by me with snom and AVM clients. Further replace the line 3556 <<<<<<<<<<<<<< int tmp = atoi(sip->sip_payload->pl_data); <<<<<<<<<<<<<< with this line: >>>>>>>>>>>>>> int tmp=switch_char_to_rfc2833(*(sip->sip_payload->pl_data)); >>>>>>>>>>>>>> That should handle all rfc2833 DTMF-Events in dtmf INFO messages. This last patch isn't tested by me, cause I've no phone which genereates INFO messages of type applicatin/dtmf... best regards and a late "happy new year" to all of u! Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkll54QACgkQ4tZeNddg3dxJgwCfULx8X9hw6xDd1q8r/Iih5unx c6EAn1T/7GK56pLLrMi140x50HLnO6i1 =3ENR -----END PGP SIGNATURE----- From kristjan.ugrin at gmail.com Thu Jan 8 04:32:13 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Thu, 08 Jan 2009 13:32:13 +0100 Subject: [Freeswitch-users] Understandig context Message-ID: Hello! I'm a bit confused how sofia profiles works. If I did understood correctly it is something like this: the param inside a external sofia profile will force to process calls via extern context (e.g. dialplan/extern.xml). Then if I use inside my jingle profile, will force calls from the specific gtalk account to be processed via gtalk context. This works fine. But when examining both internal and external default sofia profiles, they have: However when making a call between two local registered sip phones it seems to use other dialplan: 2009-01-08 13:26:19 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1003->1001 in context default Why it uses default instead of public context? And if you have one local (5060) and one external (5080) sip phone, when you make a call between them, which dialplan should be executed in this case? -- kriko From vhatz at kinetix.gr Thu Jan 8 05:01:37 2009 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Thu, 08 Jan 2009 15:01:37 +0200 Subject: [Freeswitch-users] mod_opal calls and records In-Reply-To: <191c3a030901071114g194ae214o9ee6e78f08aa0d6c@mail.gmail.com> References: <4F14448B-6DD4-4B05-96A5-22289C8E5093@wcgltd.com> <4964F0F1.6000803@kinetix.gr> <87f2f3b90901071030j2406e448qa291ebde435d4390@mail.gmail.com> <4964F923.3060100@kinetix.gr> <191c3a030901071114g194ae214o9ee6e78f08aa0d6c@mail.gmail.com> Message-ID: <4965F931.8010609@kinetix.gr> Hello Antony, I remember the fund raiser, but I didn't know that a mod_opal was available for testing some time know. Perhaps I missed the announcement I guess, this is why I thought that no work was done on it. I'll proceed to testing it asap. Best regards, Vlasis Hatzistavrou. Anthony Minessale wrote: > yes, > > The author of opal himself has put a lot of effort into making this > possible. > I announced it several times and asked people to test it and nobody > seemed to notice. > We had a fundraiser and only a small handful of people contributed so it > took a long time to get it in tree. > > > On Wed, Jan 7, 2009 at 12:49 PM, Vlasis Hatzistavrou (KTI) > > wrote: > > Hello Michael, > > I went through a quick view of the files of mod_opal. Is it a > continuation of Tuyan Ozipek's work on mod_opal? I was under the > impression that that mod_opal was to be discarded in order to be > replaced by a new one... > > Best regards, > Vlasis Hatzistavrou. > > Michael Collins wrote: > > Please look very carefully at Tony's comments in the commit: > > http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=10734 > > > > ;) > > -MC > > > > On Wed, Jan 7, 2009 at 10:14 AM, Vlasis Hatzistavrou (KTI) > > > >> wrote: > > > > Hello, > > > > Josh Forman wrote: > > > I started to do some testing with h323 calls using > mod_opal and I > > have > > > been having a number of issues. > > > > > > This is interesting, as I wasn't aware that there was a > working mod_opal > > with FS, yet. > > > > Which FS version are you using? > > > > Best regards, > > Vlasis Hatzistavrou > > Kinetix Tele.com International Inc. > > 306 Victoria House, > > Victoria, Mahe, > > Seychelles > > Tel.: +302310556134 > > Fax: +302310556134 (ext. 0) > > GSM: +306977835653 > > e-mail: vhatz at kinetixtele.com > > > > http://www.kinetixtele.com > > > > Postal address: > > Monastiriou 9 & Enotikon > > 54627 > > Thessaloniki > > Greece > > > > > > > > I've been creating calls that come into freeswitch as SIP > and output > > > as h323 and while I have a call open I did not see any > entries from > > > fs_cli when using the "show channels" or "show calls" > commands. Also > > > no CDR is generated from the call. > > > > > > I was wondering if someone could tell me if these issues > are from > > > something that just has not been implemented yet in > mod_opal or if > > > this is most likely a configuration issue on my side. > > > > > > Thank you, > > > > > > Josh > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at 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 at 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 at 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 at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kristjan.ugrin at gmail.com Thu Jan 8 05:32:08 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Thu, 08 Jan 2009 14:32:08 +0100 Subject: [Freeswitch-users] [ringback] problems with dingaling In-Reply-To: References: Message-ID: Now I've made a small dialplan to call from sip phone directly to gtalk: Simple, calling works. However still can't get ringback to work. In this case the first leg is not yet aswered. If I apply same stuff onto SIP to SIP call then ringback works. Dingaling problem? Log: http://pastebin.com/m37354677 This is all that 2009-01-08 14:25:56 [NOTICE] mod_dingaling.c:1110 negotiate_media() Ring-Ready dingaling/gmail.com/atomic.devterium at gmail.com! 2009-01-08 14:25:56 [DEBUG] mod_dingaling.c:1058 do_describe() Don't have my codec yet here's one -- kriko From erik at erikwickstrom.com Wed Jan 7 16:09:17 2009 From: erik at erikwickstrom.com (Erik Wickstrom) Date: Wed, 7 Jan 2009 16:09:17 -0800 Subject: [Freeswitch-users] Trouble getting session.setInputCallback working. In-Reply-To: <3d381e170901071603m1943287fwba1ef838678117bb@mail.gmail.com> References: <3d381e170901071603m1943287fwba1ef838678117bb@mail.gmail.com> Message-ID: <3d381e170901071609h3cbefccga01778685952cd60@mail.gmail.com> Hi all, I'm trying to get a setInputCallback function working with mod_python. I'm using a current svn checkout for my build and the hello world via call example from the wiki ( http://wiki.freeswitch.org/wiki/Mod_python#Hello_World_via_call ). I've tried repeatedly, but I can't get the callback function to execute. I keep pressing various digits on my phone, but none of them are being logged to the console. My understanding is that this callback is supposed to fire when a dtmf (123456789*#, right??) and also acts as an event handler for things like mod_vmd. Is this part of the wiki example working for anyone else? Any idea what I might be doing wrong? Thanks! Erik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090107/eb67c486/attachment-0001.html From anthony.minessale at gmail.com Thu Jan 8 06:05:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Jan 2009 08:05:39 -0600 Subject: [Freeswitch-users] polycom one-way audio problem In-Reply-To: <4965A261.9030708@matthew.at> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> <4964F893.6020302@matthew.at> <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> <4964FBCE.8090008@matthew.at> <49650637.7040703@matthew.at> <496588FE.60409@matthew.at> <4965A261.9030708@matthew.at> Message-ID: <191c3a030901080605p72904873yd7f107e8fa336aed@mail.gmail.com> Did you ever find out if the rtp was making it to your phone? Did you get around to testing the echo exten? That is the most basic call you can do it is 1 leg call just playing your own audio back. Also 9998 plays the tetris song with the tone generator. We for sure can see rtp packets in the pcap bound for your phone. This is a very unique problem as many people get this basic situation working daily so it must be a network issue of some sort. Can we rule out your network where the phones live by testing some phones on the same network as FS? Do you have a hub you could put the phones on so you can packet sniff the traffic to them? On Thu, Jan 8, 2009 at 12:51 AM, Matthew Kaufman wrote: > Matthew Kaufman wrote: > > Matthew Kaufman wrote: > > > >> I have upgraded to the latest boot roms and SIP firmware, 3.1.1.0137. I > >> get the same one-way audio between the polycom and x-lite, and now on a > >> polycom-polycom call I get no audio in *either* direction. (Not much of > >> an improvement, but different) > >> > >> > > For those following on the list, a successful workaround is to set > > "inbound-proxy-media" to true. Why that should be necessary, and why it > > behaves the way it does when that is set to false (the strangest part > > being that calls that go directly to VM have good audio, but calls that > > ring the far end for a time and then go to VM have no audio *even* when > > they've gone over to VM), I still don't understand. > > > > > > I spoke too soon. If I turn on "inbound-proxy-media", then it goes back > to "called party can hear calling party, but calling party calling party > cannot hear called party" (the same as it was before upgrading to the > latest Polycom firmware), and additionally the calling party now gets > ringback (didn't before), but if the called party doesn't answer instead > of dropping to voicemail it goes to fast busy. > > The last is probably related to: "[ERR] sofia_glue.c:1608 > sofia_glue_tech_set_codec() No audio codec available" which then fires > INCOMPATIBLE_DESTINATION. > > I also picked up version 11089 *and* threw out all the conf directory > and regenerated it from the sample source, so this is 100% today-build, > default-settings (except for the adjustments to inbound-proxy-media). > > Matthew Kaufman > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/8ba45a8b/attachment.html From anthony.minessale at gmail.com Thu Jan 8 06:07:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Jan 2009 08:07:04 -0600 Subject: [Freeswitch-users] Understandig context In-Reply-To: References: Message-ID: <191c3a030901080607n16ddee98h9e50623dcf4d6db2@mail.gmail.com> authenticated users can override the context setting with variables in their account or domain. On Thu, Jan 8, 2009 at 6:32 AM, kriko wrote: > Hello! > > I'm a bit confused how sofia profiles works. > If I did understood correctly it is something like this: > the param inside a external sofia > profile will force > to process calls via extern context (e.g. dialplan/extern.xml). > > Then if I use inside my jingle > profile, will force calls > from the specific gtalk account to be processed via gtalk context. > This works fine. > > But when examining both internal and external default sofia profiles, they > have: > > > However when making a call between two local registered sip phones it seems > to use other dialplan: > 2009-01-08 13:26:19 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 1003->1001 in context default > > Why it uses default instead of public context? > > And if you have one local (5060) and one external (5080) sip phone, when > you make a call between them, > which dialplan should be executed in this case? > > -- > kriko > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/99254717/attachment.html From mike at jerris.com Thu Jan 8 06:09:23 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 8 Jan 2009 09:09:23 -0500 Subject: [Freeswitch-users] Hint for DTMF handling in sofia.c In-Reply-To: <4965E784.5030803@ewetel.de> References: <4965E784.5030803@ewetel.de> Message-ID: This workaround was added to address phones that specifically send info dtmf incorrectly. Do you have a specific device that is not working with 1.0.2? If so, can you please show the exact packet it is sending. Mike On Jan 8, 2009, at 6:46 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > today we found a DTMF problem in an older FS build. We weren't able to > pass "*" and "#" to PSTN. FS sents those chars as "0". I fixed that on > my own. After that I searched in build 11055M to if it was already > fixed > in trunk. > > I found it was fixed in parts. I found that in INFO messages of > application type "dtmf-relay" the rfc2833 DTMF events "*#ABCD" are > now > handled correct, but "EF" are still ignored. In INFO messages of > application type "dtmf" EVENTS 0-9 are considered. The others are > still > ignored resp. changed to "0". > > > So I would recommend to replace the code beginning in line 3535 > > <<<<<<<<<<<<<<<<<< > if (*signal_ptr && (*signal_ptr == '*' || *signal_ptr == '#' || > *signal_ptr == 'A' || *signal_ptr == 'B' || *signal_ptr == 'C' || > *signal_ptr == 'D')) { > dtmf.digit = > *signal_ptr; > } else { > tmp = atoi(signal_ptr); > dtmf.digit = > switch_rfc2833_to_char(tmp); > } > <<<<<<<<<<<<<<<<<< > > > with this two lines: > > >>>>>>>>>>>>>>> > tmp=switch_char_to_rfc2833(*signal_ptr); > dtmf.digit = switch_rfc2833_to_char(tmp); >>>>>>>>>>>>>>> > > > This will handle all rfc2833 dtmf events in dtmf-relay INFO messages > correctly. This is tested by me with snom and AVM clients. > > > Further replace the line 3556 > > <<<<<<<<<<<<<< > int tmp = atoi(sip->sip_payload->pl_data); > <<<<<<<<<<<<<< > > with this line: >>>>>>>>>>>>>>> > int tmp=switch_char_to_rfc2833(*(sip->sip_payload->pl_data)); >>>>>>>>>>>>>>> > > That should handle all rfc2833 DTMF-Events in dtmf INFO messages. > This last patch isn't tested by me, cause I've no phone which > genereates > INFO messages of type applicatin/dtmf... > > best regards and a late "happy new year" to all of u! > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkll54QACgkQ4tZeNddg3dxJgwCfULx8X9hw6xDd1q8r/Iih5unx > c6EAn1T/7GK56pLLrMi140x50HLnO6i1 > =3ENR > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Jan 8 06:09:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Jan 2009 08:09:41 -0600 Subject: [Freeswitch-users] [ringback] problems with dingaling In-Reply-To: References: Message-ID: <191c3a030901080609p15aa6935i99beb4cf192fe33c@mail.gmail.com> you may want to try jingle has no concept of telephony early media waiting for answer and all that so it's not an exact fit into sip. On Thu, Jan 8, 2009 at 7:32 AM, kriko wrote: > Now I've made a small dialplan to call from sip phone directly to gtalk: > > > > expression="^gmail\+([^\@]+)\@?(.*)$"> > > > > > > > > > > Simple, calling works. However still can't get ringback to work. In this > case the first leg is not yet aswered. > If I apply same stuff onto SIP to SIP call then ringback works. Dingaling > problem? > > Log: > http://pastebin.com/m37354677 > > This is all that > 2009-01-08 14:25:56 [NOTICE] mod_dingaling.c:1110 negotiate_media() > Ring-Ready dingaling/gmail.com/atomic.devterium at gmail.com! > 2009-01-08 14:25:56 [DEBUG] mod_dingaling.c:1058 do_describe() Don't have > my codec yet here's one > > > > > -- > kriko > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/8c1b2b2c/attachment-0001.html From anthony.minessale at gmail.com Thu Jan 8 06:14:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Jan 2009 08:14:32 -0600 Subject: [Freeswitch-users] Export variables from originate command In-Reply-To: <2d9149cd0901072250o4d08be02m1f4a3e6276d750e4@mail.gmail.com> References: <2d9149cd0901072250o4d08be02m1f4a3e6276d750e4@mail.gmail.com> Message-ID: <191c3a030901080614o4f0da4dkd070da9530ad7828@mail.gmail.com> put them in {} comma separated. {foo=bar,test=true}sofia/default/user at dom.com if you are doing forked dial you can set them per leg with [] [var1=foo]sofia/default/user at dom.com,[var1=bar]sofia/default/user2 at dom.com On Thu, Jan 8, 2009 at 12:50 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > I'd like to give some variables to the originate command that pass > (export) to the application I am calling. > > So, if I am calling the transfer app (from originate) to go back > into the dialplan I'd like to use the variables in the dialplan that > were passed to my original originate command (via the transfer > application). Clear as mud, right? ;) > > Is this possible? > > Thanks! > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/84afc1c3/attachment.html From kristjan.ugrin at gmail.com Thu Jan 8 06:14:31 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Thu, 08 Jan 2009 15:14:31 +0100 Subject: [Freeswitch-users] Understandig context In-Reply-To: <191c3a030901080607n16ddee98h9e50623dcf4d6db2@mail.gmail.com> References: <191c3a030901080607n16ddee98h9e50623dcf4d6db2@mail.gmail.com> Message-ID: I see, looking at users inside directory I found why. Thanks On Thu, 08 Jan 2009 15:07:04 +0100, Anthony Minessale wrote: > authenticated users can override the context setting with variables in > their > account or domain. > > > On Thu, Jan 8, 2009 at 6:32 AM, kriko wrote: > >> Hello! >> >> I'm a bit confused how sofia profiles works. >> If I did understood correctly it is something like this: >> the param inside a external sofia >> profile will force >> to process calls via extern context (e.g. dialplan/extern.xml). >> >> Then if I use inside my jingle >> profile, will force calls >> from the specific gtalk account to be processed via gtalk context. >> This works fine. >> >> But when examining both internal and external default sofia profiles, >> they >> have: >> >> >> However when making a call between two local registered sip phones it >> seems >> to use other dialplan: >> 2009-01-08 13:26:19 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing 1003->1001 in context default >> >> Why it uses default instead of public context? >> >> And if you have one local (5060) and one external (5080) sip phone, when >> you make a call between them, >> which dialplan should be executed in this case? >> >> -- >> kriko >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- kriko From kristjan.ugrin at gmail.com Thu Jan 8 06:19:09 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Thu, 08 Jan 2009 15:19:09 +0100 Subject: [Freeswitch-users] [ringback] problems with dingaling In-Reply-To: <191c3a030901080609p15aa6935i99beb4cf192fe33c@mail.gmail.com> References: <191c3a030901080609p15aa6935i99beb4cf192fe33c@mail.gmail.com> Message-ID: Nope. Currently only gtalk ? sip ringback works, sip ? gtalk doesn't. If soemone needs, I'm pasting my extensions used. sip ? gtalk (ringback not working): gtalk ? sip: Thanks for your help! On Thu, 08 Jan 2009 15:09:41 +0100, Anthony Minessale wrote: > you may want to try > > > > jingle has no concept of telephony early media waiting for answer and all > that so it's not an exact fit into sip. > > > On Thu, Jan 8, 2009 at 7:32 AM, kriko wrote: > >> Now I've made a small dialplan to call from sip phone directly to gtalk: >> >> >> >> > expression="^gmail\+([^\@]+)\@?(.*)$"> >> >> >> >> > data="ringback=%(2000,4000,440.0,480.0)"/> >> >> > data="dingaling/gmail.com/$1 at gmail.com >> "/> >> >> >> >> Simple, calling works. However still can't get ringback to work. In this >> case the first leg is not yet aswered. >> If I apply same stuff onto SIP to SIP call then ringback works. >> Dingaling >> problem? >> >> Log: >> http://pastebin.com/m37354677 >> >> This is all that >> 2009-01-08 14:25:56 [NOTICE] mod_dingaling.c:1110 negotiate_media() >> Ring-Ready dingaling/gmail.com/atomic.devterium at gmail.com! >> 2009-01-08 14:25:56 [DEBUG] mod_dingaling.c:1058 do_describe() Don't >> have >> my codec yet here's one >> >> >> >> >> -- >> kriko >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- kriko From andy at fabulous4.co.uk Thu Jan 8 06:37:55 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Thu, 8 Jan 2009 14:37:55 -0000 Subject: [Freeswitch-users] recordFile bitrate Message-ID: <2D43DCC193FC4AE7ACEC8892144B1F94@wsandy> Hi, Is the bitrate, sample rate or format of the audio stream created by session.recordFile configurable at all? Apologies if I've missed something in the docs. cheers Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/57e1c8bd/attachment.html From brian at freeswitch.org Thu Jan 8 06:51:03 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jan 2009 08:51:03 -0600 Subject: [Freeswitch-users] recordFile bitrate In-Reply-To: <2D43DCC193FC4AE7ACEC8892144B1F94@wsandy> References: <2D43DCC193FC4AE7ACEC8892144B1F94@wsandy> Message-ID: bitrate nor sample rate are configurable. The format depends on the extension of the filename. The sample rate is recorded at the channels native rate. /b On Jan 8, 2009, at 8:37 AM, Andy Ayers wrote: > Hi, > > Is the bitrate, sample rate or format of the audio stream created by > session.recordFile configurable at all? Apologies if I've missed > something in the docs. > > cheers > Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/781b66de/attachment.html From helmut.kuper at ewetel.de Thu Jan 8 07:32:26 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 08 Jan 2009 16:32:26 +0100 Subject: [Freeswitch-users] Hint for DTMF handling in sofia.c In-Reply-To: References: <4965E784.5030803@ewetel.de> Message-ID: <49661C8A.5020806@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Mike, I guess the workaround refers to INFO messages with "dtmf" mime type instead of "dtmf-relay"? With the actual trunk (11090) it works. DTMF event F is still converted to zero (0). Im not sure if this event will ever transmitted via INFO. If you use FS switch_utils function "SWITCH_DECLARE(unsigned char) switch_char_to_rfc2833(char key)" instead of the big if-statement (checking for *,#,A,...) befor calling "switch_rfc2833_to_char()" function you can clean up the code a bit and handle all rfc2388 DTMF events. The INFO message my Snom and AVM phones send, looks like this: recv 506 bytes from udp/[85.16.245.220]:1024 at 14:59:40.229120: ------------------------------------------------------------------------ INFO sip:mod_sofia at 85.16.246.6:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 85.16.245.220:1024;branch=z9hG4bK-0a38f681korf;rport From: "HK at FreeSWITCH" ;tag=v0fmpqh4nz To: ;tag=m0NcZ8BDZ8pcD Call-ID: 3c26a924030f-p23y096nom3i CSeq: 3 INFO Max-Forwards: 70 Contact: ;reg-id=1 User-Agent: snom370/7.3.12 Content-Type: application/dtmf-relay Content-Length: 27 Signal=* Duration=160 ------------------------------------------------------------------------ regards Helmut Am 08.01.2009 15:09, schrieb Michael Jerris: > This workaround was added to address phones that specifically send > info dtmf incorrectly. Do you have a specific device that is not > working with 1.0.2? If so, can you please show the exact packet it is > sending. > > Mike -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklmHIoACgkQ4tZeNddg3dzR9gCghxxkOUeYyBEq82BfWsdZ/IWy zMUAnRZ+pf1FmdipqwUZ8YYqbvNscd2q =TzvT -----END PGP SIGNATURE----- From testeador01 at gmail.com Thu Jan 8 07:42:06 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 8 Jan 2009 10:42:06 -0500 Subject: [Freeswitch-users] fs doesn't realize caller hanged up when a call from PSTN is on hold In-Reply-To: <87f2f3b90901071555n6b6d6d11x54c88a3ce920d21b@mail.gmail.com> References: <87f2f3b90901071555n6b6d6d11x54c88a3ce920d21b@mail.gmail.com> Message-ID: I am using fs 1.0.2 running on CentOS 5.2. You're right, the gxw is not alerting fs at all, i did some testing and there is absolutely nothing on the debug console (after pressing F8) between the moment i put the user on hold and the moment i pick up the call again. "You might need to use tone_detect. See the example at the bottom (this is how I use it to detect a hang up as my telco doesn't vary the voltage levels on hangup)": It recognizes when the hang up is done before picking up, but after picking up it won't recognize the hang up. I tried configuring this on my dialplan configuration to-pstn then i dialed to an extension that would pick up and then hang up on me, but it didn't do the magic, I also tried adding it to the anti-action options on the default dialplan for the extensions 100X and testing again but it didn't work either, I tried calling to my telco to get info on what kind of hangup signal did they send but i got stuck into a cycle of a person bouncing me to the "assigned area" over and over again. Maybe the frequences i tried were not the right ones... maybe the gateway can be configured without having information about my telco... i don't know, i'll try asking for support on the Grandstream site. Thank you very much anyways. 2009/1/7 Michael Collins > What version of FreeSWITCH are you on? Also, can you get a debug trace from > when the far end hangs up? We definitely need to know if the gxw is alerting > FS to the fact that the far end actually hung up, which is in doubt right > now. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/03e79ee4/attachment-0001.html From brian at freeswitch.org Thu Jan 8 07:43:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jan 2009 09:43:06 -0600 Subject: [Freeswitch-users] Hint for DTMF handling in sofia.c In-Reply-To: <49661C8A.5020806@ewetel.de> References: <4965E784.5030803@ewetel.de> <49661C8A.5020806@ewetel.de> Message-ID: I just opened a bug with Snom on this one.. it should NEVER send a * or # in the signal line. But we do work around this but its wrong. /b On Jan 8, 2009, at 9:32 AM, Helmut Kuper wrote: > INFO sip:mod_sofia at 85.16.246.6:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP > 85.16.245.220:1024;branch=z9hG4bK-0a38f681korf;rport > From: "HK at FreeSWITCH" ;tag=v0fmpqh4nz > To: ;tag=m0NcZ8BDZ8pcD > Call-ID: 3c26a924030f-p23y096nom3i > CSeq: 3 INFO > Max-Forwards: 70 > Contact: ;reg-id=1 > User-Agent: snom370/7.3.12 > Content-Type: application/dtmf-relay > Content-Length: 27 > > Signal=* > Duration=160 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/8c77a8b8/attachment.html From kristjan.ugrin at gmail.com Thu Jan 8 08:02:11 2009 From: kristjan.ugrin at gmail.com (kriko) Date: Thu, 08 Jan 2009 17:02:11 +0100 Subject: [Freeswitch-users] originate and caller number Message-ID: I wrote a java socket client that originates a call, for e.g.: originate dingaling/gmail.com/atomic.devterium at gmail.com &bridge(loopback/1003/java_gmail_bridge) This works fine, however both ends doesn't really see each other numbers, instead they see freeswitch number and id. Is it possible to show every user the opposite caller number? dialplan used: http://pastebin.com/m71525ac6 -- kriko From regs at kinetix.gr Thu Jan 8 08:06:52 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 08 Jan 2009 18:06:52 +0200 Subject: [Freeswitch-users] mod_opal compile error Message-ID: <4966249C.40801@kinetix.gr> Hi, I have installed ptlib 2.4.3 and opal 3.4.3 and still cannot get freeswitch to compile mod_opal. The error I am getting is this : mod_opal.h:58: error: expected class-name before ?{? token I am using the 11094 revision of FS on CentOS 5.2. Has anyone faced anything similar? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From brian at freeswitch.org Thu Jan 8 08:09:20 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jan 2009 10:09:20 -0600 Subject: [Freeswitch-users] mod_opal compile error In-Reply-To: <4966249C.40801@kinetix.gr> References: <4966249C.40801@kinetix.gr> Message-ID: You have to use SVN of both opal and ptlib. Its bleeding edge. /b On Jan 8, 2009, at 10:06 AM, Apostolos Pantsiopoulos wrote: > Hi, > > I have installed ptlib 2.4.3 and opal 3.4.3 and still cannot get > freeswitch to > compile mod_opal. The error I am getting is this : > > mod_opal.h:58: error: expected class-name before ?{? token > > I am using the 11094 revision of FS on CentOS 5.2. > > Has anyone faced anything similar? From andy at fabulous4.co.uk Thu Jan 8 08:15:27 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Thu, 8 Jan 2009 16:15:27 -0000 Subject: [Freeswitch-users] recordFile bitrate In-Reply-To: Message-ID: <58AF962434D94B9E83969C745DFDCD99@wsandy> Thanks Brian, am I correct in saying therefore that all mp3 streams generated by the recordFile command(with mod_shout installed) will be 64Kbps? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 08 January 2009 14:51 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] recordFile bitrate bitrate nor sample rate are configurable. The format depends on the extension of the filename. The sample rate is recorded at the channels native rate. /b On Jan 8, 2009, at 8:37 AM, Andy Ayers wrote: Hi, Is the bitrate, sample rate or format of the audio stream created by session.recordFile configurable at all? Apologies if I've missed something in the docs. cheers Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/0431a4d2/attachment.html From brian at freeswitch.org Thu Jan 8 08:18:10 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jan 2009 10:18:10 -0600 Subject: [Freeswitch-users] recordFile bitrate In-Reply-To: <58AF962434D94B9E83969C745DFDCD99@wsandy> References: <58AF962434D94B9E83969C745DFDCD99@wsandy> Message-ID: <01586041-736C-4091-9A42-B4A7B4CA287A@freeswitch.org> And if its stereo it will be a bit bigger. Also I don't recommend recording in mp3 at all if you want to scale far. /b On Jan 8, 2009, at 10:15 AM, Andy Ayers wrote: > Thanks Brian, am I correct in saying therefore that all mp3 streams > generated by the recordFile command(with mod_shout installed) will > be 64Kbps? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/8f32c2c1/attachment.html From helmut.kuper at ewetel.de Thu Jan 8 08:24:57 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 08 Jan 2009 17:24:57 +0100 Subject: [Freeswitch-users] Hint for DTMF handling in sofia.c In-Reply-To: References: <4965E784.5030803@ewetel.de> <49661C8A.5020806@ewetel.de> Message-ID: <496628D9.5050805@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Brian, sorry, got lost ... It's a bug in Snom and AVM's phones? Maybe we misunderstood. RFC 2833 emtioned in chapter 3.10 following DTMF Events: 3.10 DTMF Events Table 1 summarizes the DTMF-related named events within the telephone-event payload format. Event encoding (decimal) _________________________ 0--9 0--9 * 10 # 11 A--D 12--15 Flash 16 Table 1: DTMF named events Most phones in germany are able to send * and # DTMF events and Phone-applications like voiceboxes or IVRs need those Events. Therefor I don't think it's a bug in Snom or AVM ... But maybe you mean Snom and AVM should send "Signal=10" instead of "Signal=*" ? Unfortunately I can't find an rfc which describes DTMF over INFO messages to find out what the right way is. regards helmut Am 08.01.2009 16:43, schrieb Brian West: > I just opened a bug with Snom on this one.. it should NEVER send a * or > # in the signal line. But we do work around this but its wrong. > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklmKNkACgkQ4tZeNddg3dwS5ACgog04UzepKkLgY6RDbzmUoAGy OYkAn3ZiaQeHXNri58+DpNv1GgtPJvoD =kcrv -----END PGP SIGNATURE----- From edwardlobo at gmail.com Thu Jan 8 07:32:16 2009 From: edwardlobo at gmail.com (bahbie) Date: Thu, 8 Jan 2009 07:32:16 -0800 (PST) Subject: [Freeswitch-users] external users -> gateway *help please* Message-ID: <21351221.post@talk.nabble.com> I have set up an external phone per this example http://wiki.freeswitch.org/wiki/Example_Offsite_phones http://wiki.freeswitch.org/wiki/Example_Offsite_phones I have set up a sip gateway that works with internal phones but does not work when dialed from external phones. What am I missing. I would be very appreciative of any help as I have been struggling with this for the past 2 days and I have searched a lot on the forum and www but to no avail. I think this is not a NAT or firewall issue as I can call to internal phones and listen to voice mails on this phone. I have pasted the log with f8 below: 2009-01-08 12:42:16 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/doublenat/2001 at 89.99.119.159:5090 [c8069e9a-dd81-11dd-84ff-d70bc5a4d2b0] 2009-01-08 12:42:16 [DEBUG] sofia.c:4311 sofia_handle_sip_i_invite() Setting NAT mode based on via received 2009-01-08 12:42:16 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/doublenat/2001 at 89.99.119.159:5090 entering state [received] 2009-01-08 12:42:16 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 4 2 IN IP4 80.97.216.202 s=CounterPath X-Lite 3.0 c=IN IP4 80.97.216.202 t=0 0 m=audio 32930 RTP/AVP 107 119 100 106 0 105 98 8 101 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 2 : lXwDfm0K ICpugw02 10.0.0.56 32930 a=alt:2 1 : J1ZGgcJa jhlsNlM+ 80.97.216.202 32930 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [BV32:107:16000]/[PCMU:0:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [BV32:107:16000]/[PCMA:8:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [BV32:107:16000]/[GSM:3:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [BV32-FEC:119:16000]/[PCMU:0:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [BV32-FEC:119:16000]/[PCMA:8:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [BV32-FEC:119:16000]/[GSM:3:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:100:16000]/[PCMU:0:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:100:16000]/[PCMA:8:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX:100:16000]/[GSM:3:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX-FEC:106:16000]/[PCMU:0:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX-FEC:106:16000]/[PCMA:8:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [SPEEX-FEC:106:16000]/[GSM:3:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:1670 sofia_glue_tech_set_codec() Set Codec sofia/doublenat/2001 at 89.99.119.159:5090 PCMU/8000 20 ms 160 samples 2009-01-08 12:42:16 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-01-08 12:42:16 [DEBUG] sofia.c:2698 sofia_handle_sip_i_state() (sofia/doublenat/2001 at 89.99.119.159:5090) State Change CS_NEW -> CS_INIT 2009-01-08 12:42:16 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/doublenat/2001 at 89.99.119.159:5090 [BREAK] 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) Running State Change CS_INIT 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) State INIT 2009-01-08 12:42:16 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/doublenat/2001 at 89.99.119.159:5090 SOFIA INIT 2009-01-08 12:42:16 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/doublenat/2001 at 89.99.119.159:5090) State Change CS_INIT -> CS_ROUTING 2009-01-08 12:42:16 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/doublenat/2001 at 89.99.119.159:5090 [BREAK] 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) State INIT going to sleep 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) Running State Change CS_ROUTING 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) State ROUTING 2009-01-08 12:42:16 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/doublenat/2001 at 89.99.119.159:5090 SOFIA ROUTING 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_routing() sofia/doublenat/2001 at 89.99.119.159:5090 Standard ROUTING 2009-01-08 12:42:16 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 2001->00447825299999 in context public 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [public_extensions] destination_number(00447825299999) =~ /^(20[01][0-9])$/ 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [unloop] true() =~ /^true$/ 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [call_debug] false() =~ /^true$/ 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [set_domain] ${domain_name}() =~ /^$/ 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [set_domain] source(mod_sofia) =~ /mod_sofia/ 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [set_domain] ${sip_req_params}() =~ /domain_name=([A-Z-a-z0-9.]+)/ 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [public_extensions] destination_number(00447825299999) =~ /^(10[01][0-9])$/ 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [public_did] destination_number(00447825299999) =~ /^(5551212)$/ 2009-01-08 12:42:16 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:100 switch_core_standard_on_routing() (sofia/doublenat/2001 at 89.99.119.159:5090) State Change CS_ROUTING -> CS_EXECUTE 2009-01-08 12:42:16 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/doublenat/2001 at 89.99.119.159:5090 [BREAK] 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) State ROUTING going to sleep 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) Running State Change CS_EXECUTE 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) State EXECUTE 2009-01-08 12:42:16 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/doublenat/2001 at 89.99.119.159:5090 SOFIA EXECUTE 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:137 switch_core_standard_on_execute() sofia/doublenat/2001 at 89.99.119.159:5090 Standard EXECUTE 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/doublenat/2001 at 89.99.119.159:5090 Execute set(outside_call=true) 2009-01-08 12:42:16 [DEBUG] mod_dptools.c:699 set_function() sofia/doublenat/2001 at 89.99.119.159:5090 SET [outside_call]=[true] 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/doublenat/2001 at 89.99.119.159:5090 Execute export(domain_name=${sip_req_host}) 2009-01-08 12:42:16 [DEBUG] switch_core_session.c:1254 switch_core_session_execute_application() sofia/doublenat/2001 at 89.99.119.159:5090 Expanded String export(domain_name=89.99.119.159) 2009-01-08 12:42:16 [DEBUG] mod_dptools.c:837 export_function() EXPORT [domain_name]=[89.99.119.159] 2009-01-08 12:42:16 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/doublenat/2001 at 89.99.119.159:5090 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-08 12:42:16 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/doublenat/2001 at 89.99.119.159:5090 [KILL] 2009-01-08 12:42:16 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/doublenat/2001 at 89.99.119.159:5090 [BREAK] 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) State EXECUTE going to sleep 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) Running State Change CS_HANGUP 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) State HANGUP 2009-01-08 12:42:16 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/doublenat/2001 at 89.99.119.159:5090 hanging up, cause: NORMAL_CLEARING 2009-01-08 12:42:16 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 480 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/doublenat/2001 at 89.99.119.159:5090 Standard HANGUP, cause: NORMAL_CLEARING 2009-01-08 12:42:16 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/doublenat/2001 at 89.99.119.159:5090) State HANGUP going to sleep 2009-01-08 12:42:16 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 3 (sofia/doublenat/2001 at 89.99.119.159:5090) Locked, Waiting on external entities 2009-01-08 12:42:16 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 3 (sofia/doublenat/2001 at 89.99.119.159:5090) Ended 2009-01-08 12:42:16 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/doublenat/2001 at 89.99.119.159:5090 [CS_HANGUP] -- View this message in context: http://www.nabble.com/external-users--%3E-gateway-*help-please*-tp21351221p21351221.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From edwardlobo at gmail.com Thu Jan 8 07:32:33 2009 From: edwardlobo at gmail.com (bahbie) Date: Thu, 8 Jan 2009 07:32:33 -0800 (PST) Subject: [Freeswitch-users] external users -> gateway *help please* Message-ID: <21352577.post@talk.nabble.com> I think there is some problem with the bridge and context. I get a message from FS saying the person you are trying to contact is not available. -- View this message in context: http://www.nabble.com/external-users--%3E-gateway-*help-please*-tp21351221p21352577.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ahgindia308 at gmail.com Thu Jan 8 08:46:40 2009 From: ahgindia308 at gmail.com (ahgindia) Date: Thu, 8 Jan 2009 08:46:40 -0800 (PST) Subject: [Freeswitch-users] Freeswitch not passing more than two calls Message-ID: <21355617.post@talk.nabble.com> Hi all, I have a freeswitch setup for bridging calls between two gateways. i.e. Originator -> Gateway A -> freeswitch -> Gateeway B -> terminator user I have it running very well for last 43 days. But from last one week, I noticed that freeswitch was not able to pass more than two calls at once. All the other calls were failing with hang up cause 41 (NORMAL_TEMPORARY_FAILURE). The freeswitch was back to normal work only I restarted freeswitch. Can anybody suggest any issue or suggestion for this problem. -- View this message in context: http://www.nabble.com/Freeswitch-not-passing-more-than-two-calls-tp21355617p21355617.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From freeswitch-users at lists.rupa.com Thu Jan 8 08:51:29 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Thu, 08 Jan 2009 10:51:29 -0600 Subject: [Freeswitch-users] originate and caller number In-Reply-To: References: Message-ID: <49662F11.2020809@lists.rupa.com> On 1/8/2009 10:02 AM, kriko wrote: > I wrote a java socket client that originates a call, for e.g.: > originate dingaling/gmail.com/atomic.devterium at gmail.com &bridge(loopback/1003/java_gmail_bridge) > > This works fine, however both ends doesn't really see each other numbers, instead they see freeswitch number and id. > Is it possible to show every user the opposite caller number? consider using channel variables for each leg to set the originating callerid information. Look at: http://wiki.freeswitch.org/wiki/Channel_Variables for how to set the vars inline with a dial string and the appropriate vars to set. > dialplan used: > http://pastebin.com/m71525ac6 > From kristian.kielhofner at gmail.com Thu Jan 8 09:06:39 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 8 Jan 2009 12:06:39 -0500 Subject: [Freeswitch-users] Export variables from originate command In-Reply-To: <191c3a030901080614o4f0da4dkd070da9530ad7828@mail.gmail.com> References: <2d9149cd0901072250o4d08be02m1f4a3e6276d750e4@mail.gmail.com> <191c3a030901080614o4f0da4dkd070da9530ad7828@mail.gmail.com> Message-ID: <2d9149cd0901080906n2445bc71s6b88f6575abfc666@mail.gmail.com> On 1/8/09, Anthony Minessale wrote: > put them in {} comma separated. > > > {foo=bar,test=true}sofia/default/user at dom.com > > if you are doing forked dial you can set them per leg with [] > > [var1=foo]sofia/default/user at dom.com,[var1=bar]sofia/default/user2 at dom.com > Tony, Thanks for getting back to me. This part I understand (using commas, etc). What I need to figure out is how to get those variables to be available to the application I am calling after the channel in the originate command. In this case transfer to put my call back into the dialplan. basically I need a way to set variables in the originate command that are available to the application I am calling, not just the channel I am creating with the URL syntax. Am I making anymore sense? Thanks again! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From Prometheus001 at gmx.net Thu Jan 8 09:15:11 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 08 Jan 2009 18:15:11 +0100 Subject: [Freeswitch-users] Openzap: every second incoming call fails Message-ID: <4966349F.7090608@gmx.net> We have a strange Issue on Openzap with a Digium PRI card (Digium TE220 inkl. VPMOCT064 Octasic DSP-based echo cancellation module) Every second incoming call is successfoll, every second incoming one fails. For me it seems as if FS tries to use a channel which may be still occupied? Anybody has an idea? Last hangup from successful!! call: 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[5a] Size:[103] CRV: 17 (0x11, CTX: Originator) 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() Changing state on 1:1 from TERMINATING to DOWN 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE [DOWN] 2009-01-08 17:58:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 40 2nd Incoming Channel fails -------------------------------------------------------------------------------- [08 02 00 46 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 34 34 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[05] Size:[179] CRV: 70 (0x46, CTX: Originator) 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0x46] 2009-01-08 17:58:32 [WARNING] ozmod_isdn.c:572 zap_isdn_931_34() Channel 2:1 ~ 2:32 is already in use waiting for it to become available. 2009-01-08 17:58:33 [ERR] ozmod_isdn.c:586 zap_isdn_931_34() Channel 2:1 ~ 2:32 is already in use. 2009-01-08 17:58:33 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 -------------------------------------------------------------------------------- [08 02 80 46 45 08 02 81 e5] 2009-01-08 17:58:33 [CRIT] ozmod_isdn.c:644 zap_isdn_931_34() Failed to open channel for new setup message 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 -------------------------------------------------------------------------------- [08 02 00 46 4d] 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0x46] 2009-01-08 17:58:34 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received Release with no matching channel 0 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse error [-3012] [Q931E_INVALID_CRV] 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 -------------------------------------------------------------------------------- [08 02 00 46 4d] 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0x46] 2009-01-08 17:58:38 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received Release with no matching channel 0 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse error [-3012] [Q931E_INVALID_CRV] Best regards Peter From Prometheus001 at gmx.net Thu Jan 8 09:21:38 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 08 Jan 2009 18:21:38 +0100 Subject: [Freeswitch-users] originate and caller number In-Reply-To: References: Message-ID: <49663622.2060401@gmx.net> I think so. I would do it the following: pass your variables for your outgoing number in front of your originate string: originate {var1=xxx, var2=xxx}dingaling/gmail.com/atomic.devterium at gmail.com Then bridge it to a destination in your app or dialplan and set some vars there. Is that what solves your problem? Best regards Peter kriko schrieb: > I wrote a java socket client that originates a call, for e.g.: > originate dingaling/gmail.com/atomic.devterium at gmail.com &bridge(loopback/1003/java_gmail_bridge) > > This works fine, however both ends doesn't really see each other numbers, instead they see freeswitch number and id. > Is it possible to show every user the opposite caller number? > > dialplan used: > http://pastebin.com/m71525ac6 > > From matthew at matthew.at Thu Jan 8 11:47:54 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Thu, 08 Jan 2009 11:47:54 -0800 Subject: [Freeswitch-users] polycom one-way audio problem (solved) In-Reply-To: <191c3a030901080605p72904873yd7f107e8fa336aed@mail.gmail.com> References: <49644A22.8050802@matthew.at> <4964F573.2050307@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> <4964F893.6020302@matthew.at> <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> <4964FBCE.8090008@matthew.at> <49650637.7040703@matthew.at> <496588FE.60409@matthew.at> <4965A261.9030708@matthew.at> <191c3a030901080605p72904873yd7f107e8fa336aed@mail.gmail.com> Message-ID: <4966586A.2000706@matthew.at> Anthony Minessale wrote: > This is a very unique problem as many people get this basic situation > working daily so > it must be a network issue of some sort. As I said yesterday, a network problem makes the most sense, but the behavior was still very strange. I have now tracked down the problem, and the issue also explains why changing firmware and changing FreeSWITCH settings *appeared* to make a difference while not actually having any effect on the root cause. The server running FreeSWITCH also had tunneling software enabled that, when traffic for RFC1918 space was detected coming from the machine, installed a policy route that forced traffic to exit via that tunnel (but had no effect on any RFC1918 coming in from the outside). The same software also ensured that if traffic from RFC1918 space came in first, a policy would instead be installed (on a per address/port basis) that allowed that traffic to flow via the native interface. As it happened, this conflicted with my routing configuration, which is that my phones were on a network using RFC1918 addresses. There was no NAT, so it should have worked to use RFC1918 addresses, but the tunnel policy routing choice of trigger addresses overlapped the actual addresses of my phones, thus the problem. This tunnel policy routing causes the following behavior: 1) SIP works (phone always sends first packet when registering, bidirectional policy installed) 2) RTP works *if* the phone sent the first RTP packet (phone sends first, bidirectional policy installed) 3) RTP is received successfully from the phone at the switch, and RTP appears to be sent (via tcpdump) from the switch to the phone but does not actually arrive at the phone *if* FreeSWITCH sends first (FreeSWITCH sends first, tunnel outbound policy is installed forcing traffic to be routed into the tunnel instead of towards the phone). As a result, things where the phone always gets the first RTP out (e.g., calling local voicemail box, where the recording-playback is being clocked by the RTP coming in) work. Things where the switch always gets the first RTP out (often the case with early media for ringback, for instance) always cause the calling party to never hear anything for the rest of the call (which also explains why transfer from ringback to voicemail greeting still isn't audible... even though there's new signalling when it goes from early media to answered by voicemail, the same (blocked-in-one-direction) RTP port is in use) Interestingly, with the old firmware the switch sent early media to the caller (breaking its ability to hear the called party) and the called party always sent RTP first when the phone was picked up... the new firmware doesn't send RTP instantly upon picking up the ringing phone, and so the incoming RTP audio from the switch triggers the same policy routing issue, thus making it impossible for either end to hear the other. Likewise, making changes to settings like inbound-proxy-media causes changes in who sends RTP first for each end of the call, also changing the behavior. Thanks to the folks who reviewed my traces and configurations to make sure that everything seemed reasonable on the switch side. As it was determined yesterday, the best next step was the verify that the packets really arrived at the phone, which as described above, they don't. Hopefully this information will be helpful to someone who encounters the same problem in the future, rare as it might be. Matthew Kaufman From edwardlobo at gmail.com Thu Jan 8 12:38:26 2009 From: edwardlobo at gmail.com (bahbie) Date: Thu, 8 Jan 2009 12:38:26 -0800 (PST) Subject: [Freeswitch-users] quick help Message-ID: <21360470.post@talk.nabble.com> On the examples to configure offsite phones I read the following: Below are steps to get remote Exts 20xx able to call one another and call on-site telephones. How to change those examples for one of the offsite phones to *also* use a trunk to call other pstn numbers ? http://wiki.freeswitch.org/wiki/Example_Offsite_phones doublenat settings are here -- View this message in context: http://www.nabble.com/quick-help-tp21360470p21360470.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From edwardlobo at gmail.com Thu Jan 8 12:54:21 2009 From: edwardlobo at gmail.com (bahbie) Date: Thu, 8 Jan 2009 12:54:21 -0800 (PST) Subject: [Freeswitch-users] quick help In-Reply-To: <21360470.post@talk.nabble.com> References: <21360470.post@talk.nabble.com> Message-ID: <21360763.post@talk.nabble.com> If i use the internal profile with the following two lines param name="ext-rtp-ip" value="$${external_rtp_ip}" param name="ext-sip-ip" value="$${external_sip_ip}" I can use the trunk however I get the following two errors on the cli. Don't know if they are serious errors 2009-01-08 20:49:59 [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2009-01-08 20:49:59 [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] bahbie wrote: > > On the examples to configure offsite phones I read the following: > > Below are steps to get remote Exts 20xx able to call one another and call > on-site telephones. > > How to change those examples for one of the offsite phones to *also* use a > trunk to call other pstn numbers ? > > http://wiki.freeswitch.org/wiki/Example_Offsite_phones > > doublenat settings are here > -- View this message in context: http://www.nabble.com/quick-help-tp21360470p21360763.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kristian.kielhofner at gmail.com Thu Jan 8 13:23:18 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 8 Jan 2009 16:23:18 -0500 Subject: [Freeswitch-users] Hint for DTMF handling in sofia.c In-Reply-To: <496628D9.5050805@ewetel.de> References: <4965E784.5030803@ewetel.de> <49661C8A.5020806@ewetel.de> <496628D9.5050805@ewetel.de> Message-ID: <2d9149cd0901081323ka0fb8br536c6b7322962d3e@mail.gmail.com> On 1/8/09, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > > Hi Brian, > > sorry, got lost ... It's a bug in Snom and AVM's phones? > > Maybe we misunderstood. RFC 2833 emtioned in chapter 3.10 following DTMF > Events: > > 3.10 DTMF Events > > Table 1 summarizes the DTMF-related named events within the > telephone-event payload format. > > Event encoding (decimal) > _________________________ > 0--9 0--9 > * 10 > # 11 > A--D 12--15 > Flash 16 > > Table 1: DTMF named events > > Most phones in germany are able to send * and # DTMF events and > Phone-applications like voiceboxes or IVRs need those Events. > > Therefor I don't think it's a bug in Snom or AVM ... > > But maybe you mean Snom and AVM should send "Signal=10" instead of > "Signal=*" ? Yes, it should use the decimal encoding from the table you mentioned. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From markgreene at gmail.com Thu Jan 8 14:13:37 2009 From: markgreene at gmail.com (Mark Greene) Date: Thu, 8 Jan 2009 16:13:37 -0600 Subject: [Freeswitch-users] polycom one-way audio problem (solved) In-Reply-To: <4966586A.2000706@matthew.at> References: <49644A22.8050802@matthew.at> <42482009-4A97-4725-931F-AA077E8359F2@freeswitch.org> <4964F893.6020302@matthew.at> <313F15D8-3786-485B-9BCD-E1DCA3B254A3@freeswitch.org> <4964FBCE.8090008@matthew.at> <49650637.7040703@matthew.at> <496588FE.60409@matthew.at> <4965A261.9030708@matthew.at> <191c3a030901080605p72904873yd7f107e8fa336aed@mail.gmail.com> <4966586A.2000706@matthew.at> Message-ID: <8ecbc2000901081413l1ddbe5f9uc33fea4fb2e9acf2@mail.gmail.com> Thanks for posting the solution. I was following the issue with much curiosity! On Thu, Jan 8, 2009 at 1:47 PM, Matthew Kaufman wrote: > Anthony Minessale wrote: > > This is a very unique problem as many people get this basic situation > > working daily so > > it must be a network issue of some sort. > As I said yesterday, a network problem makes the most sense, but the > behavior was still very strange. > > I have now tracked down the problem, and the issue also explains why > changing firmware and changing FreeSWITCH settings *appeared* to make a > difference while not actually having any effect on the root cause. > > The server running FreeSWITCH also had tunneling software enabled that, > when traffic for RFC1918 space was detected coming from the machine, > installed a policy route that forced traffic to exit via that tunnel > (but had no effect on any RFC1918 coming in from the outside). The same > software also ensured that if traffic from RFC1918 space came in first, > a policy would instead be installed (on a per address/port basis) that > allowed that traffic to flow via the native interface. As it happened, > this conflicted with my routing configuration, which is that my phones > were on a network using RFC1918 addresses. There was no NAT, so it > should have worked to use RFC1918 addresses, but the tunnel policy > routing choice of trigger addresses overlapped the actual addresses of > my phones, thus the problem. > > This tunnel policy routing causes the following behavior: > 1) SIP works (phone always sends first packet when registering, > bidirectional policy installed) > 2) RTP works *if* the phone sent the first RTP packet (phone sends > first, bidirectional policy installed) > 3) RTP is received successfully from the phone at the switch, and RTP > appears to be sent (via tcpdump) from the switch to the phone but does > not actually arrive at the phone *if* FreeSWITCH sends first (FreeSWITCH > sends first, tunnel outbound policy is installed forcing traffic to be > routed into the tunnel instead of towards the phone). > > As a result, things where the phone always gets the first RTP out (e.g., > calling local voicemail box, where the recording-playback is being > clocked by the RTP coming in) work. Things where the switch always gets > the first RTP out (often the case with early media for ringback, for > instance) always cause the calling party to never hear anything for the > rest of the call (which also explains why transfer from ringback to > voicemail greeting still isn't audible... even though there's new > signalling when it goes from early media to answered by voicemail, the > same (blocked-in-one-direction) RTP port is in use) > > Interestingly, with the old firmware the switch sent early media to the > caller (breaking its ability to hear the called party) and the called > party always sent RTP first when the phone was picked up... the new > firmware doesn't send RTP instantly upon picking up the ringing phone, > and so the incoming RTP audio from the switch triggers the same policy > routing issue, thus making it impossible for either end to hear the other. > > Likewise, making changes to settings like inbound-proxy-media causes > changes in who sends RTP first for each end of the call, also changing > the behavior. > > Thanks to the folks who reviewed my traces and configurations to make > sure that everything seemed reasonable on the switch side. As it was > determined yesterday, the best next step was the verify that the packets > really arrived at the phone, which as described above, they don't. > > Hopefully this information will be helpful to someone who encounters the > same problem in the future, rare as it might be. > > Matthew Kaufman > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/9905ad30/attachment.html From brian at freeswitch.org Thu Jan 8 15:38:34 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jan 2009 17:38:34 -0600 Subject: [Freeswitch-users] Hint for DTMF handling in sofia.c In-Reply-To: <2d9149cd0901081323ka0fb8br536c6b7322962d3e@mail.gmail.com> References: <4965E784.5030803@ewetel.de> <49661C8A.5020806@ewetel.de> <496628D9.5050805@ewetel.de> <2d9149cd0901081323ka0fb8br536c6b7322962d3e@mail.gmail.com> Message-ID: <2FFC8E58-470D-40D7-B6C3-47198D64FC53@freeswitch.org> Snom has already responded to my issue and are going to be providing me a firmware for testing this tomorrow.. its still going to default to the WRONG way.. but has a toggle to turn it to the right way. /b On Jan 8, 2009, at 3:23 PM, Kristian Kielhofner wrote: > > Yes, it should use the decimal encoding from the table you mentioned. From msc at freeswitch.org Thu Jan 8 15:50:37 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Jan 2009 15:50:37 -0800 Subject: [Freeswitch-users] Export variables from originate command In-Reply-To: <2d9149cd0901080906n2445bc71s6b88f6575abfc666@mail.gmail.com> References: <2d9149cd0901072250o4d08be02m1f4a3e6276d750e4@mail.gmail.com> <191c3a030901080614o4f0da4dkd070da9530ad7828@mail.gmail.com> <2d9149cd0901080906n2445bc71s6b88f6575abfc666@mail.gmail.com> Message-ID: <87f2f3b90901081550q6c13f739n378e27af38ca4c6a@mail.gmail.com> On Thu, Jan 8, 2009 at 9:06 AM, Kristian Kielhofner wrote: > On 1/8/09, Anthony Minessale wrote: >> put them in {} comma separated. >> >> >> {foo=bar,test=true}sofia/default/user at dom.com >> >> if you are doing forked dial you can set them per leg with [] >> >> [var1=foo]sofia/default/user at dom.com,[var1=bar]sofia/default/user2 at dom.com >> > > Tony, > > Thanks for getting back to me. This part I understand (using > commas, etc). What I need to figure out is how to get those variables > to be available to the application I am calling after the channel in > the originate command. In this case transfer to put my call back into > the dialplan. > > basically I need a way to set variables in the originate command > that are available to the application I am calling, not just the > channel I am creating with the URL syntax. What application(s) are you calling? Could you post an example? -MC > > Am I making anymore sense? > > Thanks again! > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Thu Jan 8 15:52:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Jan 2009 15:52:42 -0800 Subject: [Freeswitch-users] external users -> gateway *help please* In-Reply-To: <21352577.post@talk.nabble.com> References: <21352577.post@talk.nabble.com> Message-ID: <87f2f3b90901081552h85f1e2bv74699cb2e424673e@mail.gmail.com> Could you go to pastebin.freeswitch.org and paste your config changes? -MC On Thu, Jan 8, 2009 at 7:32 AM, bahbie wrote: > > I think there is some problem with the bridge and context. I get a message > from FS saying the person you are trying to contact is not available. > -- > View this message in context: http://www.nabble.com/external-users--%3E-gateway-*help-please*-tp21351221p21352577.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Thu Jan 8 15:55:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Jan 2009 17:55:40 -0600 Subject: [Freeswitch-users] Export variables from originate command In-Reply-To: <87f2f3b90901081550q6c13f739n378e27af38ca4c6a@mail.gmail.com> References: <2d9149cd0901072250o4d08be02m1f4a3e6276d750e4@mail.gmail.com> <191c3a030901080614o4f0da4dkd070da9530ad7828@mail.gmail.com> <2d9149cd0901080906n2445bc71s6b88f6575abfc666@mail.gmail.com> <87f2f3b90901081550q6c13f739n378e27af38ca4c6a@mail.gmail.com> Message-ID: <191c3a030901081555yb17c0c4lb24e1cf87fd23b71@mail.gmail.com> the variables should still be set on the channel in your application? you mean a remote call-leg or the channel itself once it's in the app? On Thu, Jan 8, 2009 at 5:50 PM, Michael Collins wrote: > On Thu, Jan 8, 2009 at 9:06 AM, Kristian Kielhofner > wrote: > > On 1/8/09, Anthony Minessale wrote: > >> put them in {} comma separated. > >> > >> > >> {foo=bar,test=true}sofia/default/user at dom.com > >> > >> if you are doing forked dial you can set them per leg with [] > >> > >> [var1=foo]sofia/default/user at dom.com,[var1=bar]sofia/default/ > user2 at dom.com > >> > > > > Tony, > > > > Thanks for getting back to me. This part I understand (using > > commas, etc). What I need to figure out is how to get those variables > > to be available to the application I am calling after the channel in > > the originate command. In this case transfer to put my call back into > > the dialplan. > > > > basically I need a way to set variables in the originate command > > that are available to the application I am calling, not just the > > channel I am creating with the URL syntax. > > What application(s) are you calling? Could you post an example? > -MC > > > > > > Am I making anymore sense? > > > > Thanks again! > > > > -- > > Kristian Kielhofner > > http://blog.krisk.org > > http://www.submityoursip.com > > http://www.astlinux.org > > http://www.star2star.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/146eb752/attachment.html From msc at freeswitch.org Thu Jan 8 15:56:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Jan 2009 15:56:17 -0800 Subject: [Freeswitch-users] Openzap: every second incoming call fails In-Reply-To: <4966349F.7090608@gmx.net> References: <4966349F.7090608@gmx.net> Message-ID: <87f2f3b90901081556t30e42d3bhaaec89e16ed8dff4@mail.gmail.com> Can you pastebin a complete call history where the first call works, gets hung up and then the second call comes in? I would like to see the entire d-chan trace. -MC On Thu, Jan 8, 2009 at 9:15 AM, Peter P GMX wrote: > We have a strange Issue on Openzap with a Digium PRI card (Digium TE220 > inkl. VPMOCT064 Octasic DSP-based echo cancellation module) > > Every second incoming call is successfoll, every second incoming one > fails. For me it seems as if FS tries to use a channel which may be > still occupied? > > Anybody has an idea? > > Last hangup from successful!! call: > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[5a] Size:[103] CRV: 17 (0x11, CTX: Originator) > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() Changing > state on 1:1 from TERMINATING to DOWN > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE > [DOWN] > 2009-01-08 17:58:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 40 > > > 2nd Incoming Channel fails > -------------------------------------------------------------------------------- > [08 02 00 46 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 34 34 > 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] > > 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[05] Size:[179] CRV: 70 (0x46, CTX: Originator) > 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 > (-1:-1) source isdn_data->channels_remote_crv[0x46] > 2009-01-08 17:58:32 [WARNING] ozmod_isdn.c:572 zap_isdn_931_34() Channel > 2:1 ~ 2:32 is already in use waiting for it to become available. > 2009-01-08 17:58:33 [ERR] ozmod_isdn.c:586 zap_isdn_931_34() Channel 2:1 > ~ 2:32 is already in use. > 2009-01-08 17:58:33 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 > -------------------------------------------------------------------------------- > [08 02 80 46 45 08 02 81 e5] > > 2009-01-08 17:58:33 [CRIT] ozmod_isdn.c:644 zap_isdn_931_34() Failed to > open channel for new setup message > 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 > -------------------------------------------------------------------------------- > [08 02 00 46 4d] > > 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) > 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 > (-1:-1) source isdn_data->channels_remote_crv[0x46] > 2009-01-08 17:58:34 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received > Release with no matching channel 0 > 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse > error [-3012] [Q931E_INVALID_CRV] > 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 > -------------------------------------------------------------------------------- > [08 02 00 46 4d] > > 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) > 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 > (-1:-1) source isdn_data->channels_remote_crv[0x46] > 2009-01-08 17:58:38 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received > Release with no matching channel 0 > 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse > error [-3012] [Q931E_INVALID_CRV] > > > Best regards Peter > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Thu Jan 8 15:58:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Jan 2009 15:58:12 -0800 Subject: [Freeswitch-users] Freeswitch not passing more than two calls In-Reply-To: <21355617.post@talk.nabble.com> References: <21355617.post@talk.nabble.com> Message-ID: <87f2f3b90901081558u13d08328q97d4c998c45f81ad@mail.gmail.com> So the issue is not happening right now? If it is then we would want you to pastebin the debug output of a few test calls. If not, please watch to see if this issue happens again and then report it back with a debug trace. -MC On Thu, Jan 8, 2009 at 8:46 AM, ahgindia wrote: > > Hi all, > I have a freeswitch setup for bridging calls between two gateways. > i.e. Originator -> Gateway A -> freeswitch -> Gateeway B -> terminator user > I have it running very well for last 43 days. > But from last one week, I noticed that freeswitch was not able to pass more > than two calls at once. All the other calls were failing with hang up cause > 41 (NORMAL_TEMPORARY_FAILURE). > The freeswitch was back to normal work only I restarted freeswitch. > Can anybody suggest any issue or suggestion for this problem. > -- > View this message in context: http://www.nabble.com/Freeswitch-not-passing-more-than-two-calls-tp21355617p21355617.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Thu Jan 8 16:01:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Jan 2009 16:01:25 -0800 Subject: [Freeswitch-users] Trouble getting session.setInputCallback working. In-Reply-To: <3d381e170901071609h3cbefccga01778685952cd60@mail.gmail.com> References: <3d381e170901071603m1943287fwba1ef838678117bb@mail.gmail.com> <3d381e170901071609h3cbefccga01778685952cd60@mail.gmail.com> Message-ID: <87f2f3b90901081601r6e0ffc69m466c9d819dde0030@mail.gmail.com> Sorry for the late followup. Did you ever get this working? (I'm not a Python guy so it's a bit out of my area of interest/expertise). -MC On Wed, Jan 7, 2009 at 4:09 PM, Erik Wickstrom wrote: > Hi all, > > I'm trying to get a setInputCallback function working with mod_python. I'm > using a current svn checkout for my build and the hello world via call > example from the wiki ( > http://wiki.freeswitch.org/wiki/Mod_python#Hello_World_via_call ). > > I've tried repeatedly, but I can't get the callback function to execute. I > keep pressing various digits on my phone, but none of them are being logged > to the console. My understanding is that this callback is supposed to fire > when a dtmf (123456789*#, right??) and also acts as an event handler for > things like mod_vmd. > > Is this part of the wiki example working for anyone else? Any idea what I > might be doing wrong? > > Thanks! > Erik > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From kristian.kielhofner at gmail.com Thu Jan 8 17:29:32 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 8 Jan 2009 20:29:32 -0500 Subject: [Freeswitch-users] Export variables from originate command In-Reply-To: <191c3a030901081555yb17c0c4lb24e1cf87fd23b71@mail.gmail.com> References: <2d9149cd0901072250o4d08be02m1f4a3e6276d750e4@mail.gmail.com> <191c3a030901080614o4f0da4dkd070da9530ad7828@mail.gmail.com> <2d9149cd0901080906n2445bc71s6b88f6575abfc666@mail.gmail.com> <87f2f3b90901081550q6c13f739n378e27af38ca4c6a@mail.gmail.com> <191c3a030901081555yb17c0c4lb24e1cf87fd23b71@mail.gmail.com> Message-ID: <2d9149cd0901081729v1e5001d5icfd0fa6b06a17ec2@mail.gmail.com> Here's an example: bgapi originate {myvar=blah}sofia/gateway/gw/19415551212 &transfer(500 XML default) 9185551212 60 So... I'd like to be able to read ${myvar} from the the 500 extension in the default context. On 1/8/09, Anthony Minessale wrote: > the variables should still be set on the channel in your application? > you mean a remote call-leg or the channel itself once it's in the app? > > > On Thu, Jan 8, 2009 at 5:50 PM, Michael Collins wrote: > > > On Thu, Jan 8, 2009 at 9:06 AM, Kristian Kielhofner > > > > wrote: > > > > > On 1/8/09, Anthony Minessale wrote: > > >> put them in {} comma separated. > > >> > > >> > > >> {foo=bar,test=true}sofia/default/user at dom.com > > >> > > >> if you are doing forked dial you can set them per leg with [] > > >> > > >> > [var1=foo]sofia/default/user at dom.com,[var1=bar]sofia/default/user2 at dom.com > > >> > > > > > > Tony, > > > > > > Thanks for getting back to me. This part I understand (using > > > commas, etc). What I need to figure out is how to get those variables > > > to be available to the application I am calling after the channel in > > > the originate command. In this case transfer to put my call back into > > > the dialplan. > > > > > > basically I need a way to set variables in the originate command > > > that are available to the application I am calling, not just the > > > channel I am creating with the URL syntax. > > > > What application(s) are you calling? Could you post an example? > > -MC > > > > > > > > > > > > > > > > Am I making anymore sense? > > > > > > Thanks again! > > > > > > -- > > > Kristian Kielhofner > > > http://blog.krisk.org > > > http://www.submityoursip.com > > > http://www.astlinux.org > > > http://www.star2star.com > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at 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 at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From anthony.minessale at gmail.com Thu Jan 8 18:02:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Jan 2009 20:02:07 -0600 Subject: [Freeswitch-users] Export variables from originate command In-Reply-To: <2d9149cd0901081729v1e5001d5icfd0fa6b06a17ec2@mail.gmail.com> References: <2d9149cd0901072250o4d08be02m1f4a3e6276d750e4@mail.gmail.com> <191c3a030901080614o4f0da4dkd070da9530ad7828@mail.gmail.com> <2d9149cd0901080906n2445bc71s6b88f6575abfc666@mail.gmail.com> <87f2f3b90901081550q6c13f739n378e27af38ca4c6a@mail.gmail.com> <191c3a030901081555yb17c0c4lb24e1cf87fd23b71@mail.gmail.com> <2d9149cd0901081729v1e5001d5icfd0fa6b06a17ec2@mail.gmail.com> Message-ID: <191c3a030901081802k30afd59fm9e843c13b648ef8c@mail.gmail.com> yes it will be there in ext 500. any vars in {} are set when the channel is created and are present the rest of the life of the channel. you can do regex on it in the dialplan by referencing it with a ${} in the field ... and any apps you execute will also be able to get the value of ${myvar} On Thu, Jan 8, 2009 at 7:29 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Here's an example: > > bgapi originate {myvar=blah}sofia/gateway/gw/19415551212 &transfer(500 > XML default) 9185551212 60 > > So... I'd like to be able to read ${myvar} from the the 500 extension > in the default context. > > On 1/8/09, Anthony Minessale wrote: > > the variables should still be set on the channel in your application? > > you mean a remote call-leg or the channel itself once it's in the app? > > > > > > On Thu, Jan 8, 2009 at 5:50 PM, Michael Collins > wrote: > > > > > On Thu, Jan 8, 2009 at 9:06 AM, Kristian Kielhofner > > > > > > wrote: > > > > > > > On 1/8/09, Anthony Minessale wrote: > > > >> put them in {} comma separated. > > > >> > > > >> > > > >> {foo=bar,test=true}sofia/default/user at dom.com > > > >> > > > >> if you are doing forked dial you can set them per leg with [] > > > >> > > > >> > > [var1=foo]sofia/default/user at dom.com,[var1=bar]sofia/default/ > user2 at dom.com > > > >> > > > > > > > > Tony, > > > > > > > > Thanks for getting back to me. This part I understand (using > > > > commas, etc). What I need to figure out is how to get those > variables > > > > to be available to the application I am calling after the channel in > > > > the originate command. In this case transfer to put my call back > into > > > > the dialplan. > > > > > > > > basically I need a way to set variables in the originate command > > > > that are available to the application I am calling, not just the > > > > channel I am creating with the URL syntax. > > > > > > What application(s) are you calling? Could you post an example? > > > -MC > > > > > > > > > > > > > > > > > > > > > > > Am I making anymore sense? > > > > > > > > Thanks again! > > > > > > > > -- > > > > Kristian Kielhofner > > > > http://blog.krisk.org > > > > http://www.submityoursip.com > > > > http://www.astlinux.org > > > > http://www.star2star.com > > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at 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 at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/079625b1/attachment.html From kristian.kielhofner at gmail.com Thu Jan 8 18:13:47 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 8 Jan 2009 21:13:47 -0500 Subject: [Freeswitch-users] mod_dptools/event docs? Message-ID: <2d9149cd0901081813o60c51965m5cd4a9e62667a25d@mail.gmail.com> mod_dptools/event doesn't have a wiki page... Anyone care to give me some syntax so I can make one? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From erik at erikwickstrom.com Thu Jan 8 18:33:38 2009 From: erik at erikwickstrom.com (Erik Wickstrom) Date: Thu, 8 Jan 2009 18:33:38 -0800 Subject: [Freeswitch-users] Trouble getting session.setInputCallback working. In-Reply-To: <87f2f3b90901081601r6e0ffc69m466c9d819dde0030@mail.gmail.com> References: <3d381e170901071603m1943287fwba1ef838678117bb@mail.gmail.com> <3d381e170901071609h3cbefccga01778685952cd60@mail.gmail.com> <87f2f3b90901081601r6e0ffc69m466c9d819dde0030@mail.gmail.com> Message-ID: <3d381e170901081833h6cfe4992saaa96fb23be0a5c3@mail.gmail.com> Yes, I think so. It seems that it just doesn't work while doing an api call such as session.execute("playback", "/path/to/file.wav") Is this the correct behaviour? Are calls made with execute "blocking"? Thanks! Erik On Thu, Jan 8, 2009 at 4:01 PM, Michael Collins wrote: > Sorry for the late followup. Did you ever get this working? (I'm not a > Python guy so it's a bit out of my area of interest/expertise). > -MC > > On Wed, Jan 7, 2009 at 4:09 PM, Erik Wickstrom > wrote: > > Hi all, > > > > I'm trying to get a setInputCallback function working with mod_python. > I'm > > using a current svn checkout for my build and the hello world via call > > example from the wiki ( > > http://wiki.freeswitch.org/wiki/Mod_python#Hello_World_via_call ). > > > > I've tried repeatedly, but I can't get the callback function to execute. > I > > keep pressing various digits on my phone, but none of them are being > logged > > to the console. My understanding is that this callback is supposed to > fire > > when a dtmf (123456789*#, right??) and also acts as an event handler for > > things like mod_vmd. > > > > Is this part of the wiki example working for anyone else? Any idea what > I > > might be doing wrong? > > > > Thanks! > > Erik > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/40647c46/attachment-0001.html From royce3 at westparkcom.net Thu Jan 8 19:13:26 2009 From: royce3 at westparkcom.net (Royce Mitchell III) Date: Thu, 08 Jan 2009 21:13:26 -0600 Subject: [Freeswitch-users] why doesn't this work? Message-ID: <4966C0D6.4010609@westparkcom.net> I'm trying to program an extension in the dialplan to do an intercom announce. I read through the wiki and wrote the following based on what I thought I understood from it, but it's not working the way I expect: ;answer-after=0]]> From wiltingtree at gmail.com Thu Jan 8 19:20:13 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Thu, 8 Jan 2009 22:20:13 -0500 Subject: [Freeswitch-users] vmd /garbled messages Message-ID: Hi, I have two issues I'd appreciate some help with. A) I'm testing VMD and I'm getting a success rate of well under 50%. I know part of the reason is that some of the voicemail beeps it's encountering are very short in length (I've noticed this for T-Mobile and Sprint voicemails, and there may be others too), and it can't detect them. So my question is about the notes in the vmd section of the wiki which states, "The industry standard is 80% detection. This module if used properly should exceeds the standard by a very wide margin". I'm curious about whether I'm using it properly, and what I can do to make it work better. Thanks for the help. B) When I place an outbound call and immediately play a prompt when the call is answered, the prompt sounds garbled to the person answering the phone. If I sleep for a second before playing the prompt, it sounds fine. Any idea of what would cause this? Thanks, Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/7ee3f523/attachment.html From kristian.kielhofner at gmail.com Thu Jan 8 19:30:28 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 8 Jan 2009 22:30:28 -0500 Subject: [Freeswitch-users] vmd /garbled messages In-Reply-To: References: Message-ID: <2d9149cd0901081930m6e04322fnd5ab84718245c5f3@mail.gmail.com> On 1/8/09, Adam Wilt wrote: > Hi, I have two issues I'd appreciate some help with. > > A) I'm testing VMD and I'm getting a success rate of well under 50%. I know > part of the reason is that some of the voicemail beeps it's encountering are > very short in length (I've noticed this for T-Mobile and Sprint voicemails, > and there may be others too), and it can't detect them. So my question is > about the notes in the vmd section of the wiki which states, "The industry > standard is 80% detection. This module if used properly should exceeds the > standard by a very wide margin". I'm curious about whether I'm using it > properly, and what I can do to make it work better. Thanks for the help. > > B) When I place an outbound call and immediately play a prompt when the call > is answered, the prompt sounds garbled to the person answering the phone. If > I sleep for a second before playing the prompt, it sounds fine. Any idea of > what would cause this? > > Thanks, > Adam > Adam, Can I ask how you are "testing" vmd? Where? How? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From jkr888 at gmail.com Thu Jan 8 20:18:16 2009 From: jkr888 at gmail.com (Johny Kadarisman) Date: Thu, 8 Jan 2009 23:18:16 -0500 Subject: [Freeswitch-users] vmd /garbled messages In-Reply-To: <2d9149cd0901081930m6e04322fnd5ab84718245c5f3@mail.gmail.com> References: <2d9149cd0901081930m6e04322fnd5ab84718245c5f3@mail.gmail.com> Message-ID: one of hidden feature of freeswitch ;) http://wiki.freeswitch.org/wiki/Mod_vmd On Thu, Jan 8, 2009 at 10:30 PM, Kristian Kielhofner wrote: > On 1/8/09, Adam Wilt wrote: >> Hi, I have two issues I'd appreciate some help with. >> >> A) I'm testing VMD and I'm getting a success rate of well under 50%. I know >> part of the reason is that some of the voicemail beeps it's encountering are >> very short in length (I've noticed this for T-Mobile and Sprint voicemails, >> and there may be others too), and it can't detect them. So my question is >> about the notes in the vmd section of the wiki which states, "The industry >> standard is 80% detection. This module if used properly should exceeds the >> standard by a very wide margin". I'm curious about whether I'm using it >> properly, and what I can do to make it work better. Thanks for the help. >> >> B) When I place an outbound call and immediately play a prompt when the call >> is answered, the prompt sounds garbled to the person answering the phone. If >> I sleep for a second before playing the prompt, it sounds fine. Any idea of >> what would cause this? >> >> Thanks, >> Adam >> > > Adam, > > Can I ask how you are "testing" vmd? Where? How? > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mileskeaton at gmail.com Thu Jan 8 14:24:56 2009 From: mileskeaton at gmail.com (Miles Keaton) Date: Thu, 8 Jan 2009 14:24:56 -0800 Subject: [Freeswitch-users] How does location of FreeSWITCH server matter? Message-ID: <59b2d39b0901081424n4b9acae0w51bfbe101b9f892c@mail.gmail.com> I'm new to FreeSWITCH and hope I could ask the community's suggestion or experience with how the location of your FreeSWITCH server affects the quality of the VoIP call. I'm using JunctionNetworks (http://pstn.junctionnetworks.com/) for SIP trunking. Their servers are in NYC. I'm trying to decide between running FreeSWITCH on a Debian dedicated server in Virginia (http://www.serverbeach.com/) or a FreeBSD dedicated server in San Diego (http://www.m5hosting.com/). This is for a VERY small business without a lot of phone traffic, but our users are spread around the world, in LA, Toronto, NYC, Berlin, London, Bangalore, and Hong Kong. I've used both for a couple months and they both seem fine. Since I'm in LA, I get a better response-speed from the San Diego server, but people on the east coast might have the opposite experience. My recent FreeSWITCH calls through the Virginia server have sounded quite lo-fi. I also like FreeBSD better than Debian. What would you recommend? Server-location doesn't matter? Server-location does matter? Location doesn't matter much but the colocation facility matters more? Any suggestions appreciated. I'm trying to make a decision, but realized I didn't have all the facts. Thanks! From jypeng at yahoo.com Thu Jan 8 18:44:15 2009 From: jypeng at yahoo.com (Jian Yuan Peng) Date: Thu, 8 Jan 2009 18:44:15 -0800 (PST) Subject: [Freeswitch-users] 404 error when try make a outbound call to voip provider. Message-ID: <460843.79689.qm@web50605.mail.re2.yahoo.com> Hi,?Can you help me to look at why I got this error. See this link for detailhttp://pastebin.com/m3be7932cAlso, one more question, what is most used java sip library connected with fs for make phone calls. I am looking at jain-sip now. I open for any.Thnaks,-Jian Yuan Peng -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/8d4705e0/attachment.html From msc at freeswitch.org Thu Jan 8 21:56:07 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 8 Jan 2009 21:56:07 -0800 Subject: [Freeswitch-users] 404 error when try make a outbound call to voip provider. In-Reply-To: <460843.79689.qm@web50605.mail.re2.yahoo.com> References: <460843.79689.qm@web50605.mail.re2.yahoo.com> Message-ID: <75E242A3-F8E9-403A-ABF6-1879F296ADEE@freeswitch.org> Jeng, Your condition expressions are not right. Could you describe what you hope to accomplish with those two expressions? Once you get the regular expressions figured out then it should all work. -MC Sent from my iPhone On Jan 8, 2009, at 6:44 PM, Jian Yuan Peng wrote: > Hi, > > Can you help me to look at why I got this error. See this link for > detail > > http://pastebin.com/m3be7932c > > Also, one more question, what is most used java sip library > connected with fs for make phone calls. I am looking at jain-sip > now. I open for any. > > Thnaks, > > -Jian Yuan Peng > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090108/06d1b3c8/attachment.html From kristian.kielhofner at gmail.com Thu Jan 8 22:45:49 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 9 Jan 2009 01:45:49 -0500 Subject: [Freeswitch-users] How does location of FreeSWITCH server matter? In-Reply-To: <59b2d39b0901081424n4b9acae0w51bfbe101b9f892c@mail.gmail.com> References: <59b2d39b0901081424n4b9acae0w51bfbe101b9f892c@mail.gmail.com> Message-ID: <2d9149cd0901082245q5a171a1bnce0775efb15e381a@mail.gmail.com> On 1/8/09, Miles Keaton wrote: > I'm new to FreeSWITCH and hope I could ask the community's suggestion > or experience with how the location of your FreeSWITCH server affects > the quality of the VoIP call. > > I'm using JunctionNetworks (http://pstn.junctionnetworks.com/) for SIP > trunking. Their servers are in NYC. I take it they at least proxy the media there whether or not they ultimately terminate/originate it on the PSTN themselves. > I'm trying to decide between running FreeSWITCH on a Debian dedicated > server in Virginia (http://www.serverbeach.com/) or a FreeBSD > dedicated server in San Diego (http://www.m5hosting.com/). Go with Virginia, and I will tell you why... > This is for a VERY small business without a lot of phone traffic, but > our users are spread around the world, in LA, Toronto, NYC, Berlin, > London, Bangalore, and Hong Kong. > > I've used both for a couple months and they both seem fine. Since I'm > in LA, I get a better response-speed from the San Diego server, but > people on the east coast might have the opposite experience. My > recent FreeSWITCH calls through the Virginia server have sounded quite > lo-fi. I also like FreeBSD better than Debian. > > What would you recommend? Virginia > Server-location doesn't matter? It matters. > Server-location does matter? It matters. > Location doesn't matter much but the colocation facility matters more? It matters, but the *bandwidth/network* used by your colo matters too. > Any suggestions appreciated. I'm trying to make a decision, but > realized I didn't have all the facts. > > Thanks! A little explanation... Your media is at least going to NYC no matter what you do. There is no other way it's going to get to/from the PSTN (that's where Junction is taking it, after all). In FreeSWITCH there are a lot of options for handling media but let's assume two basic possibilities: 1) You have to handle the media from your endpoints with FreeSWITCH. Your media will have to go from Endpoint <-> VA <-> NYC <-> PSTN 2) You don't have to handle the media with FreeSWITCH and it can go directly to New York: Endpoint <-> NYC <-> PSTN Either way, the path will be shorter if the server is in VA. The only way you could lose is if you really hate Debian or ServerBeach sucks. My experience with ServerBeach in VA is pretty good. The PEER1 network is well peered and quite decent. It can get you to New York quite quickly. Trust me, I know this ;). If your server was in LA your endpoints could still go directly to New York if possible (no nat, etc). If not, the media is now proxied in LA on it's way to NYC and vice versa. That's a long haul... Also, I would imagine you're planning on having media always served from FreeSWITCH - voicemail, conferencing, IVR, etc. In this case Virginia wins yet again. Virginia. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Thu Jan 8 22:58:55 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 9 Jan 2009 01:58:55 -0500 Subject: [Freeswitch-users] How does location of FreeSWITCH server matter? In-Reply-To: <2d9149cd0901082245q5a171a1bnce0775efb15e381a@mail.gmail.com> References: <59b2d39b0901081424n4b9acae0w51bfbe101b9f892c@mail.gmail.com> <2d9149cd0901082245q5a171a1bnce0775efb15e381a@mail.gmail.com> Message-ID: <2d9149cd0901082258o39de010akca4408b18c3ac864@mail.gmail.com> On 1/9/09, Kristian Kielhofner wrote: > On 1/8/09, Miles Keaton wrote: > > I'm new to FreeSWITCH and hope I could ask the community's suggestion > > or experience with how the location of your FreeSWITCH server affects > > the quality of the VoIP call. > > > > I'm using JunctionNetworks (http://pstn.junctionnetworks.com/) for SIP > > trunking. Their servers are in NYC. > > > I take it they at least proxy the media there whether or not they > ultimately terminate/originate it on the PSTN themselves. BTW, Junction uses Global Crossing for SIP origination and they appear to be colo'd with these guys: http://www.fortressitx.com. Just in case you need more information to make your decision ;). -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From matthew at matthew.at Fri Jan 9 00:19:13 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Fri, 09 Jan 2009 00:19:13 -0800 Subject: [Freeswitch-users] polycom shared line Message-ID: <49670881.7090207@matthew.at> Can FreeSWITCH and Polycom's line type="shared" work together? If so, how do I get there from here? I've tried registering as the same extension with multiple registrations enabled, registering as different extensions with the first extension set as the third-party in the second... looked at how manage-presence works... noted that the line-seize event returns an error because that's not an event that's registered for... etc. None of it gets me having it do what I want, which is seeing the status of the one on the other, and from watching the SIP I'm not seeing how it could work anyway in any of the configurations I've tried... I assume I am missing something in the dialplan that tells it to notify the others when something happens on the first (given that line-seize doesn't do anything, so the only way to know is if the first either gets a call or places one, which must go via the dialplan), and if that's the case are there any examples of this? Matthew Kaufman From Prometheus001 at gmx.net Fri Jan 9 01:00:10 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 09 Jan 2009 10:00:10 +0100 Subject: [Freeswitch-users] Openzap: every second incoming call fails In-Reply-To: <87f2f3b90901081556t30e42d3bhaaec89e16ed8dff4@mail.gmail.com> References: <4966349F.7090608@gmx.net> <87f2f3b90901081556t30e42d3bhaaec89e16ed8dff4@mail.gmail.com> Message-ID: <4967121A.4010306@gmx.net> Hello Michael, here is a log of 2 calls. The first is one successfull, the second not. Bestr regards Peter 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[5a] Size:[103] CRV: 16 (0x10, CTX: Originator) 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan a001fb20 (1:1) source isdn_data->channels_remote_crv[0x10] 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() Changing state on 1:1 from TERMINATING to DOWN 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE [DOWN] 2009-01-08 17:57:29 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:57:45 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 40 -------------------------------------------------------------------------------- [08 02 00 45 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 34 34 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] 2009-01-08 17:57:45 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[05] Size:[179] CRV: 69 (0x45, CTX: Originator) 2009-01-08 17:57:45 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0x45] 2009-01-08 17:57:45 [WARNING] ozmod_isdn.c:572 zap_isdn_931_34() Channel 2:1 ~ 2:32 is already in use waiting for it to become available. 2009-01-08 17:57:46 [ERR] ozmod_isdn.c:586 zap_isdn_931_34() Channel 2:1 ~ 2:32 is already in use. 2009-01-08 17:57:46 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 -------------------------------------------------------------------------------- [08 02 80 45 45 08 02 81 e5] 2009-01-08 17:57:46 [CRIT] ozmod_isdn.c:644 zap_isdn_931_34() Failed to open channel for new setup message 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 -------------------------------------------------------------------------------- [08 02 00 45 4d] 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[4d] Size:[103] CRV: 69 (0x45, CTX: Originator) 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0x45] 2009-01-08 17:57:47 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received Release with no matching channel 0 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse error [-3012] [Q931E_INVALID_CRV] 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 -------------------------------------------------------------------------------- [08 02 00 45 4d] 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[4d] Size:[103] CRV: 69 (0x45, CTX: Originator) 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0x45] 2009-01-08 17:57:51 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received Release with no matching channel 0 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse error [-3012] [Q931E_INVALID_CRV] 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 40 -------------------------------------------------------------------------------- [08 02 00 11 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 34 34 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[05] Size:[179] CRV: 17 (0x11, CTX: Originator) 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0x11] 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:690 zap_isdn_931_34() Changing state on 1:1 from DOWN to RING 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE [RING] 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() got clear channel sig [START] 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMA 20ms 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:1125 zap_channel_from_event() Connect inbound channel OpenZAP/1:1/21658519 2009-01-08 17:58:10 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/21658519 [87c6dbc8-dda5-11dd-9836-2fb1a1f66971] 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:1129 zap_channel_from_event() (OpenZAP/1:1/21658519) State Change CS_NEW -> CS_INIT 2009-01-08 17:58:10 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/1:1/21658519 [BREAK] 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change CS_INIT 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (OpenZAP/1:1/21658519) State INIT 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:364 channel_on_init() (OpenZAP/1:1/21658519) State Change CS_INIT -> CS_ROUTING 2009-01-08 17:58:10 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/1:1/21658519 [BREAK] 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (OpenZAP/1:1/21658519) State INIT going to sleep 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change CS_ROUTING 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (OpenZAP/1:1/21658519) State ROUTING 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:387 channel_on_routing() OpenZAP/1:1/21658519 CHANNEL ROUTING 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_routing() OpenZAP/1:1/21658519 Standard ROUTING 2009-01-08 17:58:10 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 6934409200->21658519 in context default 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [liverpie_test-caller] destination_number(21658519) =~ /^(50[0-9][0-9])/ 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [liverpie_inform_hangup] destination_number(21658519) =~ /8888/ 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [liverpie_error_inform_hangup] destination_number(21658519) =~ /8887/ 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [park position] destination_number(21658519) =~ /8886/ 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [liverpie_test_consultant] destination_number(21658519) =~ /5002/ 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [liverpie_rejump into state machine] destination_number(21658519) =~ /5004/ 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [liverpie_test-caller_56] destination_number(21658519) =~ /5056/ 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [nb_conferencesfrom external] destination_number(21658519) =~ /^(21658599)$/ 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [Inbound Zaptel] destination_number(21658519) =~ /^(216585[0-9]+)$/ 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:100 switch_core_standard_on_routing() (OpenZAP/1:1/21658519) State Change CS_ROUTING -> CS_EXECUTE 2009-01-08 17:58:10 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/1:1/21658519 [BREAK] 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (OpenZAP/1:1/21658519) State ROUTING going to sleep 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change CS_EXECUTE 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (OpenZAP/1:1/21658519) State EXECUTE 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:404 channel_on_execute() OpenZAP/1:1/21658519 CHANNEL EXECUTE 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:137 switch_core_standard_on_execute() OpenZAP/1:1/21658519 Standard EXECUTE 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute answer() 2009-01-08 17:58:10 [DEBUG] mod_dptools.c:600 answer_function() OpenZAP/1:1/21658519 receive message [ANSWER] 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:750 channel_receive_message_b() Changing state on 1:1 from RING to PROGRESS 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE [PROGRESS] 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 15 -------------------------------------------------------------------------------- [08 02 80 11 02 04 03 80 90 a3 18 03 a1 83 81] 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:753 channel_receive_message_b() Changing state on 1:1 from PROGRESS to PROGRESS_MEDIA 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE [PROGRESS_MEDIA] 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 15 -------------------------------------------------------------------------------- [08 02 80 11 01 04 03 80 90 a3 18 03 a1 83 81] 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:755 channel_receive_message_b() Changing state on 1:1 from PROGRESS_MEDIA to UP 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE [UP] 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 10 -------------------------------------------------------------------------------- [08 02 80 11 07 18 03 a1 83 81] 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:511 switch_core_session_perform_receive_message() Send signal OpenZAP/1:1/21658519 [BREAK] 2009-01-08 17:58:11 [NOTICE] mod_dptools.c:600 answer_function() Channel [OpenZAP/1:1/21658519] has been answered 2009-01-08 17:58:11 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/1:1/21658519 receive message [AUDIO_SYNC] 2009-01-08 17:58:11 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute export(service_number=true) 2009-01-08 17:58:11 [DEBUG] mod_dptools.c:837 export_function() EXPORT [service_number]=[true] 2009-01-08 17:58:11 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute export(sip_secure_media=false) 2009-01-08 17:58:11 [DEBUG] mod_dptools.c:837 export_function() EXPORT [sip_secure_media]=[false] 2009-01-08 17:58:11 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute socket(127.0.0.1:8085 async full) 2009-01-08 17:58:11 [DEBUG] mod_event_socket.c:1797 listener_run() Connection Open 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 14 -------------------------------------------------------------------------------- [08 02 00 11 7d 08 04 82 e3 98 04 14 01 09] 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[7d] Size:[114] CRV: 17 (0x11, CTX: Originator) 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] 2009-01-08 17:58:11 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 14 -------------------------------------------------------------------------------- [08 02 00 11 7d 08 04 82 e3 98 04 14 01 07] 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[7d] Size:[114] CRV: 17 (0x11, CTX: Originator) 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] 2009-01-08 17:58:11 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:694 switch_core_session_queue_private_event() Send signal OpenZAP/1:1/21658519 [BREAK] 2009-01-08 17:58:11 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/1:1/21658519 receive message [AUDIO_SYNC] 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 -------------------------------------------------------------------------------- [08 02 00 11 0f] 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[0f] Size:[103] CRV: 17 (0x11, CTX: Originator) 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:709 zap_isdn_931_34() Received CONNECT_ACK message for channel 0 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:694 switch_core_session_queue_private_event() Send signal OpenZAP/1:1/21658519 [BREAK] 2009-01-08 17:58:11 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/1:1/21658519 receive message [AUDIO_SYNC] 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:694 switch_core_session_queue_private_event() Send signal OpenZAP/1:1/21658519 [BREAK] 2009-01-08 17:58:11 [DEBUG] switch_ivr.c:455 switch_ivr_parse_event() OpenZAP/1:1/21658519 Command Execute read(0 1 customer/hallo.wav interrupt_digit 0 ) 2009-01-08 17:58:11 [DEBUG] switch_ivr_play_say.c:932 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-01-08 17:58:11 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() OpenZAP/1:1/21658519 receive message [TRANSCODING_NECESSARY] 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 13 -------------------------------------------------------------------------------- [08 02 00 11 45 08 02 85 90 1e 02 82 88] 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[45] Size:[115] CRV: 17 (0x11, CTX: Originator) 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:455 zap_isdn_931_34() Changing state on 1:1 from UP to TERMINATING 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE [TERMINATING] 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:1121 state_advance() Terminating: Direction Inbound 2009-01-08 17:58:14 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() got clear channel sig [STOP] 2009-01-08 17:58:14 [NOTICE] mod_openzap.c:1437 on_clear_channel_signal() Hangup OpenZAP/1:1/21658519 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-08 17:58:14 [DEBUG] switch_channel.c:1513 switch_channel_perform_hangup() Send signal OpenZAP/1:1/21658519 [KILL] 2009-01-08 17:58:14 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/1:1/21658519 [BREAK] 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 -------------------------------------------------------------------------------- [08 02 80 11 4d] 2009-01-08 17:58:14 [DEBUG] mod_event_socket.c:1922 listener_run() Session complete, waiting for children 2009-01-08 17:58:14 [DEBUG] mod_event_socket.c:1946 listener_run() Connection Closed 2009-01-08 17:58:14 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (OpenZAP/1:1/21658519) State EXECUTE going to sleep 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change CS_HANGUP 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/1:1/21658519) State HANGUP 2009-01-08 17:58:14 [DEBUG] mod_openzap.c:472 channel_on_hangup() OpenZAP/1:1/21658519 CHANNEL HANGUP 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/1:1/21658519 Standard HANGUP, cause: NORMAL_CLEARING 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/1:1/21658519) State HANGUP going to sleep 2009-01-08 17:58:14 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 46 (OpenZAP/1:1/21658519) Locked, Waiting on external entities 2009-01-08 17:58:14 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 46 (OpenZAP/1:1/21658519) Ended 2009-01-08 17:58:14 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel OpenZAP/1:1/21658519 [CS_HANGUP] 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 -------------------------------------------------------------------------------- [08 02 00 11 5a] 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[5a] Size:[103] CRV: 17 (0x11, CTX: Originator) 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() Changing state on 1:1 from TERMINATING to DOWN 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE [DOWN] 2009-01-08 17:58:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 40 -------------------------------------------------------------------------------- [08 02 00 46 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 34 34 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[05] Size:[179] CRV: 70 (0x46, CTX: Originator) 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0x46] 2009-01-08 17:58:32 [WARNING] ozmod_isdn.c:572 zap_isdn_931_34() Channel 2:1 ~ 2:32 is already in use waiting for it to become available. 2009-01-08 17:58:33 [ERR] ozmod_isdn.c:586 zap_isdn_931_34() Channel 2:1 ~ 2:32 is already in use. 2009-01-08 17:58:33 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 -------------------------------------------------------------------------------- [08 02 80 46 45 08 02 81 e5] 2009-01-08 17:58:33 [CRIT] ozmod_isdn.c:644 zap_isdn_931_34() Failed to open channel for new setup message 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 -------------------------------------------------------------------------------- [08 02 00 46 4d] 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0x46] 2009-01-08 17:58:34 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received Release with no matching channel 0 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse error [-3012] [Q931E_INVALID_CRV] 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 -------------------------------------------------------------------------------- [08 02 00 46 4d] 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0x46] 2009-01-08 17:58:38 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received Release with no matching channel 0 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse error [-3012] [Q931E_INVALID_CRV] To see which channels he actually used in former times I grepped for "channel done" in the log an got the following: Channel 2:1 ~ 2:32 should not be blocked as currently there are no more than 2 concurrent calls while testing (1 incoming and 1 outgoing, we try to spread outgoing over span1 and span2) 2009-01-08 14:16:20 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 14:17:38 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 14:19:26 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 14:22:11 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 14:24:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 14:27:50 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 14:32:40 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 14:39:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 2:24 2009-01-08 14:39:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 14:40:48 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 14:44:20 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 14:46:33 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 14:50:42 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:07:13 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:12:36 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:16:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:18:31 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 2:29 2009-01-08 15:18:33 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:19:21 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:24:20 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:26:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:27:10 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:28:07 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:33:54 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 2:1 2009-01-08 15:35:05 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:39:55 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:48:27 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:51:55 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 15:58:53 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 16:05:41 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 16:09:15 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 16:18:21 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 16:18:57 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 16:19:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:24:07 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 2:8 2009-01-08 17:24:15 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:29:13 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:31:49 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:32:28 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:35:35 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:38:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 2:10 2009-01-08 17:38:50 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:39:44 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:39:58 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:2 2009-01-08 17:40:02 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:40:10 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:40:18 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:40:32 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:40:43 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:40:55 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:41:07 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:42:53 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 2:10 2009-01-08 17:42:53 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:43:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:44:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:56:05 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:56:48 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:57:29 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-08 17:58:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 Michael Collins schrieb: > Can you pastebin a complete call history where the first call works, > gets hung up and then the second call comes in? I would like to see > the entire d-chan trace. > -MC > > On Thu, Jan 8, 2009 at 9:15 AM, Peter P GMX wrote: > >> We have a strange Issue on Openzap with a Digium PRI card (Digium TE220 >> inkl. VPMOCT064 Octasic DSP-based echo cancellation module) >> >> Every second incoming call is successfoll, every second incoming one >> fails. For me it seems as if FS tries to use a channel which may be >> still occupied? >> >> Anybody has an idea? >> >> Last hangup from successful!! call: >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got >> an event! Type:[5a] Size:[103] CRV: 17 (0x11, CTX: Originator) >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan >> a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() Changing >> state on 1:1 from TERMINATING to DOWN >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE >> [DOWN] >> 2009-01-08 17:58:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 40 >> >> >> 2nd Incoming Channel fails >> -------------------------------------------------------------------------------- >> [08 02 00 46 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 34 34 >> 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] >> >> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got >> an event! Type:[05] Size:[179] CRV: 70 (0x46, CTX: Originator) >> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 >> (-1:-1) source isdn_data->channels_remote_crv[0x46] >> 2009-01-08 17:58:32 [WARNING] ozmod_isdn.c:572 zap_isdn_931_34() Channel >> 2:1 ~ 2:32 is already in use waiting for it to become available. >> 2009-01-08 17:58:33 [ERR] ozmod_isdn.c:586 zap_isdn_931_34() Channel 2:1 >> ~ 2:32 is already in use. >> 2009-01-08 17:58:33 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 >> -------------------------------------------------------------------------------- >> [08 02 80 46 45 08 02 81 e5] >> >> 2009-01-08 17:58:33 [CRIT] ozmod_isdn.c:644 zap_isdn_931_34() Failed to >> open channel for new setup message >> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 >> -------------------------------------------------------------------------------- >> [08 02 00 46 4d] >> >> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got >> an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) >> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 >> (-1:-1) source isdn_data->channels_remote_crv[0x46] >> 2009-01-08 17:58:34 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received >> Release with no matching channel 0 >> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse >> error [-3012] [Q931E_INVALID_CRV] >> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 >> -------------------------------------------------------------------------------- >> [08 02 00 46 4d] >> >> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got >> an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) >> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 >> (-1:-1) source isdn_data->channels_remote_crv[0x46] >> 2009-01-08 17:58:38 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received >> Release with no matching channel 0 >> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse >> error [-3012] [Q931E_INVALID_CRV] >> >> >> Best regards Peter >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From timb0311 at hotmail.com Fri Jan 9 02:55:54 2009 From: timb0311 at hotmail.com (Tim B) Date: Fri, 9 Jan 2009 05:55:54 -0500 Subject: [Freeswitch-users] Compile Mod_Managed on Centos 5.2 Message-ID: I want to use C# with freeswitch. How do I go about compiling the mod_managed package on Centos 5.2? The wiki just shows how to compile it on windows. And I try doing a make in the package directory and get errors. Tim _________________________________________________________________ Windows Live? Hotmail?: Chat. Store. Share. Do more with mail. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_hm_justgotbetter_explore_012009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090109/1ceb0582/attachment.html From yudha2008 at gmail.com Fri Jan 9 04:34:02 2009 From: yudha2008 at gmail.com (Baskar) Date: Fri, 9 Jan 2009 18:04:02 +0530 Subject: [Freeswitch-users] Fax Tone Detect Message-ID: *Hi, I want to get the events for the tone detect but i cant able to get any events, Procedures i follow to done detect:* Step 1:I have added the line in default.xml * ===>I have add this line * * I have added the line to detect fax.* Step 2:Then reloaded freeswitch console using reloadxml command reloadxml API CALL [reloadxml()] output: +OK [Success] 2009-01-09 17:49:51 [INFO] mod_enum.c:806 event_handler() ENUM Reloaded 2009-01-09 17:49:51 [INFO] switch_time.c:656 switch_load_timezones() Timezone reloaded 530 definitions Step3:Through x-lite i have dial the fax no *(43951333)* Step4:I have passed events through event socket * event plain DETECTED_TONE* Content-Type: command/reply Reply-Text: +OK event listener enabled plain ====>But i did not get any events in the event socket when i dial the fax no Step 5:I have pasted the freeswitch log in the link * http://pastebin.freeswitch.org/6713* *I need to get the fax tone detect but i cant able to detect the fax tone. If any thing is wrong correct me and help to solve the problem.* -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090109/775474e5/attachment.html From ahgindia308 at gmail.com Fri Jan 9 04:34:47 2009 From: ahgindia308 at gmail.com (ahgindia) Date: Fri, 9 Jan 2009 04:34:47 -0800 (PST) Subject: [Freeswitch-users] Freeswitch not passing more than two calls In-Reply-To: <87f2f3b90901081558u13d08328q97d4c998c45f81ad@mail.gmail.com> References: <21355617.post@talk.nabble.com> <87f2f3b90901081558u13d08328q97d4c998c45f81ad@mail.gmail.com> Message-ID: <21371461.post@talk.nabble.com> How can I make the debug mode on in freeswitch? Can it be done from cli on fly? If I put debug mode on, will it affect performance of the freeswitch, as the freeswitch is currently used as production system. Michael S Collins wrote: > > So the issue is not happening right now? If it is then we would want > you to pastebin the debug output of a few test calls. If not, please > watch to see if this issue happens again and then report it back with > a debug trace. > -MC > > On Thu, Jan 8, 2009 at 8:46 AM, ahgindia wrote: >> >> Hi all, >> I have a freeswitch setup for bridging calls between two gateways. >> i.e. Originator -> Gateway A -> freeswitch -> Gateeway B -> terminator >> user >> I have it running very well for last 43 days. >> But from last one week, I noticed that freeswitch was not able to pass >> more >> than two calls at once. All the other calls were failing with hang up >> cause >> 41 (NORMAL_TEMPORARY_FAILURE). >> The freeswitch was back to normal work only I restarted freeswitch. >> Can anybody suggest any issue or suggestion for this problem. >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-not-passing-more-than-two-calls-tp21355617p21355617.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-not-passing-more-than-two-calls-tp21355617p21371461.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From wiltingtree at gmail.com Fri Jan 9 05:59:57 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Fri, 9 Jan 2009 08:59:57 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 31, Issue 44 In-Reply-To: References: Message-ID: Thanks for the replies. I wrote a script in SpiderMonkey to place a call, and upon connct turn on vmd, play a "press 1" prompt to see if there is a human, and then play some more prompts. If a voicemail beep is heard it starts playing a different prompt. I tested this with almost 300 phone numbers, but I suspect a disproportionate number of these are cell phones. Among the calls not answered by a human (nobody pressed 1), about 60% of the calls failed to recognize a beep. My cell phone is T-Mobile, and it doesn't detect the beep for it. > On 1/8/09, Adam Wilt wrote: > > Hi, I have two issues I'd appreciate some help with. > > > > A) I'm testing VMD and I'm getting a success rate of well under 50%. I > know > > part of the reason is that some of the voicemail beeps it's encountering > are > > very short in length (I've noticed this for T-Mobile and Sprint > voicemails, > > and there may be others too), and it can't detect them. So my question > is > > about the notes in the vmd section of the wiki which states, "The > industry > > standard is 80% detection. This module if used properly should exceeds > the > > standard by a very wide margin". I'm curious about whether I'm using it > > properly, and what I can do to make it work better. Thanks for the help. > > > > B) When I place an outbound call and immediately play a prompt when the > call > > is answered, the prompt sounds garbled to the person answering the phone. > If > > I sleep for a second before playing the prompt, it sounds fine. Any idea > of > > what would cause this? > > > > Thanks, > > Adam > > > > Adam, > > Can I ask how you are "testing" vmd? Where? How? > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090109/ab4efc84/attachment.html From msc at freeswitch.org Fri Jan 9 06:14:02 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 9 Jan 2009 06:14:02 -0800 Subject: [Freeswitch-users] Freeswitch not passing more than two calls In-Reply-To: <21371461.post@talk.nabble.com> References: <21355617.post@talk.nabble.com> <87f2f3b90901081558u13d08328q97d4c998c45f81ad@mail.gmail.com> <21371461.post@talk.nabble.com> Message-ID: <4DD06273-247F-4494-9BB9-CCDB82963B67@freeswitch.org> At the Freeswitch cmd line press F8 or type "console loglevel 7" To go back to normal console messages type "console loglevel 6" The debug message won't degrade your performance. -MC Sent from my iPhone On Jan 9, 2009, at 4:34 AM, ahgindia wrote: > > How can I make the debug mode on in freeswitch? > Can it be done from cli on fly? > If I put debug mode on, will it affect performance of the > freeswitch, as the > freeswitch is currently used as production system. > > > Michael S Collins wrote: >> >> So the issue is not happening right now? If it is then we would want >> you to pastebin the debug output of a few test calls. If not, please >> watch to see if this issue happens again and then report it back with >> a debug trace. >> -MC >> >> On Thu, Jan 8, 2009 at 8:46 AM, ahgindia >> wrote: >>> >>> Hi all, >>> I have a freeswitch setup for bridging calls between two gateways. >>> i.e. Originator -> Gateway A -> freeswitch -> Gateeway B -> >>> terminator >>> user >>> I have it running very well for last 43 days. >>> But from last one week, I noticed that freeswitch was not able to >>> pass >>> more >>> than two calls at once. All the other calls were failing with hang >>> up >>> cause >>> 41 (NORMAL_TEMPORARY_FAILURE). >>> The freeswitch was back to normal work only I restarted freeswitch. >>> Can anybody suggest any issue or suggestion for this problem. >>> -- >>> View this message in context: >>> http://www.nabble.com/Freeswitch-not-passing-more-than-two-calls-tp21355617p21355617.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Freeswitch-not-passing-more-than-two-calls-tp21355617p21371461.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pmhshz at gmail.com Fri Jan 9 06:42:06 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 9 Jan 2009 06:42:06 -0800 (PST) Subject: [Freeswitch-users] Idea/Suggestion required In-Reply-To: <191c3a030901050611r3cc3705fy7691c357903c2cb@mail.gmail.com> References: <21291002.post@talk.nabble.com> <191c3a030901050611r3cc3705fy7691c357903c2cb@mail.gmail.com> Message-ID: <21373597.post@talk.nabble.com> Thanks Anthony, Without answering the call, the 'read' application does, what I wanted, :) Anthony Minessale-2 wrote: > > if you make a dialplan that plays a file but does not answer then it will > happen in the early media stage. > DTMF working during the call is dependent on the other side. We certainly > support it but some carriers purposely do not allow DTMF during progress > since running an ivr in progress stage means you are not paying them any > money for the calls. > > > On Mon, Jan 5, 2009 at 7:27 AM, shehzad p wrote: > >> >> Hi All, >> Is there a way in freeswitch, such that we can play sound and receive a >> DTMF >> from the end user without actually answering a call like in session >> progress >> stage. >> Basically, the system should accept the call and play a progress tone >> during >> call progress (SIP 183). The system should be able to collect DTMF input. >> All this should happen while in SIP 183 (Progress) and the call shouldn't >> change its state to CONNECT, which means, no 200 OK should be issued to >> the >> use via the system. This is more like a IVR based system, however it all >> happens during call progress and not after connect. >> Can anyone suggest any idea about this situation? >> >> Thanks in advance. >> -- >> View this message in context: >> http://www.nabble.com/Idea-Suggestion-required-tp21291002p21291002.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Idea-Suggestion-required-tp21291002p21373597.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From freeswitch-users at lists.rupa.com Fri Jan 9 07:34:35 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Fri, 09 Jan 2009 09:34:35 -0600 Subject: [Freeswitch-users] polycom shared line In-Reply-To: <49670881.7090207@matthew.at> References: <49670881.7090207@matthew.at> Message-ID: <49676E8B.7070601@lists.rupa.com> On 1/9/2009 2:19 AM, Matthew Kaufman wrote: > Can FreeSWITCH and Polycom's line type="shared" work together? If so, > how do I get there from here? [snip] Shared line, I don't know. But if you just want BLF support (see the state of the other line), try the guide I wrote on the wiki. It is for the 320, but seems to work for other polycom phones as well. http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup > Matthew Kaufman > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Prometheus001 at gmx.net Fri Jan 9 07:38:38 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 09 Jan 2009 16:38:38 +0100 Subject: [Freeswitch-users] Openzap: every second incoming call fails In-Reply-To: <4967121A.4010306@gmx.net> References: <4966349F.7090608@gmx.net> <87f2f3b90901081556t30e42d3bhaaec89e16ed8dff4@mail.gmail.com> <4967121A.4010306@gmx.net> Message-ID: <49676F7E.5030805@gmx.net> Hello Michael, sorry for the inconvenience. It turned out that our Telco had to reset the second PRI line. Now it works. Best regards Peter Peter P GMX schrieb: > Hello Michael, > > here is a log of 2 calls. The first is one successfull, the second not. > > Bestr regards > Peter > > 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[5a] Size:[103] CRV: 16 (0x10, CTX: Originator) > 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > a001fb20 (1:1) source isdn_data->channels_remote_crv[0x10] > 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() Changing > state on 1:1 from TERMINATING to DOWN > 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE > [DOWN] > 2009-01-08 17:57:29 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:57:45 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 40 > -------------------------------------------------------------------------------- > [08 02 00 45 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 34 34 > 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] > > 2009-01-08 17:57:45 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[05] Size:[179] CRV: 69 (0x45, CTX: Originator) > 2009-01-08 17:57:45 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 > (-1:-1) source isdn_data->channels_remote_crv[0x45] > 2009-01-08 17:57:45 [WARNING] ozmod_isdn.c:572 zap_isdn_931_34() Channel > 2:1 ~ 2:32 is already in use waiting for it to become available. > 2009-01-08 17:57:46 [ERR] ozmod_isdn.c:586 zap_isdn_931_34() Channel 2:1 > ~ 2:32 is already in use. > 2009-01-08 17:57:46 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 > -------------------------------------------------------------------------------- > [08 02 80 45 45 08 02 81 e5] > > 2009-01-08 17:57:46 [CRIT] ozmod_isdn.c:644 zap_isdn_931_34() Failed to > open channel for new setup message > 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 > -------------------------------------------------------------------------------- > [08 02 00 45 4d] > > 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[4d] Size:[103] CRV: 69 (0x45, CTX: Originator) > 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 > (-1:-1) source isdn_data->channels_remote_crv[0x45] > 2009-01-08 17:57:47 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received > Release with no matching channel 0 > 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse > error [-3012] [Q931E_INVALID_CRV] > 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 > -------------------------------------------------------------------------------- > [08 02 00 45 4d] > > 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[4d] Size:[103] CRV: 69 (0x45, CTX: Originator) > 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 > (-1:-1) source isdn_data->channels_remote_crv[0x45] > 2009-01-08 17:57:51 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received > Release with no matching channel 0 > 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse > error [-3012] [Q931E_INVALID_CRV] > 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 40 > -------------------------------------------------------------------------------- > [08 02 00 11 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 34 34 > 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] > > 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[05] Size:[179] CRV: 17 (0x11, CTX: Originator) > 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 > (-1:-1) source isdn_data->channels_remote_crv[0x11] > 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:690 zap_isdn_931_34() Changing > state on 1:1 from DOWN to RING > 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE > [RING] > 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() > got clear channel sig [START] > 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMA > 20ms > 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:1125 zap_channel_from_event() > Connect inbound channel OpenZAP/1:1/21658519 > 2009-01-08 17:58:10 [NOTICE] switch_channel.c:565 > switch_channel_set_name() New Channel OpenZAP/1:1/21658519 > [87c6dbc8-dda5-11dd-9836-2fb1a1f66971] > 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:1129 zap_channel_from_event() > (OpenZAP/1:1/21658519) State Change CS_NEW -> CS_INIT > 2009-01-08 17:58:10 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > OpenZAP/1:1/21658519 [BREAK] > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change > CS_INIT > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (OpenZAP/1:1/21658519) State INIT > 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:364 channel_on_init() > (OpenZAP/1:1/21658519) State Change CS_INIT -> CS_ROUTING > 2009-01-08 17:58:10 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > OpenZAP/1:1/21658519 [BREAK] > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (OpenZAP/1:1/21658519) State INIT going to sleep > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change > CS_ROUTING > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (OpenZAP/1:1/21658519) State ROUTING > 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:387 channel_on_routing() > OpenZAP/1:1/21658519 CHANNEL ROUTING > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:64 > switch_core_standard_on_routing() OpenZAP/1:1/21658519 Standard ROUTING > 2009-01-08 17:58:10 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 6934409200->21658519 in context default > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: > [liverpie_test-caller] destination_number(21658519) =~ /^(50[0-9][0-9])/ > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex > mismatch > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: > [liverpie_inform_hangup] destination_number(21658519) =~ /8888/ > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex > mismatch > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: > [liverpie_error_inform_hangup] destination_number(21658519) =~ /8887/ > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex > mismatch > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: > [park position] destination_number(21658519) =~ /8886/ > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex > mismatch > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: > [liverpie_test_consultant] destination_number(21658519) =~ /5002/ > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex > mismatch > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: > [liverpie_rejump into state machine] destination_number(21658519) =~ /5004/ > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex > mismatch > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: > [liverpie_test-caller_56] destination_number(21658519) =~ /5056/ > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex > mismatch > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: > [nb_conferencesfrom external] destination_number(21658519) =~ /^(21658599)$/ > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex > mismatch > 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: > [Inbound Zaptel] destination_number(21658519) =~ /^(216585[0-9]+)$/ > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:100 > switch_core_standard_on_routing() (OpenZAP/1:1/21658519) State Change > CS_ROUTING -> CS_EXECUTE > 2009-01-08 17:58:10 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > OpenZAP/1:1/21658519 [BREAK] > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (OpenZAP/1:1/21658519) State ROUTING going to > sleep > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change > CS_EXECUTE > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (OpenZAP/1:1/21658519) State EXECUTE > 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:404 channel_on_execute() > OpenZAP/1:1/21658519 CHANNEL EXECUTE > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:137 > switch_core_standard_on_execute() OpenZAP/1:1/21658519 Standard EXECUTE > 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute answer() > 2009-01-08 17:58:10 [DEBUG] mod_dptools.c:600 answer_function() > OpenZAP/1:1/21658519 receive message [ANSWER] > 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:750 > channel_receive_message_b() Changing state on 1:1 from RING to PROGRESS > 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE > [PROGRESS] > 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 15 > -------------------------------------------------------------------------------- > [08 02 80 11 02 04 03 80 90 a3 18 03 a1 83 81] > > 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:753 > channel_receive_message_b() Changing state on 1:1 from PROGRESS to > PROGRESS_MEDIA > 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE > [PROGRESS_MEDIA] > 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 15 > -------------------------------------------------------------------------------- > [08 02 80 11 01 04 03 80 90 a3 18 03 a1 83 81] > > 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:755 > channel_receive_message_b() Changing state on 1:1 from PROGRESS_MEDIA to UP > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE [UP] > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 10 > -------------------------------------------------------------------------------- > [08 02 80 11 07 18 03 a1 83 81] > > 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:511 > switch_core_session_perform_receive_message() Send signal > OpenZAP/1:1/21658519 [BREAK] > 2009-01-08 17:58:11 [NOTICE] mod_dptools.c:600 answer_function() Channel > [OpenZAP/1:1/21658519] has been answered > 2009-01-08 17:58:11 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() OpenZAP/1:1/21658519 receive message > [AUDIO_SYNC] > 2009-01-08 17:58:11 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute > export(service_number=true) > 2009-01-08 17:58:11 [DEBUG] mod_dptools.c:837 export_function() EXPORT > [service_number]=[true] > 2009-01-08 17:58:11 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute > export(sip_secure_media=false) > 2009-01-08 17:58:11 [DEBUG] mod_dptools.c:837 export_function() EXPORT > [sip_secure_media]=[false] > 2009-01-08 17:58:11 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute > socket(127.0.0.1:8085 async full) > 2009-01-08 17:58:11 [DEBUG] mod_event_socket.c:1797 listener_run() > Connection Open > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 14 > -------------------------------------------------------------------------------- > [08 02 00 11 7d 08 04 82 e3 98 04 14 01 09] > > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[7d] Size:[114] CRV: 17 (0x11, CTX: Originator) > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] > 2009-01-08 17:58:11 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > unhandled message 125 (0x7d) > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 14 > -------------------------------------------------------------------------------- > [08 02 00 11 7d 08 04 82 e3 98 04 14 01 07] > > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[7d] Size:[114] CRV: 17 (0x11, CTX: Originator) > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] > 2009-01-08 17:58:11 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received > unhandled message 125 (0x7d) > 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:694 > switch_core_session_queue_private_event() Send signal > OpenZAP/1:1/21658519 [BREAK] > 2009-01-08 17:58:11 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() OpenZAP/1:1/21658519 receive message > [AUDIO_SYNC] > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 > -------------------------------------------------------------------------------- > [08 02 00 11 0f] > > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[0f] Size:[103] CRV: 17 (0x11, CTX: Originator) > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] > 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:709 zap_isdn_931_34() Received > CONNECT_ACK message for channel 0 > 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:694 > switch_core_session_queue_private_event() Send signal > OpenZAP/1:1/21658519 [BREAK] > 2009-01-08 17:58:11 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() OpenZAP/1:1/21658519 receive message > [AUDIO_SYNC] > 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:694 > switch_core_session_queue_private_event() Send signal > OpenZAP/1:1/21658519 [BREAK] > 2009-01-08 17:58:11 [DEBUG] switch_ivr.c:455 switch_ivr_parse_event() > OpenZAP/1:1/21658519 Command Execute read(0 1 customer/hallo.wav > interrupt_digit 0 ) > 2009-01-08 17:58:11 [DEBUG] switch_ivr_play_say.c:932 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-01-08 17:58:11 [DEBUG] switch_core_io.c:652 > switch_core_session_write_frame() OpenZAP/1:1/21658519 receive message > [TRANSCODING_NECESSARY] > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 13 > -------------------------------------------------------------------------------- > [08 02 00 11 45 08 02 85 90 1e 02 82 88] > > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[45] Size:[115] CRV: 17 (0x11, CTX: Originator) > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:455 zap_isdn_931_34() Changing > state on 1:1 from UP to TERMINATING > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE > [TERMINATING] > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:1121 state_advance() > Terminating: Direction Inbound > 2009-01-08 17:58:14 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() > got clear channel sig [STOP] > 2009-01-08 17:58:14 [NOTICE] mod_openzap.c:1437 > on_clear_channel_signal() Hangup OpenZAP/1:1/21658519 [CS_EXECUTE] > [NORMAL_CLEARING] > 2009-01-08 17:58:14 [DEBUG] switch_channel.c:1513 > switch_channel_perform_hangup() Send signal OpenZAP/1:1/21658519 [KILL] > 2009-01-08 17:58:14 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > OpenZAP/1:1/21658519 [BREAK] > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 > -------------------------------------------------------------------------------- > [08 02 80 11 4d] > > 2009-01-08 17:58:14 [DEBUG] mod_event_socket.c:1922 listener_run() > Session complete, waiting for children > 2009-01-08 17:58:14 [DEBUG] mod_event_socket.c:1946 listener_run() > Connection Closed > 2009-01-08 17:58:14 [DEBUG] switch_ivr_play_say.c:1222 > switch_ivr_play_file() done playing file > 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (OpenZAP/1:1/21658519) State EXECUTE going to > sleep > 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change > CS_HANGUP > 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/1:1/21658519) State HANGUP > 2009-01-08 17:58:14 [DEBUG] mod_openzap.c:472 channel_on_hangup() > OpenZAP/1:1/21658519 CHANNEL HANGUP > 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() OpenZAP/1:1/21658519 Standard HANGUP, > cause: NORMAL_CLEARING > 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/1:1/21658519) State HANGUP going to sleep > 2009-01-08 17:58:14 [DEBUG] switch_core_session.c:939 > switch_core_session_thread() Session 46 (OpenZAP/1:1/21658519) Locked, > Waiting on external entities > 2009-01-08 17:58:14 [NOTICE] switch_core_session.c:957 > switch_core_session_thread() Session 46 (OpenZAP/1:1/21658519) Ended > 2009-01-08 17:58:14 [NOTICE] switch_core_session.c:959 > switch_core_session_thread() Close Channel OpenZAP/1:1/21658519 [CS_HANGUP] > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 > -------------------------------------------------------------------------------- > [08 02 00 11 5a] > > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[5a] Size:[103] CRV: 17 (0x11, CTX: Originator) > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() Changing > state on 1:1 from TERMINATING to DOWN > 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE > [DOWN] > 2009-01-08 17:58:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 40 > -------------------------------------------------------------------------------- > [08 02 00 46 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 34 34 > 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] > > 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[05] Size:[179] CRV: 70 (0x46, CTX: Originator) > 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 > (-1:-1) source isdn_data->channels_remote_crv[0x46] > 2009-01-08 17:58:32 [WARNING] ozmod_isdn.c:572 zap_isdn_931_34() Channel > 2:1 ~ 2:32 is already in use waiting for it to become available. > 2009-01-08 17:58:33 [ERR] ozmod_isdn.c:586 zap_isdn_931_34() Channel 2:1 > ~ 2:32 is already in use. > 2009-01-08 17:58:33 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 > -------------------------------------------------------------------------------- > [08 02 80 46 45 08 02 81 e5] > > 2009-01-08 17:58:33 [CRIT] ozmod_isdn.c:644 zap_isdn_931_34() Failed to > open channel for new setup message > 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 > -------------------------------------------------------------------------------- > [08 02 00 46 4d] > > 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) > 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 > (-1:-1) source isdn_data->channels_remote_crv[0x46] > 2009-01-08 17:58:34 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received > Release with no matching channel 0 > 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse > error [-3012] [Q931E_INVALID_CRV] > 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 > -------------------------------------------------------------------------------- > [08 02 00 46 4d] > > 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) > 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 > (-1:-1) source isdn_data->channels_remote_crv[0x46] > 2009-01-08 17:58:38 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received > Release with no matching channel 0 > 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse > error [-3012] [Q931E_INVALID_CRV] > > To see which channels he actually used in former times I grepped for > "channel done" in the log an got the following: Channel 2:1 ~ 2:32 > should not be blocked as currently there are no more than 2 concurrent > calls while testing (1 incoming and 1 outgoing, we try to spread > outgoing over span1 and span2) > > 2009-01-08 14:16:20 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 14:17:38 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 14:19:26 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 14:22:11 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 14:24:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 14:27:50 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 14:32:40 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 14:39:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 2:24 > 2009-01-08 14:39:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 14:40:48 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 14:44:20 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 14:46:33 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 14:50:42 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:07:13 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:12:36 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:16:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:18:31 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 2:29 > 2009-01-08 15:18:33 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:19:21 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:24:20 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:26:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:27:10 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:28:07 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:33:54 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 2:1 > 2009-01-08 15:35:05 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:39:55 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:48:27 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:51:55 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 15:58:53 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 16:05:41 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 16:09:15 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 16:18:21 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 16:18:57 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 16:19:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:24:07 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 2:8 > 2009-01-08 17:24:15 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:29:13 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:31:49 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:32:28 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:35:35 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:38:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 2:10 > 2009-01-08 17:38:50 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:39:44 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:39:58 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:2 > 2009-01-08 17:40:02 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:40:10 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:40:18 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:40:32 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:40:43 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:40:55 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:41:07 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:42:53 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 2:10 > 2009-01-08 17:42:53 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:43:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:44:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:56:05 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:56:48 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:57:29 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > 2009-01-08 17:58:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 1:1 > > > Michael Collins schrieb: > >> Can you pastebin a complete call history where the first call works, >> gets hung up and then the second call comes in? I would like to see >> the entire d-chan trace. >> -MC >> >> On Thu, Jan 8, 2009 at 9:15 AM, Peter P GMX wrote: >> >> >>> We have a strange Issue on Openzap with a Digium PRI card (Digium TE220 >>> inkl. VPMOCT064 Octasic DSP-based echo cancellation module) >>> >>> Every second incoming call is successfoll, every second incoming one >>> fails. For me it seems as if FS tries to use a channel which may be >>> still occupied? >>> >>> Anybody has an idea? >>> >>> Last hangup from successful!! call: >>> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got >>> an event! Type:[5a] Size:[103] CRV: 17 (0x11, CTX: Originator) >>> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan >>> a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] >>> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() Changing >>> state on 1:1 from TERMINATING to DOWN >>> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 STATE >>> [DOWN] >>> 2009-01-08 17:58:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>> done 1:1 >>> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 40 >>> >>> >>> 2nd Incoming Channel fails >>> -------------------------------------------------------------------------------- >>> [08 02 00 46 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 34 34 >>> 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] >>> >>> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got >>> an event! Type:[05] Size:[179] CRV: 70 (0x46, CTX: Originator) >>> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 >>> (-1:-1) source isdn_data->channels_remote_crv[0x46] >>> 2009-01-08 17:58:32 [WARNING] ozmod_isdn.c:572 zap_isdn_931_34() Channel >>> 2:1 ~ 2:32 is already in use waiting for it to become available. >>> 2009-01-08 17:58:33 [ERR] ozmod_isdn.c:586 zap_isdn_931_34() Channel 2:1 >>> ~ 2:32 is already in use. >>> 2009-01-08 17:58:33 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 >>> -------------------------------------------------------------------------------- >>> [08 02 80 46 45 08 02 81 e5] >>> >>> 2009-01-08 17:58:33 [CRIT] ozmod_isdn.c:644 zap_isdn_931_34() Failed to >>> open channel for new setup message >>> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 >>> -------------------------------------------------------------------------------- >>> [08 02 00 46 4d] >>> >>> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got >>> an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) >>> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 >>> (-1:-1) source isdn_data->channels_remote_crv[0x46] >>> 2009-01-08 17:58:34 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received >>> Release with no matching channel 0 >>> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse >>> error [-3012] [Q931E_INVALID_CRV] >>> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 >>> -------------------------------------------------------------------------------- >>> [08 02 00 46 4d] >>> >>> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got >>> an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) >>> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 >>> (-1:-1) source isdn_data->channels_remote_crv[0x46] >>> 2009-01-08 17:58:38 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received >>> Release with no matching channel 0 >>> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse >>> error [-3012] [Q931E_INVALID_CRV] >>> >>> >>> Best regards Peter >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From ahgindia308 at gmail.com Fri Jan 9 07:51:14 2009 From: ahgindia308 at gmail.com (ahgindia) Date: Fri, 9 Jan 2009 07:51:14 -0800 (PST) Subject: [Freeswitch-users] Freeswitch not passing more than two calls In-Reply-To: <4DD06273-247F-4494-9BB9-CCDB82963B67@freeswitch.org> References: <21355617.post@talk.nabble.com> <87f2f3b90901081558u13d08328q97d4c998c45f81ad@mail.gmail.com> <21371461.post@talk.nabble.com> <4DD06273-247F-4494-9BB9-CCDB82963B67@freeswitch.org> Message-ID: <21374865.post@talk.nabble.com> Here is the log I collected from freeswitch log messages : http://pastebin.freeswitch.org/6714 Please check this and let me know if you find anything. Is it reliable safe to use new release of freeswitch 1.0.2 on the production server? Will there be this issue in that release. Please advice me for the issue. Michael S Collins wrote: > > At the Freeswitch cmd line press F8 or type "console loglevel 7" > > To go back to normal console messages type "console loglevel 6" > > The debug message won't degrade your performance. > > -MC > > Sent from my iPhone > > On Jan 9, 2009, at 4:34 AM, ahgindia wrote: > >> >> How can I make the debug mode on in freeswitch? >> Can it be done from cli on fly? >> If I put debug mode on, will it affect performance of the >> freeswitch, as the >> freeswitch is currently used as production system. >> >> >> Michael S Collins wrote: >>> >>> So the issue is not happening right now? If it is then we would want >>> you to pastebin the debug output of a few test calls. If not, please >>> watch to see if this issue happens again and then report it back with >>> a debug trace. >>> -MC >>> >>> On Thu, Jan 8, 2009 at 8:46 AM, ahgindia >>> wrote: >>>> >>>> Hi all, >>>> I have a freeswitch setup for bridging calls between two gateways. >>>> i.e. Originator -> Gateway A -> freeswitch -> Gateeway B -> >>>> terminator >>>> user >>>> I have it running very well for last 43 days. >>>> But from last one week, I noticed that freeswitch was not able to >>>> pass >>>> more >>>> than two calls at once. All the other calls were failing with hang >>>> up >>>> cause >>>> 41 (NORMAL_TEMPORARY_FAILURE). >>>> The freeswitch was back to normal work only I restarted freeswitch. >>>> Can anybody suggest any issue or suggestion for this problem. >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/Freeswitch-not-passing-more-than-two-calls-tp21355617p21355617.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-not-passing-more-than-two-calls-tp21355617p21371461.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-not-passing-more-than-two-calls-tp21355617p21374865.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Fri Jan 9 07:57:30 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jan 2009 09:57:30 -0600 Subject: [Freeswitch-users] Freeswitch not passing more than two calls In-Reply-To: <21374865.post@talk.nabble.com> References: <21355617.post@talk.nabble.com> <87f2f3b90901081558u13d08328q97d4c998c45f81ad@mail.gmail.com> <21371461.post@talk.nabble.com> <4DD06273-247F-4494-9BB9-CCDB82963B67@freeswitch.org> <21374865.post@talk.nabble.com> Message-ID: <5BA25000-CF3F-4704-B3E0-2336581E1B77@freeswitch.org> Are you not on svn trunk or 1.0.2? If not I would highly recommend you update to that before we move forward. /b On Jan 9, 2009, at 9:51 AM, ahgindia wrote: > > Here is the log I collected from freeswitch log messages : > http://pastebin.freeswitch.org/6714 > Please check this and let me know if you find anything. > > Is it reliable safe to use new release of freeswitch 1.0.2 on the > production > server? Will there be this issue in that release. Please advice me > for the > issue. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090109/421b4765/attachment.html From jmesquita at gmail.com Fri Jan 9 08:05:30 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 9 Jan 2009 14:05:30 -0200 Subject: [Freeswitch-users] XML lib curl - what is the best practice for directory binding? In-Reply-To: <20090105151604.264500@gmx.net> References: <20090105151604.264500@gmx.net> Message-ID: Take a look at the wiki for this module. I have been updating it constantly and there are a lot of new information there. http://wiki.freeswitch.org/wiki/Mod_xml_curl Regards, Mesquita On Jan 5, 2009, at 1:16 PM, can_man at gmx.de wrote: > > Hello, > > I have been looking into the xml curl directory binding and I would > like to update the wiki with the accepted best practice. I have > listed the HTTP POST request I am getting and how I respond. If > there is a better way please let me know and I will update the wiki > accordingly. Btw, what I have done works - so no bug hunting this > time ;-) > I will make a pylons webserver available in the next few days, > starting with dialplan and directory support. > > Thank you, > Phil > > > At boot: > HTTP POST request 1 > > [('hostname', u'voip'), ('section', u'directory'), ('tag_name', > u''), ('key_name', u''), ('key_value', u'')] > > my response: > > >
>
> > > I have left the response empty as I want to provide the users at > runtime. > > ----------------------------------------------------------------------- > At boot: > HTTP POST request 2 > > [('hostname', u'voip'), ('section', u'directory'), ('tag_name', > u''), ('key_name', u''), ('key_value', u'')] > > my response: > > >
>
> > > ----------------------------------------------------------------------- > At boot: > HTTP POST request 3 > > [('hostname', u'voip'), ('section', u'directory'), ('tag_name', > u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'), > ('domain', u'192.168.178.22'), ('purpose', u'network-list')] > > my response: > > >
>
> > > > What is meant by network list here? If all the users should be > loaded at boot time, is this the request which should get a response > with the complete list? > > ---------------------------------------------------------------------- > > During runtime following this action: > > > >
> > > > > > > >
> > > > HTTP POST request: > ('hostname', u'voip'), ('section', u'directory'), ('tag_name', > u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'), > ('mailbox', u'315'), ('key', u'id'), ('user', u'315'), ('domain', > u'192.168.178.22'), ('ip', u'217.10.79.9') > > my response: > > >
> //change to your domain > > > > > > > > > > > > > > > > > > > > > >
> > -- > Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit > allen: http://www.gmx.net/de/go/multimessenger > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Fri Jan 9 09:12:58 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 9 Jan 2009 12:12:58 -0500 Subject: [Freeswitch-users] Hint for DTMF handling in sofia.c In-Reply-To: <2FFC8E58-470D-40D7-B6C3-47198D64FC53@freeswitch.org> References: <4965E784.5030803@ewetel.de> <49661C8A.5020806@ewetel.de> <496628D9.5050805@ewetel.de> <2d9149cd0901081323ka0fb8br536c6b7322962d3e@mail.gmail.com> <2FFC8E58-470D-40D7-B6C3-47198D64FC53@freeswitch.org> Message-ID: <89690A33-09CB-4B65-A382-ABA9476FD81A@jerris.com> Update on this... the latest of anything to speak of in the standards process is that they are killing dtmf-relay. The last draft on it was: http://tools.ietf.org/html/draft-kaplan-sipping-dtmf-package-00 Which says snom is sending it right. We will continue to accept both ways but need to think a bit about how we send. Should we match the draft or the cisco way (cisco by the way was on record saying that info dtmf-relay should die in favor of new specs such as kpml). If anyone wants a good read of what the ietf has standardized in place of info based dtmf, see http://tools.ietf.org/html/rfc4730 Mike On Jan 8, 2009, at 6:38 PM, Brian West wrote: > Snom has already responded to my issue and are going to be providing > me a firmware for testing this tomorrow.. its still going to default > to the WRONG way.. but has a toggle to turn it to the right way. > > /b > > On Jan 8, 2009, at 3:23 PM, Kristian Kielhofner wrote: > >> >> Yes, it should use the decimal encoding from the table you mentioned. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090109/af0b3680/attachment.html From mike at jerris.com Fri Jan 9 09:15:40 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 9 Jan 2009 12:15:40 -0500 Subject: [Freeswitch-users] Trouble getting session.setInputCallback working. In-Reply-To: <3d381e170901081833h6cfe4992saaa96fb23be0a5c3@mail.gmail.com> References: <3d381e170901071603m1943287fwba1ef838678117bb@mail.gmail.com> <3d381e170901071609h3cbefccga01778685952cd60@mail.gmail.com> <87f2f3b90901081601r6e0ffc69m466c9d819dde0030@mail.gmail.com> <3d381e170901081833h6cfe4992saaa96fb23be0a5c3@mail.gmail.com> Message-ID: That is correct, if you want to use the input callback you need to use the session methods (streamFile ?) to play the file. Mike On Jan 8, 2009, at 9:33 PM, Erik Wickstrom wrote: > It seems that it just doesn't work while doing an api call such as > session.execute("playback", "/path/to/file.wav") > > Is this the correct behaviour? Are calls made with execute > "blocking"? From msc at freeswitch.org Fri Jan 9 09:21:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 9 Jan 2009 09:21:52 -0800 Subject: [Freeswitch-users] Compile Mod_Managed on Centos 5.2 In-Reply-To: References: Message-ID: <87f2f3b90901090921w107263e2obd4669fd936b1dfe@mail.gmail.com> Are you using 1.0.2 or the latest SVN? Could you post the content of the errors that you are getting? Thanks, MC On Fri, Jan 9, 2009 at 2:55 AM, Tim B wrote: > I want to use C# with freeswitch. How do I go about compiling the > mod_managed package on Centos 5.2? The wiki just shows how to compile it on > windows. And I try doing a make in the package directory and get errors. > > Tim > > ________________________________ > Windows Live? Hotmail(R): Chat. Store. Share. Do more with mail. Check it out. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mike at jerris.com Fri Jan 9 09:22:02 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 9 Jan 2009 12:22:02 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 31, Issue 44 In-Reply-To: References: Message-ID: Can you get a recording of calling your voicemail and post it online somewhere, I am sure there are some on the list who could tweak to fix this. Mike On Jan 9, 2009, at 8:59 AM, Adam Wilt wrote: > Thanks for the replies. I wrote a script in SpiderMonkey to place a > call, and upon connct turn on vmd, play a "press 1" prompt to see if > there is a human, and then play some more prompts. If a voicemail > beep is heard it starts playing a different prompt. I tested this > with almost 300 phone numbers, but I suspect a disproportionate > number of these are cell phones. Among the calls not answered by a > human (nobody pressed 1), about 60% of the calls failed to recognize > a beep. My cell phone is T-Mobile, and it doesn't detect the beep > for it. > > > > On 1/8/09, Adam Wilt wrote: > > Hi, I have two issues I'd appreciate some help with. > > > > A) I'm testing VMD and I'm getting a success rate of well under > 50%. I know > > part of the reason is that some of the voicemail beeps it's > encountering are > > very short in length (I've noticed this for T-Mobile and Sprint > voicemails, > > and there may be others too), and it can't detect them. So my > question is > > about the notes in the vmd section of the wiki which states, "The > industry > > standard is 80% detection. This module if used properly should > exceeds the > > standard by a very wide margin". I'm curious about whether I'm > using it > > properly, and what I can do to make it work better. Thanks for > the help. > > > > B) When I place an outbound call and immediately play a prompt > when the call > > is answered, the prompt sounds garbled to the person answering the > phone. If > > I sleep for a second before playing the prompt, it sounds fine. > Any idea of > > what would cause this? > > > > Thanks, > > Adam > > > > Adam, > > Can I ask how you are "testing" vmd? Where? How? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090109/9b33911e/attachment-0001.html From matthew at matthew.at Fri Jan 9 10:47:02 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Fri, 09 Jan 2009 10:47:02 -0800 Subject: [Freeswitch-users] polycom shared line In-Reply-To: <49676E8B.7070601@lists.rupa.com> References: <49670881.7090207@matthew.at> <49676E8B.7070601@lists.rupa.com> Message-ID: <49679BA6.9010708@matthew.at> Rupa Schomaker (lists) wrote: > Shared line, I don't know. But if you just want BLF support (see the > state of the other line), try the guide I wrote on the wiki. It is for > the 320, but seems to work for other polycom phones as well. > > http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup > Thanks, I'll play with that and see if I can get enough working to solve my current needs. I think the bigger issue is that when I see "shared line appearance" and "bridged line appearance" in a features list, I think of call appearances with key-system-like shared extensions with busy lamps and barge-in-on-pickup, which I gather requires some broadsoft-authored SIP extensions (including line-seize events) which don't appear to be supported. Of course, the source code's all there... Matthew Kaufman From freeswitch-users at lists.rupa.com Fri Jan 9 10:54:38 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Fri, 09 Jan 2009 12:54:38 -0600 Subject: [Freeswitch-users] polycom shared line In-Reply-To: <49679BA6.9010708@matthew.at> References: <49670881.7090207@matthew.at> <49676E8B.7070601@lists.rupa.com> <49679BA6.9010708@matthew.at> Message-ID: <49679D6E.8030907@lists.rupa.com> On 1/9/2009 12:47 PM, Matthew Kaufman wrote: > Rupa Schomaker (lists) wrote: >> Shared line, I don't know. But if you just want BLF support (see the >> state of the other line), try the guide I wrote on the wiki. It is for >> the 320, but seems to work for other polycom phones as well. >> >> http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup >> > > Thanks, I'll play with that and see if I can get enough working to solve > my current needs. > > I think the bigger issue is that when I see "shared line appearance" and > "bridged line appearance" in a features list, I think of call > appearances with key-system-like shared extensions with busy lamps and > barge-in-on-pickup, which I gather requires some broadsoft-authored SIP > extensions (including line-seize events) which don't appear to be supported. Right. So you'll get the lights with this setup but not full SLA/BLA where you can barge in on the line. I think there are ways to do it with snom phones and perhaps others but I only have one poly so can't do anything in that department for 'em. > > Of course, the source code's all there... > beauty of open source > Matthew Kaufman > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From saigop at gmail.com Fri Jan 9 10:56:58 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Sat, 10 Jan 2009 00:26:58 +0530 Subject: [Freeswitch-users] SIP response code in Freeswitch In-Reply-To: <4468a6770901050130u61ee7602x3b57b6d2495e6fdd@mail.gmail.com> References: <2ea4d47e0901042353v608a0747r51e577e91e22108c@mail.gmail.com> <4468a6770901050130u61ee7602x3b57b6d2495e6fdd@mail.gmail.com> Message-ID: <2ea4d47e0901091056s3ac81404x64f6504b26afd26a@mail.gmail.com> Hi, I was trying to see from the events plain all by following this link http://wiki.freeswitch.org/wiki/Event_list#DETECTED_TONE but cant able to see any detected fax event, since I have dialed a fax number, and also I cant able to get any SIP response code in the event socket console after the command events plain all. Any help would be appreciated. On Mon, Jan 5, 2009 at 3:00 PM, Ognjen Seslija wrote: > Hi, > > there is proto_specific_hangup_cause switch variable you can use for the > cdr i.e. You can also use SIP messages number for a continue_on_fail action > like: > > data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,486,503"/> > Regards, > Ognjen > > > On Mon, Jan 5, 2009 at 8:53 AM, Gopalakrishnan A.N wrote: > >> Hi, >> Is there any possibilities that Freeswitch may detect the SIP response >> code from the IP media gateway. >> >> -- >> Thank you with regards, >> Gopal, >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090110/80090871/attachment.html From jypeng at yahoo.com Fri Jan 9 11:32:12 2009 From: jypeng at yahoo.com (Jian Yuan Peng) Date: Fri, 9 Jan 2009 11:32:12 -0800 (PST) Subject: [Freeswitch-users] 404 error when adding new element to dialplan. Message-ID: <629231.28011.qm@web50610.mail.re2.yahoo.com> FYI, I have post message yesterday, and I did a quick tried on following step to add enum. I tried to use java sip client to make outbound call using gafachi or qwest. So, I added ?? ?????? ?????? ???????? ???????? ???????? ????? ??? First, created new file matchpoint_caller.xml, and put under conf/dialplan, (i also tried it at conf/dialplan/default) directory. For some reason, It never pickup this enum (404 error), when java sip sent the invite. Then remove the matchpoint_caller.xml and add the to conf/dialplan/default.xml. The trick thing is that if put this element above then 404 is gone. If I put it below then i still got 404 error. From timb0311 at hotmail.com Fri Jan 9 11:36:14 2009 From: timb0311 at hotmail.com (Tim B) Date: Fri, 9 Jan 2009 14:36:14 -0500 Subject: [Freeswitch-users] Compile Mod_Managed on Centos 5.2 In-Reply-To: References: Message-ID: MC was using 1.0.2. I don't have access to the machine right now. I can post the errors later tonight or try to build with latest build. Which do you recommend? Tim > > Are you using 1.0.2 or the latest SVN? Could you post the content of> the errors that you are getting?> > Thanks,> MC> > > I want to use C# with freeswitch. How do I go about compiling the> > mod_managed package on Centos 5.2? The wiki just shows how to compile it on> > windows. And I try doing a make in the package directory and get errors.> >> > Tim> > _________________________________________________________________ Windows Live? Hotmail?: Chat. Store. Share. Do more with mail. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_hm_justgotbetter_howitworks_012009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090109/e24b16fd/attachment.html From msc at freeswitch.org Fri Jan 9 11:40:13 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 9 Jan 2009 11:40:13 -0800 Subject: [Freeswitch-users] Compile Mod_Managed on Centos 5.2 In-Reply-To: References: Message-ID: <680B1CD0-7B7C-4E9A-9A36-5DBE6467B7D7@freeswitch.org> Please try latest first. If you still have errors then put them on pastebin. -MC Sent from my iPhone On Jan 9, 2009, at 11:36 AM, Tim B wrote: > MC was using 1.0.2. I don't have access to the machine right now. > I can post the errors later tonight or try to build with latest > build. Which do you recommend? > > Tim > > > > > Are you using 1.0.2 or the latest SVN? Could you post the content of > > the errors that you are getting? > > > > Thanks, > > MC > > > > > I want to use C# with freeswitch. How do I go about compiling the > > > mod_managed package on Centos 5.2? The wiki just shows how to > compile it on > > > windows. And I try doing a make in the package directory and get > errors. > > > > > > Tim > > > > > > Windows Live? Hotmail?: Chat. Store. Share. Do more with mail. See > how it works. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090109/92e3f7ff/attachment.html From Prometheus001 at gmx.net Fri Jan 9 12:03:53 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 09 Jan 2009 21:03:53 +0100 Subject: [Freeswitch-users] different behaviour on uuid_bridge, doesn't really bridge In-Reply-To: <4965D95A.10905@gmx.net> References: <49638DC1.2000307@gmx.net> <4963A228.3000103@gmx.net> <191c3a030901061609m19d91c15o8fad4b54efba017b@mail.gmail.com> <49648B4C.8050505@gmx.net> <191c3a030901071026r3ec5866cy5b1cf2938573e0a0@mail.gmail.com> <4965D95A.10905@gmx.net> Message-ID: <4967ADA9.3020003@gmx.net> I investigated a bit more in this. If I use my event_socket app, it does bridge but event_socket stays in the line. I tested the following scenarios: - bridge openzap with openzap - bridge openzap with sip - bridge sip with sip So it's not an openzap-Issue. Before I bridge, I transfer both legs of the call into park and then bridge via uuid_bridge. Still I have no idea why this is different from a former intstallation. Best regards Peter Peter P GMX schrieb: > Yes, I am doing it via event socket. > On a system with pure SIP it works, but on a system where on both logs > openzap is used, it doesn't work. So maybe it's an openzap problem? > I do the following via event socket: > Leg B is doing the bridge via "uuid_bridge B>" while leg A is playing a soundfile via uuid_playback > I can see on the FS side that FS tries to bridge and then then does an > unbridge: (see also at the end of this mail history) > >> > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message >> > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] >> > > I recognized that my application detects the bridge approach and > initiates the playback of another sound file. This causes, that the > soundfile is played, and so the parties cannot hear each other. I > changed the behaviour of my application, so that it doesn't try to play > another sound file. Now both legs can hear each other. > So the call is bridged, although FS indicates a different message. > > The most important part for my application is though, that for me FS > doesn't really complete the bridge so that event socket is still in the > loop. This keeps 2 connections for event socket open and blocks > resources (I am working with Ruby and eventmachine, there is a limit of > 20 simultaneous connections). > > > Best regards > Peter > > > Anthony Minessale schrieb: > >> how are you exactly doing this? >> with event socket? >> >> can you give a more detailed account of exactly what commands you are >> sending extensions used etc. >> >> >> On Wed, Jan 7, 2009 at 5:00 AM, Peter P GMX > > wrote: >> >> Hello Anthony, >> >> I updated to SNV=11084 and still have the problem. The behaviour is >> slightly changed now. >> Step 4+5 (a s below in my mail) >> 4) When I bridge A and B, A and B canNOT hear each other (e.g. for 1/2 >> sec). A continues to hear its messages, B does not hear anything >> 5) When I hangup B then A is still active and does not recognize >> hangup >> of B. >> >> At former times, when the call was bridged, I had an "unbind" for each >> call on the event_socket interface, so event_socket was out of the >> loop >> after uuid_bridge. Now I have an "unbind" for each party only at the >> time when the party hangs up. >> >> Best regards >> Peter >> >> >> Anthony Minessale schrieb: >> > update one more time and see how that is >> > >> > >> > On Tue, Jan 6, 2009 at 12:25 PM, Peter P GMX >> >> > >> >> wrote: >> > >> > One more info: >> > I have updated to the newest SVN version of FS. >> > A and B can actually hear each other (just a bit, some >> scratching) >> > while >> > the announcement to A is very slow (~50% speed) and very choppy. >> > >> > Best regards >> > peter >> > >> > Peter P GMX schrieb: >> > > I have setup a test machine and a production machine. Since >> > recently the >> > > production machine behaves differently in terms of uuid >> bridge. >> > > >> > > How it should work (and how it worked before) >> > > 1) call A comes in >> > > 2) I play some messages to A >> > > 3) In the meantime I originate a call to B and transfer >> to an >> > > extension, where also some messages are played >> > > 4) Then I bridge A and B, so they are dropped off the >> current >> > > announcements an speak to each other >> > > 5) when either A or B hangs up, both legs are terminated >> > > >> > > New behaviour >> > > 1) call A comes in >> > > 2) I play some messages to A >> > > 3) In the meantime I originate a call to B and transfer >> to an >> > > extension, where also some messages are played >> > > 4) Then I bridge A and B, A and B can hear each other >> for 1/2 sec, >> > > then A constinues to hear its messages, B does not hear >> anything >> > > 5) when either A or B hangs up, both legs are terminated >> > > >> > > 4) is different now! >> > > >> > > The FS console show some messages about unbridge >> > > (SWITCH_MESSAGE_INDICATE_UNBRIDGE), but somehow the calls >> are still >> > > connected until A or B hangs up. >> > > >> > > Anybody has a clue? >> > > Best regards >> > > Peter >> > > >> > > 3) is finished, 4) starts >> > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >> > Yay I got >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) >> > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:352 >> zap_isdn_931_34() zchan >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] >> > > 2009-01-06 17:35:31 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() >> > Received >> > > unhandled message 98 (0x62) >> > > 2009-01-06 17:35:34 [DEBUG] switch_ivr_play_say.c:1222 >> > > switch_ivr_play_file() done playing file >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1094 >> > > switch_ivr_uuid_bridge() (OpenZAP/2:3/49171xxxxxxx) State >> Change >> > > CS_EXECUTE -> CS_RESET >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 >> > > switch_core_session_signal_state_change() Send signal >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1095 >> > > switch_ivr_uuid_bridge() (OpenZAP/2:1/216xxxxx) State Change >> > CS_EXECUTE >> > > -> CS_RESET >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 >> > > switch_core_session_signal_state_change() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_play_say.c:1222 >> > > switch_ivr_play_file() done playing file >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State EXECUTE >> > going to >> > > sleep >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State >> > Change >> > > CS_RESET >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET >> > > 4) Start >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 >> > > uuid_bridge_on_reset() OpenZAP/2:1/216xxxxx CUSTOM RESET >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:53 >> > > switch_core_standard_on_reset() OpenZAP/2:1/216xxxxx >> Standard RESET >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET >> > going to sleep >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State >> > EXECUTE going >> > > to sleep >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) >> Running State >> > > Change CS_RESET >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State >> RESET >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 >> > > uuid_bridge_on_reset() OpenZAP/2:3/49171xxxxxxx CUSTOM RESET >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:533 >> > > uuid_bridge_on_reset() (OpenZAP/2:3/49171xxxxxxx) State Change >> > CS_RESET >> > > -> CS_SOFT_EXECUTE >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 >> > > switch_core_session_signal_state_change() Send signal >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State >> RESET >> > going >> > > to sleep >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) >> Running State >> > > Change CS_SOFT_EXECUTE >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State >> > SOFT_EXECUTE >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 >> > channel_on_soft_execute() >> > > CHANNEL SOFT_EXECUTE >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 >> > > uuid_bridge_on_soft_execute() OpenZAP/2:3/49171xxxxxxx CUSTOM >> > TRANSMIT >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:571 >> > > uuid_bridge_on_soft_execute() (OpenZAP/2:1/216xxxxx) State >> Change >> > > CS_RESET -> CS_SOFT_EXECUTE >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 >> > > switch_core_session_signal_state_change() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State >> > Change >> > > CS_SOFT_EXECUTE >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State >> SOFT_EXECUTE >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 >> > channel_on_soft_execute() >> > > CHANNEL SOFT_EXECUTE >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 >> > > uuid_bridge_on_soft_execute() OpenZAP/2:1/216xxxxx CUSTOM >> TRANSMIT >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:181 >> > > switch_core_standard_on_soft_execute() >> OpenZAP/2:1/216xxxxx Standard >> > > SOFT_EXECUTE >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State >> SOFT_EXECUTE >> > > going to sleep >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:814 >> > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) >> State >> > Change >> > > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 >> > > switch_core_session_signal_state_change() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:856 >> > > switch_ivr_multi_threaded_bridge() OpenZAP/2:1/216xxxxx >> receive >> > message >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:863 >> > > switch_ivr_multi_threaded_bridge() >> OpenZAP/2:3/49171xxxxxxx receive >> > > message [SWITCH_MESSAGE_INDICATE_BRIDGE] >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:667 >> > > switch_core_session_queue_private_event() Send signal >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:907 >> > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) >> State >> > Change >> > > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 >> > > switch_core_session_signal_state_change() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running State >> > Change >> > > CS_EXCHANGE_MEDIA >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:436 >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State >> > EXCHANGE_MEDIA >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:511 >> > > channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:667 >> > > switch_core_session_queue_private_event() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 >> > switch_ivr_parse_event() >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 >> > > custom/warteschleife_30.wav interrupt_digit 0 ) >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_play_say.c:932 >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 >> channels 20ms >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_io.c:610 >> > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive >> > message >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 >> > switch_ivr_parse_event() >> > > OpenZAP/2:3/49171xxxxxxx Command Execute stop_dtmf(true) >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:229 >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:231 >> > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:667 >> > > switch_core_session_queue_private_event() Send signal >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:226 >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:229 >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive message >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:231 >> > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() >> > READ 31 >> > > >> > >> -------------------------------------------------------------------------------- >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7d 02 01 22 30 0d a1 >> 05 30 >> > 03 02 01 >> > > 08 82 01 00 83 01 00] >> > > >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >> > Yay I got >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:352 >> zap_isdn_931_34() zchan >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] >> > > 2009-01-06 17:35:46 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() >> > Received >> > > unhandled message 98 (0x62) >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() >> > READ 31 >> > > >> > >> -------------------------------------------------------------------------------- >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7e 02 01 22 30 0d a1 >> 05 30 >> > 03 02 01 >> > > 09 82 01 00 83 01 00] >> > > >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >> > Yay I got >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:352 >> zap_isdn_931_34() zchan >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] >> > > 2009-01-06 17:36:02 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() >> > Received >> > > unhandled message 98 (0x62) >> > > 2009-01-06 17:36:06 [DEBUG] switch_ivr_play_say.c:1222 >> > > switch_ivr_play_file() done playing file >> > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:229 >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] >> > > 2009-01-06 17:36:07 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:231 >> > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx >> [BREAK] >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:667 >> > > switch_core_session_queue_private_event() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_bridge.c:226 >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr.c:391 >> > switch_ivr_parse_event() >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 >> > > custom/warteschleife_30.wav interrupt_digit 0 ) >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_play_say.c:932 >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 >> channels 20ms >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_io.c:610 >> > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive >> > message >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() >> > READ 31 >> > > >> > >> -------------------------------------------------------------------------------- >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7f 02 01 22 30 0d a1 >> 05 30 >> > 03 02 01 >> > > 0a 82 01 00 83 01 00] >> > > >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >> > Yay I got >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:352 >> zap_isdn_931_34() zchan >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] >> > > 2009-01-06 17:36:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() >> > Received >> > > unhandled message 98 (0x62) >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() >> > READ 32 >> > > >> > >> -------------------------------------------------------------------------------- >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 80 02 01 22 30 0d >> a1 05 >> > 30 03 02 >> > > 01 0b 82 01 00 83 01 00] >> > > >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >> > Yay I got >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:352 >> zap_isdn_931_34() zchan >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] >> > > 2009-01-06 17:36:33 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() >> > Received >> > > unhandled message 98 (0x62) >> > > 2009-01-06 17:36:38 [DEBUG] switch_ivr_play_say.c:1222 >> > > switch_ivr_play_file() done playing file >> > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:229 >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] >> > > 2009-01-06 17:36:39 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:231 >> > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx >> [BREAK] >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:667 >> > > switch_core_session_queue_private_event() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_bridge.c:226 >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr.c:391 >> > switch_ivr_parse_event() >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 >> > > >> > >> custom/dieser_berater_ist_leider_zur_zeit_nicht_erreichbar_bitte_waehlen_sie_einen_ande.wav >> > > interrupt_digit 0 ) >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_play_say.c:932 >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 >> channels 20ms >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_io.c:610 >> > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive >> > message >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:45 [DEBUG] switch_ivr_play_say.c:1222 >> > > switch_ivr_play_file() done playing file >> > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:229 >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] >> > > 2009-01-06 17:36:46 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:231 >> > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx >> [BREAK] >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:667 >> > > switch_core_session_queue_private_event() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_bridge.c:226 >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr.c:391 >> > switch_ivr_parse_event() >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 4 >> > > >> > >> custom/geben_sie_bitte_nach_dem_signalton_die_kennung_des_gewuenschten_beraters_ein_sch2.wav >> > > dtmfdtmf 10000 #,*) >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_play_say.c:932 >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 >> channels 20ms >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_io.c:610 >> > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx receive >> > message >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 >> > > switch_core_session_perform_receive_message() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() >> > READ 32 >> > > >> > >> -------------------------------------------------------------------------------- >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 81 02 01 22 30 0d >> a1 05 >> > 30 03 02 >> > > 01 0c 82 01 00 83 01 00] >> > > >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >> > Yay I got >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:352 >> zap_isdn_931_34() zchan >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] >> > > 2009-01-06 17:36:48 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() >> > Received >> > > unhandled message 98 (0x62) >> > > 2009-01-06 17:37:03 [DEBUG] switch_ivr_play_say.c:1222 >> > > switch_ivr_play_file() done playing file >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() >> > READ 32 >> > > >> > >> -------------------------------------------------------------------------------- >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 82 02 01 22 30 0d >> a1 05 >> > 30 03 02 >> > > 01 0d 82 01 00 83 01 00] >> > > >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >> > Yay I got >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:352 >> zap_isdn_931_34() zchan >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] >> > > 2009-01-06 17:37:04 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() >> > Received >> > > unhandled message 98 (0x62) >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() >> > READ 13 >> > > >> > >> -------------------------------------------------------------------------------- >> > > [08 02 00 35 45 08 02 80 90 1e 02 82 88] >> > > >> > > 5) Hangup >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >> > Yay I got >> > > an event! Type:[45] Size:[115] CRV: 53 (0x35, CTX: Originator) >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:352 >> zap_isdn_931_34() zchan >> > > 77df70 (2:1) source isdn_data->channels_remote_crv[0x35] >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:455 zap_isdn_931_34() >> > Changing >> > > state on 2:1 from UP to TERMINATING >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:813 >> state_advance() 2:1 >> > STATE >> > > [TERMINATING] >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1121 state_advance() >> > > Terminating: Direction Inbound >> > > 2009-01-06 17:37:12 [DEBUG] mod_openzap.c:1418 >> > on_clear_channel_signal() >> > > got clear channel sig [STOP] >> > > 2009-01-06 17:37:12 [NOTICE] mod_openzap.c:1437 >> > > on_clear_channel_signal() Hangup OpenZAP/2:1/216xxxxx >> > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> > > 2009-01-06 17:37:12 [DEBUG] switch_channel.c:1441 >> > > switch_channel_perform_hangup() Send signal >> OpenZAP/2:1/216xxxxx >> > [KILL] >> > > 2009-01-06 17:37:12 [DEBUG] switch_core_session.c:779 >> > > switch_core_session_signal_state_change() Send signal >> > > OpenZAP/2:1/216xxxxx [BREAK] >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() >> WRITE 5 >> > > >> > > >> > > >> > > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 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 >> >> > > > >> > iax:guest at conference.freeswitch.org/888 >> >> > >> > googletalk:conf+888 at conference.freeswitch.org >> >> > > > >> > pstn:213-799-1400 >> > >> ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at 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 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 >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Fri Jan 9 12:11:43 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jan 2009 14:11:43 -0600 Subject: [Freeswitch-users] polycom shared line In-Reply-To: <49679BA6.9010708@matthew.at> References: <49670881.7090207@matthew.at> <49676E8B.7070601@lists.rupa.com> <49679BA6.9010708@matthew.at> Message-ID: <4D5F0333-6929-402D-8252-85546E3544A8@freeswitch.org> We do not do line-seize. Why impose artificial limitations? You can accomplish pickup and such using nothing but dialplan thats how we do it with Snom. /b On Jan 9, 2009, at 12:47 PM, Matthew Kaufman wrote: > Thanks, I'll play with that and see if I can get enough working to > solve > my current needs. > > I think the bigger issue is that when I see "shared line appearance" > and > "bridged line appearance" in a features list, I think of call > appearances with key-system-like shared extensions with busy lamps and > barge-in-on-pickup, which I gather requires some broadsoft-authored > SIP > extensions (including line-seize events) which don't appear to be > supported. > > Of course, the source code's all there... > > Matthew Kaufman From brian at freeswitch.org Fri Jan 9 12:12:47 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jan 2009 14:12:47 -0600 Subject: [Freeswitch-users] 404 error when adding new element to dialplan. In-Reply-To: <629231.28011.qm@web50610.mail.re2.yahoo.com> References: <629231.28011.qm@web50610.mail.re2.yahoo.com> Message-ID: <13D2E62A-0208-462D-838F-9D38C9319D30@freeswitch.org> You have two conditions. the callerid has to be 16508101200 /b On Jan 9, 2009, at 1:32 PM, Jian Yuan Peng wrote: > FYI, > I have post message yesterday, and I did a quick tried on following > step to add enum. > I tried to use java sip client to make outbound call using gafachi > or qwest. So, I added > > expression="^16508101200$"/> > > data="effective_caller_id_number=16508101200"/> > data="effective_caller_id_name=Matchpoint Inc."/> > > > From msc at freeswitch.org Fri Jan 9 12:40:48 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 9 Jan 2009 12:40:48 -0800 Subject: [Freeswitch-users] Openzap: every second incoming call fails In-Reply-To: <49676F7E.5030805@gmx.net> References: <4966349F.7090608@gmx.net> <87f2f3b90901081556t30e42d3bhaaec89e16ed8dff4@mail.gmail.com> <4967121A.4010306@gmx.net> <49676F7E.5030805@gmx.net> Message-ID: <57A9C039-31F8-482B-A358-D6AE8294A31A@freeswitch.org> :) Sent from my iPhone On Jan 9, 2009, at 7:38 AM, Peter P GMX wrote: > Hello Michael, > > sorry for the inconvenience. It turned out that our Telco had to reset > the second PRI line. Now it works. > > Best regards > Peter > > Peter P GMX schrieb: >> Hello Michael, >> >> here is a log of 2 calls. The first is one successfull, the second >> not. >> >> Bestr regards >> Peter >> >> 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[5a] Size:[103] CRV: 16 (0x10, CTX: Originator) >> 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan >> a001fb20 (1:1) source isdn_data->channels_remote_crv[0x10] >> 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() >> Changing >> state on 1:1 from TERMINATING to DOWN >> 2009-01-08 17:57:29 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 >> STATE >> [DOWN] >> 2009-01-08 17:57:29 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:57:45 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ >> 40 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 45 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 >> 34 34 >> 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] >> >> 2009-01-08 17:57:45 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[05] Size:[179] CRV: 69 (0x45, CTX: Originator) >> 2009-01-08 17:57:45 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() >> zchan 0 >> (-1:-1) source isdn_data->channels_remote_crv[0x45] >> 2009-01-08 17:57:45 [WARNING] ozmod_isdn.c:572 zap_isdn_931_34() >> Channel >> 2:1 ~ 2:32 is already in use waiting for it to become available. >> 2009-01-08 17:57:46 [ERR] ozmod_isdn.c:586 zap_isdn_931_34() >> Channel 2:1 >> ~ 2:32 is already in use. >> 2009-01-08 17:57:46 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 80 45 45 08 02 81 e5] >> >> 2009-01-08 17:57:46 [CRIT] ozmod_isdn.c:644 zap_isdn_931_34() >> Failed to >> open channel for new setup message >> 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 45 4d] >> >> 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[4d] Size:[103] CRV: 69 (0x45, CTX: Originator) >> 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() >> zchan 0 >> (-1:-1) source isdn_data->channels_remote_crv[0x45] >> 2009-01-08 17:57:47 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() >> Received >> Release with no matching channel 0 >> 2009-01-08 17:57:47 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 >> parse >> error [-3012] [Q931E_INVALID_CRV] >> 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 45 4d] >> >> 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[4d] Size:[103] CRV: 69 (0x45, CTX: Originator) >> 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() >> zchan 0 >> (-1:-1) source isdn_data->channels_remote_crv[0x45] >> 2009-01-08 17:57:51 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() >> Received >> Release with no matching channel 0 >> 2009-01-08 17:57:51 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 >> parse >> error [-3012] [Q931E_INVALID_CRV] >> 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ >> 40 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 11 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 >> 34 34 >> 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] >> >> 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[05] Size:[179] CRV: 17 (0x11, CTX: Originator) >> 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() >> zchan 0 >> (-1:-1) source isdn_data->channels_remote_crv[0x11] >> 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:690 zap_isdn_931_34() >> Changing >> state on 1:1 from DOWN to RING >> 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 >> STATE >> [RING] >> 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:1418 >> on_clear_channel_signal() >> got clear channel sig [START] >> 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:340 tech_init() Set codec >> PCMA >> 20ms >> 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:1125 >> zap_channel_from_event() >> Connect inbound channel OpenZAP/1:1/21658519 >> 2009-01-08 17:58:10 [NOTICE] switch_channel.c:565 >> switch_channel_set_name() New Channel OpenZAP/1:1/21658519 >> [87c6dbc8-dda5-11dd-9836-2fb1a1f66971] >> 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:1129 >> zap_channel_from_event() >> (OpenZAP/1:1/21658519) State Change CS_NEW -> CS_INIT >> 2009-01-08 17:58:10 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal >> OpenZAP/1:1/21658519 [BREAK] >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change >> CS_INIT >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:444 >> switch_core_session_run() (OpenZAP/1:1/21658519) State INIT >> 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:364 channel_on_init() >> (OpenZAP/1:1/21658519) State Change CS_INIT -> CS_ROUTING >> 2009-01-08 17:58:10 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal >> OpenZAP/1:1/21658519 [BREAK] >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:444 >> switch_core_session_run() (OpenZAP/1:1/21658519) State INIT going >> to sleep >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change >> CS_ROUTING >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:447 >> switch_core_session_run() (OpenZAP/1:1/21658519) State ROUTING >> 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:387 channel_on_routing() >> OpenZAP/1:1/21658519 CHANNEL ROUTING >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:64 >> switch_core_standard_on_routing() OpenZAP/1:1/21658519 Standard >> ROUTING >> 2009-01-08 17:58:10 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing 6934409200->21658519 in context default >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: >> [liverpie_test-caller] destination_number(21658519) =~ /^(50[0-9] >> [0-9])/ >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() >> Regex >> mismatch >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: >> [liverpie_inform_hangup] destination_number(21658519) =~ /8888/ >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() >> Regex >> mismatch >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: >> [liverpie_error_inform_hangup] destination_number(21658519) =~ /8887/ >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() >> Regex >> mismatch >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: >> [park position] destination_number(21658519) =~ /8886/ >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() >> Regex >> mismatch >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: >> [liverpie_test_consultant] destination_number(21658519) =~ /5002/ >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() >> Regex >> mismatch >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: >> [liverpie_rejump into state machine] destination_number(21658519) >> =~ /5004/ >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() >> Regex >> mismatch >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: >> [liverpie_test-caller_56] destination_number(21658519) =~ /5056/ >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() >> Regex >> mismatch >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: >> [nb_conferencesfrom external] destination_number(21658519) =~ / >> ^(21658599)$/ >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:119 parse_exten() >> Regex >> mismatch >> 2009-01-08 17:58:10 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: >> [Inbound Zaptel] destination_number(21658519) =~ /^(216585[0-9]+)$/ >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:100 >> switch_core_standard_on_routing() (OpenZAP/1:1/21658519) State Change >> CS_ROUTING -> CS_EXECUTE >> 2009-01-08 17:58:10 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal >> OpenZAP/1:1/21658519 [BREAK] >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:447 >> switch_core_session_run() (OpenZAP/1:1/21658519) State ROUTING >> going to >> sleep >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change >> CS_EXECUTE >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:454 >> switch_core_session_run() (OpenZAP/1:1/21658519) State EXECUTE >> 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:404 channel_on_execute() >> OpenZAP/1:1/21658519 CHANNEL EXECUTE >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:137 >> switch_core_standard_on_execute() OpenZAP/1:1/21658519 Standard >> EXECUTE >> 2009-01-08 17:58:10 [DEBUG] switch_core_state_machine.c:152 >> switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute >> answer() >> 2009-01-08 17:58:10 [DEBUG] mod_dptools.c:600 answer_function() >> OpenZAP/1:1/21658519 receive message [ANSWER] >> 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:750 >> channel_receive_message_b() Changing state on 1:1 from RING to >> PROGRESS >> 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 >> STATE >> [PROGRESS] >> 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 15 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 80 11 02 04 03 80 90 a3 18 03 a1 83 81] >> >> 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:753 >> channel_receive_message_b() Changing state on 1:1 from PROGRESS to >> PROGRESS_MEDIA >> 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 >> STATE >> [PROGRESS_MEDIA] >> 2009-01-08 17:58:10 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 15 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 80 11 01 04 03 80 90 a3 18 03 a1 83 81] >> >> 2009-01-08 17:58:10 [DEBUG] mod_openzap.c:755 >> channel_receive_message_b() Changing state on 1:1 from >> PROGRESS_MEDIA to UP >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 >> STATE [UP] >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 10 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 80 11 07 18 03 a1 83 81] >> >> 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:511 >> switch_core_session_perform_receive_message() Send signal >> OpenZAP/1:1/21658519 [BREAK] >> 2009-01-08 17:58:11 [NOTICE] mod_dptools.c:600 answer_function() >> Channel >> [OpenZAP/1:1/21658519] has been answered >> 2009-01-08 17:58:11 [DEBUG] switch_channel.c:177 >> switch_channel_audio_sync() OpenZAP/1:1/21658519 receive message >> [AUDIO_SYNC] >> 2009-01-08 17:58:11 [DEBUG] switch_core_state_machine.c:152 >> switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute >> export(service_number=true) >> 2009-01-08 17:58:11 [DEBUG] mod_dptools.c:837 export_function() >> EXPORT >> [service_number]=[true] >> 2009-01-08 17:58:11 [DEBUG] switch_core_state_machine.c:152 >> switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute >> export(sip_secure_media=false) >> 2009-01-08 17:58:11 [DEBUG] mod_dptools.c:837 export_function() >> EXPORT >> [sip_secure_media]=[false] >> 2009-01-08 17:58:11 [DEBUG] switch_core_state_machine.c:152 >> switch_core_standard_on_execute() OpenZAP/1:1/21658519 Execute >> socket(127.0.0.1:8085 async full) >> 2009-01-08 17:58:11 [DEBUG] mod_event_socket.c:1797 listener_run() >> Connection Open >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ >> 14 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 11 7d 08 04 82 e3 98 04 14 01 09] >> >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[7d] Size:[114] CRV: 17 (0x11, CTX: Originator) >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan >> a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] >> 2009-01-08 17:58:11 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() >> Received >> unhandled message 125 (0x7d) >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ >> 14 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 11 7d 08 04 82 e3 98 04 14 01 07] >> >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[7d] Size:[114] CRV: 17 (0x11, CTX: Originator) >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan >> a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] >> 2009-01-08 17:58:11 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() >> Received >> unhandled message 125 (0x7d) >> 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:694 >> switch_core_session_queue_private_event() Send signal >> OpenZAP/1:1/21658519 [BREAK] >> 2009-01-08 17:58:11 [DEBUG] switch_channel.c:177 >> switch_channel_audio_sync() OpenZAP/1:1/21658519 receive message >> [AUDIO_SYNC] >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 11 0f] >> >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[0f] Size:[103] CRV: 17 (0x11, CTX: Originator) >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan >> a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] >> 2009-01-08 17:58:11 [DEBUG] ozmod_isdn.c:709 zap_isdn_931_34() >> Received >> CONNECT_ACK message for channel 0 >> 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:694 >> switch_core_session_queue_private_event() Send signal >> OpenZAP/1:1/21658519 [BREAK] >> 2009-01-08 17:58:11 [DEBUG] switch_channel.c:177 >> switch_channel_audio_sync() OpenZAP/1:1/21658519 receive message >> [AUDIO_SYNC] >> 2009-01-08 17:58:11 [DEBUG] switch_core_session.c:694 >> switch_core_session_queue_private_event() Send signal >> OpenZAP/1:1/21658519 [BREAK] >> 2009-01-08 17:58:11 [DEBUG] switch_ivr.c:455 switch_ivr_parse_event() >> OpenZAP/1:1/21658519 Command Execute read(0 1 customer/hallo.wav >> interrupt_digit 0 ) >> 2009-01-08 17:58:11 [DEBUG] switch_ivr_play_say.c:932 >> switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms >> 2009-01-08 17:58:11 [DEBUG] switch_core_io.c:652 >> switch_core_session_write_frame() OpenZAP/1:1/21658519 receive >> message >> [TRANSCODING_NECESSARY] >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ >> 13 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 11 45 08 02 85 90 1e 02 82 88] >> >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[45] Size:[115] CRV: 17 (0x11, CTX: Originator) >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan >> a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:455 zap_isdn_931_34() >> Changing >> state on 1:1 from UP to TERMINATING >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 >> STATE >> [TERMINATING] >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:1121 state_advance() >> Terminating: Direction Inbound >> 2009-01-08 17:58:14 [DEBUG] mod_openzap.c:1418 >> on_clear_channel_signal() >> got clear channel sig [STOP] >> 2009-01-08 17:58:14 [NOTICE] mod_openzap.c:1437 >> on_clear_channel_signal() Hangup OpenZAP/1:1/21658519 [CS_EXECUTE] >> [NORMAL_CLEARING] >> 2009-01-08 17:58:14 [DEBUG] switch_channel.c:1513 >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/21658519 >> [KILL] >> 2009-01-08 17:58:14 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal >> OpenZAP/1:1/21658519 [BREAK] >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 80 11 4d] >> >> 2009-01-08 17:58:14 [DEBUG] mod_event_socket.c:1922 listener_run() >> Session complete, waiting for children >> 2009-01-08 17:58:14 [DEBUG] mod_event_socket.c:1946 listener_run() >> Connection Closed >> 2009-01-08 17:58:14 [DEBUG] switch_ivr_play_say.c:1222 >> switch_ivr_play_file() done playing file >> 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:454 >> switch_core_session_run() (OpenZAP/1:1/21658519) State EXECUTE >> going to >> sleep >> 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/1:1/21658519) Running State Change >> CS_HANGUP >> 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/1:1/21658519) State HANGUP >> 2009-01-08 17:58:14 [DEBUG] mod_openzap.c:472 channel_on_hangup() >> OpenZAP/1:1/21658519 CHANNEL HANGUP >> 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() OpenZAP/1:1/21658519 Standard >> HANGUP, >> cause: NORMAL_CLEARING >> 2009-01-08 17:58:14 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/1:1/21658519) State HANGUP going >> to sleep >> 2009-01-08 17:58:14 [DEBUG] switch_core_session.c:939 >> switch_core_session_thread() Session 46 (OpenZAP/1:1/21658519) >> Locked, >> Waiting on external entities >> 2009-01-08 17:58:14 [NOTICE] switch_core_session.c:957 >> switch_core_session_thread() Session 46 (OpenZAP/1:1/21658519) Ended >> 2009-01-08 17:58:14 [NOTICE] switch_core_session.c:959 >> switch_core_session_thread() Close Channel OpenZAP/1:1/21658519 >> [CS_HANGUP] >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 11 5a] >> >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[5a] Size:[103] CRV: 17 (0x11, CTX: Originator) >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan >> a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() >> Changing >> state on 1:1 from TERMINATING to DOWN >> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 >> STATE >> [DOWN] >> 2009-01-08 17:58:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ >> 40 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 46 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 33 >> 34 34 >> 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] >> >> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[05] Size:[179] CRV: 70 (0x46, CTX: Originator) >> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() >> zchan 0 >> (-1:-1) source isdn_data->channels_remote_crv[0x46] >> 2009-01-08 17:58:32 [WARNING] ozmod_isdn.c:572 zap_isdn_931_34() >> Channel >> 2:1 ~ 2:32 is already in use waiting for it to become available. >> 2009-01-08 17:58:33 [ERR] ozmod_isdn.c:586 zap_isdn_931_34() >> Channel 2:1 >> ~ 2:32 is already in use. >> 2009-01-08 17:58:33 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 80 46 45 08 02 81 e5] >> >> 2009-01-08 17:58:33 [CRIT] ozmod_isdn.c:644 zap_isdn_931_34() >> Failed to >> open channel for new setup message >> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 46 4d] >> >> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) >> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() >> zchan 0 >> (-1:-1) source isdn_data->channels_remote_crv[0x46] >> 2009-01-08 17:58:34 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() >> Received >> Release with no matching channel 0 >> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 >> parse >> error [-3012] [Q931E_INVALID_CRV] >> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 >> --- >> --- >> --- >> --- >> -------------------------------------------------------------------- >> [08 02 00 46 4d] >> >> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay >> I got >> an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) >> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() >> zchan 0 >> (-1:-1) source isdn_data->channels_remote_crv[0x46] >> 2009-01-08 17:58:38 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() >> Received >> Release with no matching channel 0 >> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 >> parse >> error [-3012] [Q931E_INVALID_CRV] >> >> To see which channels he actually used in former times I grepped for >> "channel done" in the log an got the following: Channel 2:1 ~ 2:32 >> should not be blocked as currently there are no more than 2 >> concurrent >> calls while testing (1 incoming and 1 outgoing, we try to spread >> outgoing over span1 and span2) >> >> 2009-01-08 14:16:20 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 14:17:38 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 14:19:26 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 14:22:11 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 14:24:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 14:27:50 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 14:32:40 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 14:39:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 2:24 >> 2009-01-08 14:39:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 14:40:48 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 14:44:20 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 14:46:33 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 14:50:42 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:07:13 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:12:36 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:16:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:18:31 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 2:29 >> 2009-01-08 15:18:33 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:19:21 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:24:20 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:26:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:27:10 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:28:07 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:33:54 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 2:1 >> 2009-01-08 15:35:05 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:39:55 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:48:27 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:51:55 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 15:58:53 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 16:05:41 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 16:09:15 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 16:18:21 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 16:18:57 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 16:19:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:24:07 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 2:8 >> 2009-01-08 17:24:15 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:29:13 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:31:49 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:32:28 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:35:35 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:38:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 2:10 >> 2009-01-08 17:38:50 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:39:44 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:39:58 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:2 >> 2009-01-08 17:40:02 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:40:10 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:40:18 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:40:32 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:40:43 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:40:55 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:41:07 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:42:53 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 2:10 >> 2009-01-08 17:42:53 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:43:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:44:47 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:56:05 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:56:48 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:57:29 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> 2009-01-08 17:58:14 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 1:1 >> >> >> Michael Collins schrieb: >> >>> Can you pastebin a complete call history where the first call works, >>> gets hung up and then the second call comes in? I would like to see >>> the entire d-chan trace. >>> -MC >>> >>> On Thu, Jan 8, 2009 at 9:15 AM, Peter P GMX >>> wrote: >>> >>> >>>> We have a strange Issue on Openzap with a Digium PRI card (Digium >>>> TE220 >>>> inkl. VPMOCT064 Octasic DSP-based echo cancellation module) >>>> >>>> Every second incoming call is successfoll, every second incoming >>>> one >>>> fails. For me it seems as if FS tries to use a channel which may be >>>> still occupied? >>>> >>>> Anybody has an idea? >>>> >>>> Last hangup from successful!! call: >>>> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >>>> Yay I got >>>> an event! Type:[5a] Size:[103] CRV: 17 (0x11, CTX: Originator) >>>> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() >>>> zchan >>>> a001fb20 (1:1) source isdn_data->channels_remote_crv[0x11] >>>> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() >>>> Changing >>>> state on 1:1 from TERMINATING to DOWN >>>> 2009-01-08 17:58:14 [DEBUG] ozmod_isdn.c:813 state_advance() 1:1 >>>> STATE >>>> [DOWN] >>>> 2009-01-08 17:58:14 [DEBUG] zap_io.c:1125 zap_channel_done() >>>> channel >>>> done 1:1 >>>> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() >>>> READ 40 >>>> >>>> >>>> 2nd Incoming Channel fails >>>> --- >>>> --- >>>> --- >>>> --- >>>> --- >>>> ----------------------------------------------------------------- >>>> [08 02 00 46 05 04 03 80 90 a3 18 03 a1 83 81 6c 0c 21 81 36 39 >>>> 33 34 34 >>>> 30 39 32 30 30 70 09 c1 32 31 36 35 38 35 31 39] >>>> >>>> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >>>> Yay I got >>>> an event! Type:[05] Size:[179] CRV: 70 (0x46, CTX: Originator) >>>> 2009-01-08 17:58:32 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() >>>> zchan 0 >>>> (-1:-1) source isdn_data->channels_remote_crv[0x46] >>>> 2009-01-08 17:58:32 [WARNING] ozmod_isdn.c:572 zap_isdn_931_34() >>>> Channel >>>> 2:1 ~ 2:32 is already in use waiting for it to become available. >>>> 2009-01-08 17:58:33 [ERR] ozmod_isdn.c:586 zap_isdn_931_34() >>>> Channel 2:1 >>>> ~ 2:32 is already in use. >>>> 2009-01-08 17:58:33 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 >>>> --- >>>> --- >>>> --- >>>> --- >>>> --- >>>> ----------------------------------------------------------------- >>>> [08 02 80 46 45 08 02 81 e5] >>>> >>>> 2009-01-08 17:58:33 [CRIT] ozmod_isdn.c:644 zap_isdn_931_34() >>>> Failed to >>>> open channel for new setup message >>>> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() >>>> READ 5 >>>> --- >>>> --- >>>> --- >>>> --- >>>> --- >>>> ----------------------------------------------------------------- >>>> [08 02 00 46 4d] >>>> >>>> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >>>> Yay I got >>>> an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) >>>> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() >>>> zchan 0 >>>> (-1:-1) source isdn_data->channels_remote_crv[0x46] >>>> 2009-01-08 17:58:34 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() >>>> Received >>>> Release with no matching channel 0 >>>> 2009-01-08 17:58:34 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() >>>> 931 parse >>>> error [-3012] [Q931E_INVALID_CRV] >>>> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() >>>> READ 5 >>>> --- >>>> --- >>>> --- >>>> --- >>>> --- >>>> ----------------------------------------------------------------- >>>> [08 02 00 46 4d] >>>> >>>> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() >>>> Yay I got >>>> an event! Type:[4d] Size:[103] CRV: 70 (0x46, CTX: Originator) >>>> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() >>>> zchan 0 >>>> (-1:-1) source isdn_data->channels_remote_crv[0x46] >>>> 2009-01-08 17:58:38 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() >>>> Received >>>> Release with no matching channel 0 >>>> 2009-01-08 17:58:38 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() >>>> 931 parse >>>> error [-3012] [Q931E_INVALID_CRV] >>>> >>>> >>>> Best regards Peter >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Fri Jan 9 12:44:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 9 Jan 2009 14:44:55 -0600 Subject: [Freeswitch-users] different behaviour on uuid_bridge, doesn't really bridge In-Reply-To: <4967ADA9.3020003@gmx.net> References: <49638DC1.2000307@gmx.net> <4963A228.3000103@gmx.net> <191c3a030901061609m19d91c15o8fad4b54efba017b@mail.gmail.com> <49648B4C.8050505@gmx.net> <191c3a030901071026r3ec5866cy5b1cf2938573e0a0@mail.gmail.com> <4965D95A.10905@gmx.net> <4967ADA9.3020003@gmx.net> Message-ID: <191c3a030901091244q726db0e9tab313822c5bcd7be@mail.gmail.com> What I am asking you for is a a step by step exact commands you are doing. Give me the verbatim description of what you do. what exact commands to you send to the socket and exact apps with args you execute in the dialplan. are you on irc, or im so i can ask you more questions live? On Fri, Jan 9, 2009 at 2:03 PM, Peter P GMX wrote: > I investigated a bit more in this. > > If I use my event_socket app, it does bridge but event_socket stays in > the line. I tested the following scenarios: > - bridge openzap with openzap > - bridge openzap with sip > - bridge sip with sip > So it's not an openzap-Issue. > > Before I bridge, I transfer both legs of the call into park and then > bridge via uuid_bridge. > > Still I have no idea why this is different from a former intstallation. > > Best regards Peter > > Peter P GMX schrieb: > > Yes, I am doing it via event socket. > > On a system with pure SIP it works, but on a system where on both logs > > openzap is used, it doesn't work. So maybe it's an openzap problem? > > I do the following via event socket: > > Leg B is doing the bridge via "uuid_bridge > B>" while leg A is playing a soundfile via uuid_playback > > I can see on the FS side that FS tries to bridge and then then does an > > unbridge: (see also at the end of this mail history) > > > >> > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > >> > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > > > I recognized that my application detects the bridge approach and > > initiates the playback of another sound file. This causes, that the > > soundfile is played, and so the parties cannot hear each other. I > > changed the behaviour of my application, so that it doesn't try to play > > another sound file. Now both legs can hear each other. > > So the call is bridged, although FS indicates a different message. > > > > The most important part for my application is though, that for me FS > > doesn't really complete the bridge so that event socket is still in the > > loop. This keeps 2 connections for event socket open and blocks > > resources (I am working with Ruby and eventmachine, there is a limit of > > 20 simultaneous connections). > > > > > > Best regards > > Peter > > > > > > Anthony Minessale schrieb: > > > >> how are you exactly doing this? > >> with event socket? > >> > >> can you give a more detailed account of exactly what commands you are > >> sending extensions used etc. > >> > >> > >> On Wed, Jan 7, 2009 at 5:00 AM, Peter P GMX >> > wrote: > >> > >> Hello Anthony, > >> > >> I updated to SNV=11084 and still have the problem. The behaviour is > >> slightly changed now. > >> Step 4+5 (a s below in my mail) > >> 4) When I bridge A and B, A and B canNOT hear each other (e.g. for > 1/2 > >> sec). A continues to hear its messages, B does not hear anything > >> 5) When I hangup B then A is still active and does not recognize > >> hangup > >> of B. > >> > >> At former times, when the call was bridged, I had an "unbind" for > each > >> call on the event_socket interface, so event_socket was out of the > >> loop > >> after uuid_bridge. Now I have an "unbind" for each party only at the > >> time when the party hangs up. > >> > >> Best regards > >> Peter > >> > >> > >> Anthony Minessale schrieb: > >> > update one more time and see how that is > >> > > >> > > >> > On Tue, Jan 6, 2009 at 12:25 PM, Peter P GMX > >> > >> > >> > >> wrote: > >> > > >> > One more info: > >> > I have updated to the newest SVN version of FS. > >> > A and B can actually hear each other (just a bit, some > >> scratching) > >> > while > >> > the announcement to A is very slow (~50% speed) and very > choppy. > >> > > >> > Best regards > >> > peter > >> > > >> > Peter P GMX schrieb: > >> > > I have setup a test machine and a production machine. Since > >> > recently the > >> > > production machine behaves differently in terms of uuid > >> bridge. > >> > > > >> > > How it should work (and how it worked before) > >> > > 1) call A comes in > >> > > 2) I play some messages to A > >> > > 3) In the meantime I originate a call to B and transfer > >> to an > >> > > extension, where also some messages are played > >> > > 4) Then I bridge A and B, so they are dropped off the > >> current > >> > > announcements an speak to each other > >> > > 5) when either A or B hangs up, both legs are terminated > >> > > > >> > > New behaviour > >> > > 1) call A comes in > >> > > 2) I play some messages to A > >> > > 3) In the meantime I originate a call to B and transfer > >> to an > >> > > extension, where also some messages are played > >> > > 4) Then I bridge A and B, A and B can hear each other > >> for 1/2 sec, > >> > > then A constinues to hear its messages, B does not hear > >> anything > >> > > 5) when either A or B hangs up, both legs are terminated > >> > > > >> > > 4) is different now! > >> > > > >> > > The FS console show some messages about unbridge > >> > > (SWITCH_MESSAGE_INDICATE_UNBRIDGE), but somehow the calls > >> are still > >> > > connected until A or B hangs up. > >> > > > >> > > Anybody has a clue? > >> > > Best regards > >> > > Peter > >> > > > >> > > 3) is finished, 4) starts > >> > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > >> > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:35:31 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:35:34 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1094 > >> > > switch_ivr_uuid_bridge() (OpenZAP/2:3/49171xxxxxxx) State > >> Change > >> > > CS_EXECUTE -> CS_RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1095 > >> > > switch_ivr_uuid_bridge() (OpenZAP/2:1/216xxxxx) State Change > >> > CS_EXECUTE > >> > > -> CS_RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > EXECUTE > >> > going to > >> > > sleep > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running > State > >> > Change > >> > > CS_RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET > >> > > 4) Start > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > >> > > uuid_bridge_on_reset() OpenZAP/2:1/216xxxxx CUSTOM RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:53 > >> > > switch_core_standard_on_reset() OpenZAP/2:1/216xxxxx > >> Standard RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State RESET > >> > going to sleep > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:433 > >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > >> > EXECUTE going > >> > > to sleep > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) > >> Running State > >> > > Change CS_RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > >> RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > >> > > uuid_bridge_on_reset() OpenZAP/2:3/49171xxxxxxx CUSTOM RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:533 > >> > > uuid_bridge_on_reset() (OpenZAP/2:3/49171xxxxxxx) State > Change > >> > CS_RESET > >> > > -> CS_SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:429 > >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > >> RESET > >> > going > >> > > to sleep > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) > >> Running State > >> > > Change CS_SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) State > >> > SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > >> > channel_on_soft_execute() > >> > > CHANNEL SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > >> > > uuid_bridge_on_soft_execute() OpenZAP/2:3/49171xxxxxxx > CUSTOM > >> > TRANSMIT > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:571 > >> > > uuid_bridge_on_soft_execute() (OpenZAP/2:1/216xxxxx) State > >> Change > >> > > CS_RESET -> CS_SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running > State > >> > Change > >> > > CS_SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > >> SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > >> > channel_on_soft_execute() > >> > > CHANNEL SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > >> > > uuid_bridge_on_soft_execute() OpenZAP/2:1/216xxxxx CUSTOM > >> TRANSMIT > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:181 > >> > > switch_core_standard_on_soft_execute() > >> OpenZAP/2:1/216xxxxx Standard > >> > > SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:439 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > >> SOFT_EXECUTE > >> > > going to sleep > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:814 > >> > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) > >> State > >> > Change > >> > > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:856 > >> > > switch_ivr_multi_threaded_bridge() OpenZAP/2:1/216xxxxx > >> receive > >> > message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:863 > >> > > switch_ivr_multi_threaded_bridge() > >> OpenZAP/2:3/49171xxxxxxx receive > >> > > message [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:907 > >> > > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/216xxxxx) > >> State > >> > Change > >> > > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:375 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) Running > State > >> > Change > >> > > CS_EXCHANGE_MEDIA > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_state_machine.c:436 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > >> > EXCHANGE_MEDIA > >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:511 > >> > > channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > >> > switch_ivr_parse_event() > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > >> > > custom/warteschleife_30.wav interrupt_digit 0 ) > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_play_say.c:932 > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > >> channels 20ms > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_io.c:610 > >> > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx > receive > >> > message > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive > message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > >> > switch_ivr_parse_event() > >> > > OpenZAP/2:3/49171xxxxxxx Command Execute stop_dtmf(true) > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:229 > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive > message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:231 > >> > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx > [BREAK] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive > message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:229 > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx receive > message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:231 > >> > > audio_bridge_thread() Send signal OpenZAP/2:1/216xxxxx > [BREAK] > >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 31 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7d 02 01 22 30 0d a1 > >> 05 30 > >> > 03 02 01 > >> > > 08 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:35:46 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 31 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7e 02 01 22 30 0d a1 > >> 05 30 > >> > 03 02 01 > >> > > 09 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:36:02 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:36:06 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:229 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:36:07 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:231 > >> > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx > >> [BREAK] > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr.c:391 > >> > switch_ivr_parse_event() > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > >> > > custom/warteschleife_30.wav interrupt_digit 0 ) > >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_play_say.c:932 > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > >> channels 20ms > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_io.c:610 > >> > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx > receive > >> > message > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 31 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7f 02 01 22 30 0d a1 > >> 05 30 > >> > 03 02 01 > >> > > 0a 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:36:17 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 32 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 80 02 01 22 30 0d > >> a1 05 > >> > 30 03 02 > >> > > 01 0b 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:36:33 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:36:38 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:229 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:36:39 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:231 > >> > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx > >> [BREAK] > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr.c:391 > >> > switch_ivr_parse_event() > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > >> > > > >> > > >> > custom/dieser_berater_ist_leider_zur_zeit_nicht_erreichbar_bitte_waehlen_sie_einen_ande.wav > >> > > interrupt_digit 0 ) > >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_play_say.c:932 > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > >> channels 20ms > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_io.c:610 > >> > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx > receive > >> > message > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:45 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:229 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:36:46 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:231 > >> > > audio_bridge_thread() Send signal OpenZAP/2:3/49171xxxxxxx > >> [BREAK] > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr.c:391 > >> > switch_ivr_parse_event() > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 4 > >> > > > >> > > >> > custom/geben_sie_bitte_nach_dem_signalton_die_kennung_des_gewuenschten_beraters_ein_sch2.wav > >> > > dtmfdtmf 10000 #,*) > >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_play_say.c:932 > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > >> channels 20ms > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_io.c:610 > >> > > switch_core_session_write_frame() OpenZAP/2:1/216xxxxx > receive > >> > message > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 32 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 81 02 01 22 30 0d > >> a1 05 > >> > 30 03 02 > >> > > 01 0c 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:36:48 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:37:03 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 32 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 82 02 01 22 30 0d > >> a1 05 > >> > 30 03 02 > >> > > 01 0d 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: Terminator) > >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:37:04 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 13 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 00 35 45 08 02 80 90 1e 02 82 88] > >> > > > >> > > 5) Hangup > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[45] Size:[115] CRV: 53 (0x35, CTX: > Originator) > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 77df70 (2:1) source isdn_data->channels_remote_crv[0x35] > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:455 > zap_isdn_931_34() > >> > Changing > >> > > state on 2:1 from UP to TERMINATING > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:813 > >> state_advance() 2:1 > >> > STATE > >> > > [TERMINATING] > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1121 > state_advance() > >> > > Terminating: Direction Inbound > >> > > 2009-01-06 17:37:12 [DEBUG] mod_openzap.c:1418 > >> > on_clear_channel_signal() > >> > > got clear channel sig [STOP] > >> > > 2009-01-06 17:37:12 [NOTICE] mod_openzap.c:1437 > >> > > on_clear_channel_signal() Hangup OpenZAP/2:1/216xxxxx > >> > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > >> > > 2009-01-06 17:37:12 [DEBUG] switch_channel.c:1441 > >> > > switch_channel_perform_hangup() Send signal > >> OpenZAP/2:1/216xxxxx > >> > [KILL] > >> > > 2009-01-06 17:37:12 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() > >> WRITE 5 > >> > > > >> > > > >> > > > >> > > > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at 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 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 > >> > > > >> > > >> > >> > >> > iax:guest at conference.freeswitch.org/888 > >> > >> > > >> > googletalk:conf+888 at conference.freeswitch.org > >> > > > >> > > >> > >> > >> > pstn:213-799-1400 > >> > > >> > ------------------------------------------------------------------------ > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at 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 at 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 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 > >> > > > >> iax:guest at conference.freeswitch.org/888 > >> > >> googletalk:conf+888 at conference.freeswitch.org > >> > > > >> pstn:213-799-1400 > >> ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090109/5bbd4813/attachment-0001.html From gmaruzz at celliax.org Fri Jan 9 15:10:01 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 10 Jan 2009 00:10:01 +0100 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax inching forward In-Reply-To: <4ca506420901091341sd916347u820c05c0485d166c@mail.gmail.com> References: <7b197bef0812310658u2cea5f0p7e81ea099af9de83@mail.gmail.com> <4ca506420901091341sd916347u820c05c0485d166c@mail.gmail.com> Message-ID: <7b197bef0901091510i4f86116bw1dba825853f8e5a8@mail.gmail.com> you can call one of the skypiax*, then press 1 :-) Sincerely, Giovanni Maruzzelli ========================================= Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039 On Fri, Jan 9, 2009 at 10:41 PM, Jason Garland wrote: > Can you make one that calls the FS conference bridge? > > On Wed, Dec 31, 2008 at 9:58 AM, Giovanni Maruzzelli > wrote: >> >> Hi FreeSWITCHers! >> >> mod_skypiax, the Skype compatible endpoint, is slowly inching toward >> release :-) >> >> When the demo is online (will go on and off for development), you can >> test it (so helping finding bugs) by calling with Skype the Skype >> Names: >> >> skypiax20, skypiax19, skypiax18, ...., skypiax1 >> >> Happy New Year !!! >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> Company : Celliax >> Website: www.celliax.org >> Address : via Pierlombardo 9, 20135 Milano >> Country/Territory : Italy >> Business Email: gmaruzz at celliax dot org >> Cell : 39-347-2665618 >> Fax : 39-02-87390039 >> >> _______________________________________________ >> 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 > > > _______________________________________________ > 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 matthew at matthew.at Fri Jan 9 19:40:35 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Fri, 09 Jan 2009 19:40:35 -0800 Subject: [Freeswitch-users] polycom shared line In-Reply-To: <4D5F0333-6929-402D-8252-85546E3544A8@freeswitch.org> References: <49670881.7090207@matthew.at> <49676E8B.7070601@lists.rupa.com> <49679BA6.9010708@matthew.at> <4D5F0333-6929-402D-8252-85546E3544A8@freeswitch.org> Message-ID: <496818B3.6080603@matthew.at> Brian West wrote: > We do not do line-seize. Why impose artificial limitations? You can > accomplish pickup and such using nothing but dialplan thats how we do > it with Snom. > Ok, so how do I configure two phones to each have a single extension button that represents the same extension (labeled as such), and which shows status (on that same line button) of the other extension? The BLF examples appear to require that I monitor the busy status of the other extension on something other than the extension line button (something configured as a speed-dial), as far as I can tell. It also seems that to get this to work, each phone needs to be at its own extension but have the label overridden to show the effective extension number that I want them to think they're at, rather than multiple registration of the same extension, correct? (This appears to be true even if true "shared line" support existed, if I'm reading documentation right) Matthew Kaufman From timb0311 at hotmail.com Fri Jan 9 20:14:32 2009 From: timb0311 at hotmail.com (Tim B) Date: Fri, 9 Jan 2009 23:14:32 -0500 Subject: [Freeswitch-users] Compile Mod_Managed on Centos 5.2 In-Reply-To: References: Message-ID: 1. downloaded from svn 2. built freeswitch .. runs fine 3. tried to build mod_managed ... following error: [root at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig [root at phone2 mod_managed]# make make[1]: Entering directory `/usr/src/freeswitch/src/mod/languages/mod_managed' Compiling freeswitch_wrap.cpp... g++ -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/local/lib/pkgconfig/../../include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp /usr/src/freeswitch/src/include/switch_core.h: In function ?void CSharp_switch_core_session_reset(void*, int)?: /usr/src/freeswitch/src/include/switch_core.h:904: error: too few arguments to function ?void switch_core_session_reset(switch_core_session_t*, switch_bool_t, switch_bool_t)? freeswitch_wrap.cpp:6152: error: at this point in file make[1]: *** [freeswitch_wrap.o] Error 1 make[1]: Leaving directory `/usr/src/freeswitch/src/mod/languages/mod_managed' make: *** [all] Error 1 _________________________________________________________________ Windows Live? Hotmail?: Chat. Store. Share. Do more with mail. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_hm_justgotbetter_explore_012009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090109/4038d752/attachment.html From jgarland at jasongarland.com Fri Jan 9 13:41:41 2009 From: jgarland at jasongarland.com (Jason Garland) Date: Fri, 9 Jan 2009 16:41:41 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax inching forward In-Reply-To: <7b197bef0812310658u2cea5f0p7e81ea099af9de83@mail.gmail.com> References: <7b197bef0812310658u2cea5f0p7e81ea099af9de83@mail.gmail.com> Message-ID: <4ca506420901091341sd916347u820c05c0485d166c@mail.gmail.com> Can you make one that calls the FS conference bridge? On Wed, Dec 31, 2008 at 9:58 AM, Giovanni Maruzzelli wrote: > Hi FreeSWITCHers! > > mod_skypiax, the Skype compatible endpoint, is slowly inching toward > release :-) > > When the demo is online (will go on and off for development), you can > test it (so helping finding bugs) by calling with Skype the Skype > Names: > > skypiax20, skypiax19, skypiax18, ...., skypiax1 > > Happy New Year !!! > > Sincerely, > > Giovanni Maruzzelli > ========================================= > Company : Celliax > Website: www.celliax.org > Address : via Pierlombardo 9, 20135 Milano > Country/Territory : Italy > Business Email: gmaruzz at celliax dot org > Cell : 39-347-2665618 > Fax : 39-02-87390039 > > _______________________________________________ > 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-users/attachments/20090109/8a373698/attachment.html From andrew at hijacked.us Fri Jan 9 15:06:50 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 9 Jan 2009 18:06:50 -0500 Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform Message-ID: <20090109230650.GF5210@hijacked.us> Hi, Today I'd like to announce the open-sourcing of a distributed callcenter platform I've been designing/building using FreeSWITCH/Erlang. The goal is to allow multiple callcenter branch offices to operate seamlessly as a whole, or even just scale one large one beyond hardware/software limits by partitioning the inbound lines/agents across multiple servers. You can read more about the design here: http://opencsm.org/wiki/index.php/Spice_Telephony And a quick overview of planned features: * Inbound/Outbound support * Media type agnostic (voice, email, chat, video(?)) * Skill based routing * Agent phone agnostic (SIP or POTS line) * Dynamic wrapup time * Web or fat client interface And you can download a very early work in progress snapshot of the code from http://opencsm.org . We're also open sourcing our customer service management application too, which is also available for download there. If anyone is interested in callcenter development using FreeSWITCH and/or Erlang, we welcome participation at any level. We don't have public source control or a bug tracker yet, but please bear with us; they're on the way. Thanks, Andrew Thompson - opencsm.org From mike at jerris.com Fri Jan 9 21:10:15 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 10 Jan 2009 00:10:15 -0500 Subject: [Freeswitch-users] Compile Mod_Managed on Centos 5.2 In-Reply-To: References: Message-ID: <93769c20901092110j5a7b3078i5eaf7b6f93b12ce7@mail.gmail.com> This is broken from a change that just went in this afternoon.. I will fix it shortly. Mike On Fri, Jan 9, 2009 at 11:14 PM, Tim B wrote: > 1. downloaded from svn > 2. built freeswitch .. runs fine > 3. tried to build mod_managed ... following error: > > [root at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig > [root at phone2 mod_managed]# make > make[1]: Entering directory > `/usr/src/freeswitch/src/mod/languages/mod_managed' > Compiling freeswitch_wrap.cpp... > g++ -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -g -O2 -D_GNU_SOURCE > -D_REENTRANT -pthread -I/usr/local/lib/pkgconfig/../../include/mono-1.0 > -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o > freeswitch_wrap.o freeswitch_wrap.cpp > /usr/src/freeswitch/src/include/switch_core.h: In function 'void > CSharp_switch_core_session_reset(void*, int)': > /usr/src/freeswitch/src/include/switch_core.h:904: error: too few arguments > to function 'void switch_core_session_reset(switch_core_session_t*, > switch_bool_t, switch_bool_t)' > freeswitch_wrap.cpp:6152: error: at this point in file > make[1]: *** [freeswitch_wrap.o] Error 1 > make[1]: Leaving directory > `/usr/src/freeswitch/src/mod/languages/mod_managed' > make: *** [all] Error 1 > > > > ------------------------------ > Windows Live? Hotmail(R): Chat. Store. Share. Do more with mail. Check it > out. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090110/49602d93/attachment.html From msc at freeswitch.org Fri Jan 9 21:15:16 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 9 Jan 2009 21:15:16 -0800 Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform In-Reply-To: <20090109230650.GF5210@hijacked.us> References: <20090109230650.GF5210@hijacked.us> Message-ID: Dude, you ROCK! -MC Sent from my iPhone On Jan 9, 2009, at 3:06 PM, Andrew Thompson wrote: > Hi, > > Today I'd like to announce the open-sourcing of a distributed > callcenter platform I've been designing/building using > FreeSWITCH/Erlang. The goal is to allow multiple callcenter branch > offices to operate seamlessly as a whole, or even just scale one large > one beyond hardware/software limits by partitioning the inbound > lines/agents across multiple servers. > > You can read more about the design here: > > http://opencsm.org/wiki/index.php/Spice_Telephony > > And a quick overview of planned features: > > * Inbound/Outbound support > * Media type agnostic (voice, email, chat, video(?)) > * Skill based routing > * Agent phone agnostic (SIP or POTS line) > * Dynamic wrapup time > * Web or fat client interface > > And you can download a very early work in progress snapshot of the > code > from http://opencsm.org . We're also open sourcing our customer > service > management application too, which is also available for download > there. > > If anyone is interested in callcenter development using FreeSWITCH > and/or Erlang, we welcome participation at any level. We don't have > public source control or a bug tracker yet, but please bear with us; > they're on the way. > > Thanks, > > Andrew Thompson - opencsm.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ahgindia308 at gmail.com Sat Jan 10 01:34:12 2009 From: ahgindia308 at gmail.com (ahgindia) Date: Sat, 10 Jan 2009 01:34:12 -0800 (PST) Subject: [Freeswitch-users] Re ad application not working as intended Message-ID: <21386369.post@talk.nabble.com> Hi all, I want have DTMF input from user without answering the call (i.e. in session progress). I used Read application for this, but it is not working as it is intended to be. It is only accepting one DTMF from the user and the channel gets hang up. Am I missing something? Here is the dialplan that I used for this: The phrase application is used to check user input. -- View this message in context: http://www.nabble.com/Read-application-not-working-as-intended-tp21386369p21386369.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ahgindia308 at gmail.com Sat Jan 10 03:01:13 2009 From: ahgindia308 at gmail.com (ahgindia) Date: Sat, 10 Jan 2009 03:01:13 -0800 (PST) Subject: [Freeswitch-users] Freeswitch crashed !!! Message-ID: <21386948.post@talk.nabble.com> Hi All, Recently I was testing the new freeswitch release 1.0.2 The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo CPU E7200 @ 2.53GHz processor. But it crashed, when there were 96 active calls in it (as can be seen from "show calls" on freeswitch cli) There is a dump file for it, in the folder from where i started the freeswitch. Let me know how can we know the cause of the crash. -- View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21386948.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Sat Jan 10 07:53:32 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 10 Jan 2009 09:53:32 -0600 Subject: [Freeswitch-users] Re ad application not working as intended In-Reply-To: <21386369.post@talk.nabble.com> References: <21386369.post@talk.nabble.com> Message-ID: <654BDDF5-750D-4851-90A9-FEF9E1C8DAA8@freeswitch.org> You need to use execute_extension so that ${res} will be expanded. FreeSWITCH doesn't expand vars as it goes like asterisk does. Easy way to see the var is to use the info app. /b Sent from my iPhone On Jan 10, 2009, at 3:34 AM, ahgindia wrote: > > Hi all, > I want have DTMF input from user without answering the call (i.e. in > session > progress). > I used Read application for this, but it is not working as it is > intended to > be. > It is only accepting one DTMF from the user and the channel gets > hang up. > Am I missing something? > Here is the dialplan that I used for this: > > > The phrase application is used to check user input. > -- > View this message in context: http://www.nabble.com/Read-application-not-working-as-intended-tp21386369p21386369.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Sat Jan 10 07:55:36 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 10 Jan 2009 09:55:36 -0600 Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <21386948.post@talk.nabble.com> References: <21386948.post@talk.nabble.com> Message-ID: <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> What is the output of the backtrace? Can you include that in your email? /b On Jan 10, 2009, at 5:01 AM, ahgindia wrote: > > Hi All, > Recently I was testing the new freeswitch release 1.0.2 > The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo > CPU E7200 > @ 2.53GHz processor. > But it crashed, when there were 96 active calls in it (as can be > seen from > "show calls" on freeswitch cli) > There is a dump file for it, in the folder from where i started the > freeswitch. > Let me know how can we know the cause of the crash. > -- > View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21386948.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Sat Jan 10 08:01:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 10 Jan 2009 10:01:58 -0600 Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <21386948.post@talk.nabble.com> References: <21386948.post@talk.nabble.com> Message-ID: <191c3a030901100801vaf6680bicd380cf9e8610e73@mail.gmail.com> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Loading_FreeSWITCH_in_GDB please get a bt and file a jira at http://jira.freeswitch.org On Sat, Jan 10, 2009 at 5:01 AM, ahgindia wrote: > > Hi All, > Recently I was testing the new freeswitch release 1.0.2 > The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo CPU E7200 > @ 2.53GHz processor. > But it crashed, when there were 96 active calls in it (as can be seen from > "show calls" on freeswitch cli) > There is a dump file for it, in the folder from where i started the > freeswitch. > Let me know how can we know the cause of the crash. > -- > View this message in context: > http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21386948.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090110/cebedf97/attachment.html From brian at freeswitch.org Sat Jan 10 08:41:17 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 10 Jan 2009 10:41:17 -0600 Subject: [Freeswitch-users] Re ad application not working as intended In-Reply-To: <21386369.post@talk.nabble.com> References: <21386369.post@talk.nabble.com> Message-ID: <6BB809A9-FD0E-4434-A4E1-7F8B36435871@freeswitch.org> Actually you can do this... I told you a bit wrong! If you'll escape the ${res} like \${res} it will do what you want. /b On Jan 10, 2009, at 3:34 AM, ahgindia wrote: > > From anthony.minessale at gmail.com Sat Jan 10 08:43:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 10 Jan 2009 10:43:39 -0600 Subject: [Freeswitch-users] Re ad application not working as intended In-Reply-To: <654BDDF5-750D-4851-90A9-FEF9E1C8DAA8@freeswitch.org> References: <21386369.post@talk.nabble.com> <654BDDF5-750D-4851-90A9-FEF9E1C8DAA8@freeswitch.org> Message-ID: <191c3a030901100843j7de39a8dkb47642185de2b90b@mail.gmail.com> escape the $ in ${res} with a \ so it will not be evaluated by the dialplan. On Sat, Jan 10, 2009 at 9:53 AM, Brian West wrote: > You need to use execute_extension so that ${res} will be expanded. > FreeSWITCH doesn't expand vars as it goes like asterisk does. > > Easy way to see the var is to use the info app. > /b > > Sent from my iPhone > > On Jan 10, 2009, at 3:34 AM, ahgindia wrote: > > > > > Hi all, > > I want have DTMF input from user without answering the call (i.e. in > > session > > progress). > > I used Read application for this, but it is not working as it is > > intended to > > be. > > It is only accepting one DTMF from the user and the channel gets > > hang up. > > Am I missing something? > > Here is the dialplan that I used for this: > > > > > > The phrase application is used to check user input. > > -- > > View this message in context: > http://www.nabble.com/Read-application-not-working-as-intended-tp21386369p21386369.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090110/70da6436/attachment.html From timb0311 at hotmail.com Sat Jan 10 09:50:07 2009 From: timb0311 at hotmail.com (Tim B) Date: Sat, 10 Jan 2009 12:50:07 -0500 Subject: [Freeswitch-users] Compile Mod_Managed on Centos 5.2 In-Reply-To: References: Message-ID: Thanks. I have a possible workaround I was thinking about... Could mod_managed just be compiled on Windows and the dlls moved to the Centos 5.2 machine with Mono installed? I already have the Mono base framework 2.0.1 compiled and installed... seems to be working. If I can, where should I put the mod_managed binaries on the Centos machine? Tim > > This is broken from a change that just went in this afternoon.. I will fix> it shortly.> > Mike> > On Fri, Jan 9, 2009 at 11:14 PM, Tim B wrote:> > > 1. downloaded from svn> > 2. built freeswitch .. runs fine> > 3. tried to build mod_managed ... following error:> >> > [root at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig> > [root at phone2 mod_managed]# make> > make[1]: Entering directory> > `/usr/src/freeswitch/src/mod/languages/mod_managed'> > Compiling freeswitch_wrap.cpp...> > g++ -I/usr/src/freeswitch/src/include> > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -g -O2 -D_GNU_SOURCE> > -D_REENTRANT -pthread -I/usr/local/lib/pkgconfig/../../include/mono-1.0> > -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o> > freeswitch_wrap.o freeswitch_wrap.cpp> > /usr/src/freeswitch/src/include/switch_core.h: In function 'void> > CSharp_switch_core_session_reset(void*, int)':> > /usr/src/freeswitch/src/include/switch_core.h:904: error: too few arguments> > to function 'void switch_core_session_reset(switch_core_session_t*,> > switch_bool_t, switch_bool_t)'> > freeswitch_wrap.cpp:6152: error: at this point in file> > make[1]: *** [freeswitch_wrap.o] Error 1> > make[1]: Leaving directory> > `/usr/src/freeswitch/src/mod/languages/mod_managed'> > make: *** [all] Error 1> >> > _________________________________________________________________ Windows Live?: Keep your life in sync. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_012009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090110/583158a4/attachment-0001.html From mgg at giagnocavo.net Sat Jan 10 10:21:39 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sat, 10 Jan 2009 13:21:39 -0500 Subject: [Freeswitch-users] Compile Mod_Managed on Centos 5.2 In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C670235BBAAF6@mse17be1.mse17.exchange.ms> That'll only work for the managed assemblies. The actual native mod_managed.so / dll are different. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim B Sent: Saturday, January 10, 2009 10:50 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Compile Mod_Managed on Centos 5.2 Thanks. I have a possible workaround I was thinking about... Could mod_managed just be compiled on Windows and the dlls moved to the Centos 5.2 machine with Mono installed? I already have the Mono base framework 2.0.1 compiled and installed... seems to be working. If I can, where should I put the mod_managed binaries on the Centos machine? Tim > > This is broken from a change that just went in this afternoon.. I will fix > it shortly. > > Mike > > On Fri, Jan 9, 2009 at 11:14 PM, Tim B wrote: > > > 1. downloaded from svn > > 2. built freeswitch .. runs fine > > 3. tried to build mod_managed ... following error: > > > > [root at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig > > [root at phone2 mod_managed]# make > > make[1]: Entering directory > > `/usr/src/freeswitch/src/mod/languages/mod_managed' > > Compiling freeswitch_wrap.cpp... > > g++ -I/usr/src/freeswitch/src/include > > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -g -O2 -D_GNU_SOURCE > > -D_REENTRANT -pthread -I/usr/local/lib/pkgconfig/../../include/mono-1.0 > > -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o > > freeswitch_wrap.o freeswitch_wrap.cpp > > /usr/src/freeswitch/src/include/switch_core.h: In function 'void > > CSharp_switch_core_session_reset(void*, int)': > > /usr/src/freeswitch/src/include/switch_core.h:904: error: too few arguments > > to function 'void switch_core_session_reset(switch_core_session_t*, > > switch_bool_t, switch_bool_t)' > > freeswitch_wrap.cpp:6152: error: at this point in file > > make[1]: *** [freeswitch_wrap.o] Error 1 > > make[1]: Leaving directory > > `/usr/src/freeswitch/src/mod/languages/mod_managed' > > make: *** [all] Error 1 > > > > ________________________________ Windows Live(tm): Keep your life in sync. See how it works. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090110/4498e1e0/attachment.html From frnkblk at iname.com Sat Jan 10 11:54:20 2009 From: frnkblk at iname.com (Frank Bulk) Date: Sat, 10 Jan 2009 13:54:20 -0600 Subject: [Freeswitch-users] Newbie question: plain FreeSWITCH or SIPfoundry Message-ID: I work for a small service provider and would like to test how well FreeSWITCH's software works against our Nortel CS-1500 to provide conferencing services (for 5 to 100 legs). I'm not looking to provide PBX functionality. Reading through the SIPfoundry 'stuff' it appears that SIPfoundry makes it easy to configure and manage FreeSWITCH, but perhaps that would be making it more complicated than necessary. Any advice? Frank From Prometheus001 at gmx.net Sun Jan 11 11:22:22 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 11 Jan 2009 20:22:22 +0100 Subject: [Freeswitch-users] different behaviour on uuid_bridge, doesn't really bridge In-Reply-To: <191c3a030901091244q726db0e9tab313822c5bcd7be@mail.gmail.com> References: <49638DC1.2000307@gmx.net> <4963A228.3000103@gmx.net> <191c3a030901061609m19d91c15o8fad4b54efba017b@mail.gmail.com> <49648B4C.8050505@gmx.net> <191c3a030901071026r3ec5866cy5b1cf2938573e0a0@mail.gmail.com> <4965D95A.10905@gmx.net> <4967ADA9.3020003@gmx.net> <191c3a030901091244q726db0e9tab313822c5bcd7be@mail.gmail.com> Message-ID: <496A46EE.9070303@gmx.net> Hello Anthony, let me describe what I do: On leg A I play a sound via mod_eventsocket thruogh the read application to make this sound interruptable call-command: execute execute-app-name: read_int execute-app-arg: 0 1 wava.wav interrupt_digit 0 event-lock:true On leg B I play a sound via mod_eventsocket thruogh the read application to make this sound interruptable call-command: execute execute-app-name: read_int execute-app-arg: 0 1 wavb.wav interrupt_digit 0 event-lock:true When this sound is finished, I do a uuid bridge via xmlrpc freeswitch.apibgapiuuid_bridge 425e65b0-e011-11dd-9580-3fea958c6de3 344b23aa-e011-11dd-9580-3fea958c6de3 Then I disable dtmf with dtmf_stop via xmlrpc I ngreped this part of the session and have put this + the freeswitch debug output to the pastebin: http://pastebin.freeswitch.org/6742 Do you need mor info? Best regards Peter Anthony Minessale schrieb: > What I am asking you for is a a step by step exact commands you are doing. > > Give me the verbatim description of what you do. > what exact commands to you send to the socket and exact apps with args > you execute in the dialplan. > > are you on irc, or im so i can ask you more questions live? > > > > On Fri, Jan 9, 2009 at 2:03 PM, Peter P GMX > wrote: > > I investigated a bit more in this. > > If I use my event_socket app, it does bridge but event_socket stays in > the line. I tested the following scenarios: > - bridge openzap with openzap > - bridge openzap with sip > - bridge sip with sip > So it's not an openzap-Issue. > > Before I bridge, I transfer both legs of the call into park and then > bridge via uuid_bridge. > > Still I have no idea why this is different from a former > intstallation. > > Best regards Peter > > Peter P GMX schrieb: > > Yes, I am doing it via event socket. > > On a system with pure SIP it works, but on a system where on > both logs > > openzap is used, it doesn't work. So maybe it's an openzap problem? > > I do the following via event socket: > > Leg B is doing the bridge via "uuid_bridge > > B>" while leg A is playing a soundfile via uuid_playback > > I can see on the FS side that FS tries to bridge and then then > does an > > unbridge: (see also at the end of this mail history) > > > >> > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > >> > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > > > I recognized that my application detects the bridge approach and > > initiates the playback of another sound file. This causes, that the > > soundfile is played, and so the parties cannot hear each other. I > > changed the behaviour of my application, so that it doesn't try > to play > > another sound file. Now both legs can hear each other. > > So the call is bridged, although FS indicates a different message. > > > > The most important part for my application is though, that for me FS > > doesn't really complete the bridge so that event socket is still > in the > > loop. This keeps 2 connections for event socket open and blocks > > resources (I am working with Ruby and eventmachine, there is a > limit of > > 20 simultaneous connections). > > > > > > Best regards > > Peter > > > > > > Anthony Minessale schrieb: > > > >> how are you exactly doing this? > >> with event socket? > >> > >> can you give a more detailed account of exactly what commands > you are > >> sending extensions used etc. > >> > >> > >> On Wed, Jan 7, 2009 at 5:00 AM, Peter P GMX > > >> >> > wrote: > >> > >> Hello Anthony, > >> > >> I updated to SNV=11084 and still have the problem. The > behaviour is > >> slightly changed now. > >> Step 4+5 (a s below in my mail) > >> 4) When I bridge A and B, A and B canNOT hear each other > (e.g. for 1/2 > >> sec). A continues to hear its messages, B does not hear > anything > >> 5) When I hangup B then A is still active and does not > recognize > >> hangup > >> of B. > >> > >> At former times, when the call was bridged, I had an > "unbind" for each > >> call on the event_socket interface, so event_socket was out > of the > >> loop > >> after uuid_bridge. Now I have an "unbind" for each party > only at the > >> time when the party hangs up. > >> > >> Best regards > >> Peter > >> > >> > >> Anthony Minessale schrieb: > >> > update one more time and see how that is > >> > > >> > > >> > On Tue, Jan 6, 2009 at 12:25 PM, Peter P GMX > >> > > > >> > >>> > >> wrote: > >> > > >> > One more info: > >> > I have updated to the newest SVN version of FS. > >> > A and B can actually hear each other (just a bit, some > >> scratching) > >> > while > >> > the announcement to A is very slow (~50% speed) and > very choppy. > >> > > >> > Best regards > >> > peter > >> > > >> > Peter P GMX schrieb: > >> > > I have setup a test machine and a production > machine. Since > >> > recently the > >> > > production machine behaves differently in terms of uuid > >> bridge. > >> > > > >> > > How it should work (and how it worked before) > >> > > 1) call A comes in > >> > > 2) I play some messages to A > >> > > 3) In the meantime I originate a call to B and > transfer > >> to an > >> > > extension, where also some messages are played > >> > > 4) Then I bridge A and B, so they are dropped off the > >> current > >> > > announcements an speak to each other > >> > > 5) when either A or B hangs up, both legs are > terminated > >> > > > >> > > New behaviour > >> > > 1) call A comes in > >> > > 2) I play some messages to A > >> > > 3) In the meantime I originate a call to B and > transfer > >> to an > >> > > extension, where also some messages are played > >> > > 4) Then I bridge A and B, A and B can hear each other > >> for 1/2 sec, > >> > > then A constinues to hear its messages, B does not hear > >> anything > >> > > 5) when either A or B hangs up, both legs are > terminated > >> > > > >> > > 4) is different now! > >> > > > >> > > The FS console show some messages about unbridge > >> > > (SWITCH_MESSAGE_INDICATE_UNBRIDGE), but somehow the > calls > >> are still > >> > > connected until A or B hangs up. > >> > > > >> > > Anybody has a clue? > >> > > Best regards > >> > > Peter > >> > > > >> > > 3) is finished, 4) starts > >> > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > Terminator) > >> > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:35:31 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:35:34 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1094 > >> > > switch_ivr_uuid_bridge() (OpenZAP/2:3/49171xxxxxxx) > State > >> Change > >> > > CS_EXECUTE -> CS_RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1095 > >> > > switch_ivr_uuid_bridge() (OpenZAP/2:1/216xxxxx) > State Change > >> > CS_EXECUTE > >> > > -> CS_RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:433 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > State EXECUTE > >> > going to > >> > > sleep > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:375 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > Running State > >> > Change > >> > > CS_RESET > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:429 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > State RESET > >> > > 4) Start > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > >> > > uuid_bridge_on_reset() OpenZAP/2:1/216xxxxx CUSTOM > RESET > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:53 > >> > > switch_core_standard_on_reset() OpenZAP/2:1/216xxxxx > >> Standard RESET > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:429 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > State RESET > >> > going to sleep > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:433 > >> > > switch_core_session_run() > (OpenZAP/2:3/49171xxxxxxx) State > >> > EXECUTE going > >> > > to sleep > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:375 > >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) > >> Running State > >> > > Change CS_RESET > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:429 > >> > > switch_core_session_run() > (OpenZAP/2:3/49171xxxxxxx) State > >> RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > >> > > uuid_bridge_on_reset() OpenZAP/2:3/49171xxxxxxx > CUSTOM RESET > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:533 > >> > > uuid_bridge_on_reset() (OpenZAP/2:3/49171xxxxxxx) > State Change > >> > CS_RESET > >> > > -> CS_SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:429 > >> > > switch_core_session_run() > (OpenZAP/2:3/49171xxxxxxx) State > >> RESET > >> > going > >> > > to sleep > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:375 > >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) > >> Running State > >> > > Change CS_SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:439 > >> > > switch_core_session_run() > (OpenZAP/2:3/49171xxxxxxx) State > >> > SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > >> > channel_on_soft_execute() > >> > > CHANNEL SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > >> > > uuid_bridge_on_soft_execute() > OpenZAP/2:3/49171xxxxxxx CUSTOM > >> > TRANSMIT > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:571 > >> > > uuid_bridge_on_soft_execute() > (OpenZAP/2:1/216xxxxx) State > >> Change > >> > > CS_RESET -> CS_SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:375 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > Running State > >> > Change > >> > > CS_SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:439 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > >> SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > >> > channel_on_soft_execute() > >> > > CHANNEL SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > >> > > uuid_bridge_on_soft_execute() OpenZAP/2:1/216xxxxx > CUSTOM > >> TRANSMIT > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:181 > >> > > switch_core_standard_on_soft_execute() > >> OpenZAP/2:1/216xxxxx Standard > >> > > SOFT_EXECUTE > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:439 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > >> SOFT_EXECUTE > >> > > going to sleep > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:814 > >> > > switch_ivr_multi_threaded_bridge() > (OpenZAP/2:1/216xxxxx) > >> State > >> > Change > >> > > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:856 > >> > > switch_ivr_multi_threaded_bridge() OpenZAP/2:1/216xxxxx > >> receive > >> > message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:863 > >> > > switch_ivr_multi_threaded_bridge() > >> OpenZAP/2:3/49171xxxxxxx receive > >> > > message [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:907 > >> > > switch_ivr_multi_threaded_bridge() > (OpenZAP/2:1/216xxxxx) > >> State > >> > Change > >> > > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:375 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > Running State > >> > Change > >> > > CS_EXCHANGE_MEDIA > >> > > 2009-01-06 17:35:35 [DEBUG] > switch_core_state_machine.c:436 > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > >> > EXCHANGE_MEDIA > >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:511 > >> > > channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > >> > switch_ivr_parse_event() > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > >> > > custom/warteschleife_30.wav interrupt_digit 0 ) > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_play_say.c:932 > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > >> channels 20ms > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_io.c:610 > >> > > switch_core_session_write_frame() > OpenZAP/2:1/216xxxxx receive > >> > message > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx > receive message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > >> > switch_ivr_parse_event() > >> > > OpenZAP/2:3/49171xxxxxxx Command Execute > stop_dtmf(true) > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:229 > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx > receive message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:231 > >> > > audio_bridge_thread() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx > receive message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:229 > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx > receive message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:231 > >> > > audio_bridge_thread() Send signal > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 31 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7d 02 01 22 30 > 0d a1 > >> 05 30 > >> > 03 02 01 > >> > > 08 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > Terminator) > >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:35:46 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 31 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7e 02 01 22 30 > 0d a1 > >> 05 30 > >> > 03 02 01 > >> > > 09 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > Terminator) > >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:36:02 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:36:06 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:229 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:36:07 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:231 > >> > > audio_bridge_thread() Send signal > OpenZAP/2:3/49171xxxxxxx > >> [BREAK] > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr.c:391 > >> > switch_ivr_parse_event() > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > >> > > custom/warteschleife_30.wav interrupt_digit 0 ) > >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_play_say.c:932 > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > >> channels 20ms > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_io.c:610 > >> > > switch_core_session_write_frame() > OpenZAP/2:1/216xxxxx receive > >> > message > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 31 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7f 02 01 22 30 > 0d a1 > >> 05 30 > >> > 03 02 01 > >> > > 0a 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > Terminator) > >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:36:17 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 32 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 80 02 01 22 > 30 0d > >> a1 05 > >> > 30 03 02 > >> > > 01 0b 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > Terminator) > >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:36:33 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:36:38 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:229 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:36:39 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:231 > >> > > audio_bridge_thread() Send signal > OpenZAP/2:3/49171xxxxxxx > >> [BREAK] > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr.c:391 > >> > switch_ivr_parse_event() > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > >> > > > >> > > >> > custom/dieser_berater_ist_leider_zur_zeit_nicht_erreichbar_bitte_waehlen_sie_einen_ande.wav > >> > > interrupt_digit 0 ) > >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_play_say.c:932 > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > >> channels 20ms > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_io.c:610 > >> > > switch_core_session_write_frame() > OpenZAP/2:1/216xxxxx receive > >> > message > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:45 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:229 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > message > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > >> > > 2009-01-06 17:36:46 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:231 > >> > > audio_bridge_thread() Send signal > OpenZAP/2:3/49171xxxxxxx > >> [BREAK] > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:667 > >> > > switch_core_session_queue_private_event() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_bridge.c:226 > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > message > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr.c:391 > >> > switch_ivr_parse_event() > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 4 > >> > > > >> > > >> > custom/geben_sie_bitte_nach_dem_signalton_die_kennung_des_gewuenschten_beraters_ein_sch2.wav > >> > > dtmfdtmf 10000 #,*) > >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_play_say.c:932 > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > >> channels 20ms > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_io.c:610 > >> > > switch_core_session_write_frame() > OpenZAP/2:1/216xxxxx receive > >> > message > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > >> > > switch_core_session_perform_receive_message() Send > signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 32 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 81 02 01 22 > 30 0d > >> a1 05 > >> > 30 03 02 > >> > > 01 0c 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > Terminator) > >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:36:48 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:37:03 [DEBUG] switch_ivr_play_say.c:1222 > >> > > switch_ivr_play_file() done playing file > >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 32 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 82 02 01 22 > 30 0d > >> a1 05 > >> > 30 03 02 > >> > > 01 0d 82 01 00 83 01 00] > >> > > > >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > Terminator) > >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > >> > > 2009-01-06 17:37:04 [CRIT] ozmod_isdn.c:760 > zap_isdn_931_34() > >> > Received > >> > > unhandled message 98 (0x62) > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:777 > zap_isdn_921_23() > >> > READ 13 > >> > > > >> > > >> > -------------------------------------------------------------------------------- > >> > > [08 02 00 35 45 08 02 80 90 1e 02 82 88] > >> > > > >> > > 5) Hangup > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:320 > zap_isdn_931_34() > >> > Yay I got > >> > > an event! Type:[45] Size:[115] CRV: 53 (0x35, CTX: > Originator) > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:352 > >> zap_isdn_931_34() zchan > >> > > 77df70 (2:1) source > isdn_data->channels_remote_crv[0x35] > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:455 > zap_isdn_931_34() > >> > Changing > >> > > state on 2:1 from UP to TERMINATING > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:813 > >> state_advance() 2:1 > >> > STATE > >> > > [TERMINATING] > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1121 > state_advance() > >> > > Terminating: Direction Inbound > >> > > 2009-01-06 17:37:12 [DEBUG] mod_openzap.c:1418 > >> > on_clear_channel_signal() > >> > > got clear channel sig [STOP] > >> > > 2009-01-06 17:37:12 [NOTICE] mod_openzap.c:1437 > >> > > on_clear_channel_signal() Hangup OpenZAP/2:1/216xxxxx > >> > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > >> > > 2009-01-06 17:37:12 [DEBUG] switch_channel.c:1441 > >> > > switch_channel_perform_hangup() Send signal > >> OpenZAP/2:1/216xxxxx > >> > [KILL] > >> > > 2009-01-06 17:37:12 [DEBUG] switch_core_session.c:779 > >> > > switch_core_session_signal_state_change() Send signal > >> > > OpenZAP/2:1/216xxxxx [BREAK] > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1529 > q931_rx_32() > >> WRITE 5 > >> > > > >> > > > >> > > > >> > > > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at 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 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 > > >> > > >> > > >> >> > >> > iax:guest at conference.freeswitch.org/888 > > >> > >> > > >> > googletalk:conf+888 at conference.freeswitch.org > > >> > > >> > > >> > >> > >> > pstn:213-799-1400 > >> > > >> > ------------------------------------------------------------------------ > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at 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 at 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 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 > > >> > > >> iax:guest at conference.freeswitch.org/888 > > >> > >> googletalk:conf+888 at conference.freeswitch.org > > >> > > >> pstn:213-799-1400 > >> > ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at 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 at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mszlazak at aol.com Sun Jan 11 14:21:59 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sun, 11 Jan 2009 17:21:59 -0500 Subject: [Freeswitch-users] Can I transfer information from local application to FS without the need of another server? Message-ID: <8CB425227745844-16E4-AB8@mblk-d46.sysops.aol.com> I would like to transfer information from an ASR application to FS (both on the same box) without setting up another server (e.g. PHP+Apache). I see that FS can do outgoing HTTP request but does it have something for handling inbound requests? I thought of transferring from the other app a to FS extension, have FS do some Javascript, then transfer back to the app but no information is passed except that it maybe implicit in the extension that is chosen. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090111/cfb48fc1/attachment.html From anthony.minessale at gmail.com Sun Jan 11 16:42:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 11 Jan 2009 18:42:22 -0600 Subject: [Freeswitch-users] different behaviour on uuid_bridge, doesn't really bridge In-Reply-To: <496A46EE.9070303@gmx.net> References: <49638DC1.2000307@gmx.net> <4963A228.3000103@gmx.net> <191c3a030901061609m19d91c15o8fad4b54efba017b@mail.gmail.com> <49648B4C.8050505@gmx.net> <191c3a030901071026r3ec5866cy5b1cf2938573e0a0@mail.gmail.com> <4965D95A.10905@gmx.net> <4967ADA9.3020003@gmx.net> <191c3a030901091244q726db0e9tab313822c5bcd7be@mail.gmail.com> <496A46EE.9070303@gmx.net> Message-ID: <191c3a030901111642j122ffd27n6d33dcc18cb8c75d@mail.gmail.com> yes where did the 2 legs come from to begin with? I want every single piece of info including every microsecond of the whole scenario starting with no calls and ending when the problem happens with a full description of what happens and what state the channels are in once they are done. Also I already mentioned it once but unless you are available to interact with us in real time i do not have time to send and email each day asking for more info. If you would just contact us in US/Central time 9am/6pm on irc it would make it much easier to make sure we get all the info we need. On Sun, Jan 11, 2009 at 1:22 PM, Peter P GMX wrote: > Hello Anthony, > > let me describe what I do: > On leg A I play a sound via mod_eventsocket thruogh the read application > to make this sound interruptable > call-command: execute > execute-app-name: read_int > execute-app-arg: 0 1 wava.wav interrupt_digit 0 > event-lock:true > > On leg B I play a sound via mod_eventsocket thruogh the read application > to make this sound interruptable > call-command: execute > execute-app-name: read_int > execute-app-arg: 0 1 wavb.wav interrupt_digit 0 > event-lock:true > > When this sound is finished, I do a uuid bridge via xmlrpc > > ?>freeswitch.apibgapiuuid_bridge > 425e65b0-e011-11dd-9580-3fea958c6de3 > > 344b23aa-e011-11dd-9580-3fea958c6de3 > > Then I disable dtmf with dtmf_stop via xmlrpc > > I ngreped this part of the session and have put this + the freeswitch > debug output to the pastebin: > http://pastebin.freeswitch.org/6742 > > Do you need mor info? > > Best regards > Peter > > > Anthony Minessale schrieb: > > What I am asking you for is a a step by step exact commands you are > doing. > > > > Give me the verbatim description of what you do. > > what exact commands to you send to the socket and exact apps with args > > you execute in the dialplan. > > > > are you on irc, or im so i can ask you more questions live? > > > > > > > > On Fri, Jan 9, 2009 at 2:03 PM, Peter P GMX > > wrote: > > > > I investigated a bit more in this. > > > > If I use my event_socket app, it does bridge but event_socket stays > in > > the line. I tested the following scenarios: > > - bridge openzap with openzap > > - bridge openzap with sip > > - bridge sip with sip > > So it's not an openzap-Issue. > > > > Before I bridge, I transfer both legs of the call into park and then > > bridge via uuid_bridge. > > > > Still I have no idea why this is different from a former > > intstallation. > > > > Best regards Peter > > > > Peter P GMX schrieb: > > > Yes, I am doing it via event socket. > > > On a system with pure SIP it works, but on a system where on > > both logs > > > openzap is used, it doesn't work. So maybe it's an openzap problem? > > > I do the following via event socket: > > > Leg B is doing the bridge via "uuid_bridge > > > > B>" while leg A is playing a soundfile via uuid_playback > > > I can see on the FS side that FS tries to bridge and then then > > does an > > > unbridge: (see also at the end of this mail history) > > > > > >> > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive message > > >> > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > >> > > > > > > I recognized that my application detects the bridge approach and > > > initiates the playback of another sound file. This causes, that the > > > soundfile is played, and so the parties cannot hear each other. I > > > changed the behaviour of my application, so that it doesn't try > > to play > > > another sound file. Now both legs can hear each other. > > > So the call is bridged, although FS indicates a different message. > > > > > > The most important part for my application is though, that for me > FS > > > doesn't really complete the bridge so that event socket is still > > in the > > > loop. This keeps 2 connections for event socket open and blocks > > > resources (I am working with Ruby and eventmachine, there is a > > limit of > > > 20 simultaneous connections). > > > > > > > > > Best regards > > > Peter > > > > > > > > > Anthony Minessale schrieb: > > > > > >> how are you exactly doing this? > > >> with event socket? > > >> > > >> can you give a more detailed account of exactly what commands > > you are > > >> sending extensions used etc. > > >> > > >> > > >> On Wed, Jan 7, 2009 at 5:00 AM, Peter P GMX > > > > >> >> > > wrote: > > >> > > >> Hello Anthony, > > >> > > >> I updated to SNV=11084 and still have the problem. The > > behaviour is > > >> slightly changed now. > > >> Step 4+5 (a s below in my mail) > > >> 4) When I bridge A and B, A and B canNOT hear each other > > (e.g. for 1/2 > > >> sec). A continues to hear its messages, B does not hear > > anything > > >> 5) When I hangup B then A is still active and does not > > recognize > > >> hangup > > >> of B. > > >> > > >> At former times, when the call was bridged, I had an > > "unbind" for each > > >> call on the event_socket interface, so event_socket was out > > of the > > >> loop > > >> after uuid_bridge. Now I have an "unbind" for each party > > only at the > > >> time when the party hangs up. > > >> > > >> Best regards > > >> Peter > > >> > > >> > > >> Anthony Minessale schrieb: > > >> > update one more time and see how that is > > >> > > > >> > > > >> > On Tue, Jan 6, 2009 at 12:25 PM, Peter P GMX > > >> > > > > > >> > > > >>> > > >> wrote: > > >> > > > >> > One more info: > > >> > I have updated to the newest SVN version of FS. > > >> > A and B can actually hear each other (just a bit, some > > >> scratching) > > >> > while > > >> > the announcement to A is very slow (~50% speed) and > > very choppy. > > >> > > > >> > Best regards > > >> > peter > > >> > > > >> > Peter P GMX schrieb: > > >> > > I have setup a test machine and a production > > machine. Since > > >> > recently the > > >> > > production machine behaves differently in terms of > uuid > > >> bridge. > > >> > > > > >> > > How it should work (and how it worked before) > > >> > > 1) call A comes in > > >> > > 2) I play some messages to A > > >> > > 3) In the meantime I originate a call to B and > > transfer > > >> to an > > >> > > extension, where also some messages are played > > >> > > 4) Then I bridge A and B, so they are dropped off > the > > >> current > > >> > > announcements an speak to each other > > >> > > 5) when either A or B hangs up, both legs are > > terminated > > >> > > > > >> > > New behaviour > > >> > > 1) call A comes in > > >> > > 2) I play some messages to A > > >> > > 3) In the meantime I originate a call to B and > > transfer > > >> to an > > >> > > extension, where also some messages are played > > >> > > 4) Then I bridge A and B, A and B can hear each > other > > >> for 1/2 sec, > > >> > > then A constinues to hear its messages, B does not > hear > > >> anything > > >> > > 5) when either A or B hangs up, both legs are > > terminated > > >> > > > > >> > > 4) is different now! > > >> > > > > >> > > The FS console show some messages about unbridge > > >> > > (SWITCH_MESSAGE_INDICATE_UNBRIDGE), but somehow the > > calls > > >> are still > > >> > > connected until A or B hangs up. > > >> > > > > >> > > Anybody has a clue? > > >> > > Best regards > > >> > > Peter > > >> > > > > >> > > 3) is finished, 4) starts > > >> > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:320 > > zap_isdn_931_34() > > >> > Yay I got > > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > > Terminator) > > >> > > 2009-01-06 17:35:31 [DEBUG] ozmod_isdn.c:352 > > >> zap_isdn_931_34() zchan > > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > >> > > 2009-01-06 17:35:31 [CRIT] ozmod_isdn.c:760 > > zap_isdn_931_34() > > >> > Received > > >> > > unhandled message 98 (0x62) > > >> > > 2009-01-06 17:35:34 [DEBUG] switch_ivr_play_say.c:1222 > > >> > > switch_ivr_play_file() done playing file > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1094 > > >> > > switch_ivr_uuid_bridge() (OpenZAP/2:3/49171xxxxxxx) > > State > > >> Change > > >> > > CS_EXECUTE -> CS_RESET > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > >> > > switch_core_session_signal_state_change() Send signal > > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:1095 > > >> > > switch_ivr_uuid_bridge() (OpenZAP/2:1/216xxxxx) > > State Change > > >> > CS_EXECUTE > > >> > > -> CS_RESET > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > >> > > switch_core_session_signal_state_change() Send signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_play_say.c:1222 > > >> > > switch_ivr_play_file() done playing file > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:433 > > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > > State EXECUTE > > >> > going to > > >> > > sleep > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:375 > > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > > Running State > > >> > Change > > >> > > CS_RESET > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:429 > > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > > State RESET > > >> > > 4) Start > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > > >> > > uuid_bridge_on_reset() OpenZAP/2:1/216xxxxx CUSTOM > > RESET > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:53 > > >> > > switch_core_standard_on_reset() OpenZAP/2:1/216xxxxx > > >> Standard RESET > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:429 > > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > > State RESET > > >> > going to sleep > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:433 > > >> > > switch_core_session_run() > > (OpenZAP/2:3/49171xxxxxxx) State > > >> > EXECUTE going > > >> > > to sleep > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:375 > > >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) > > >> Running State > > >> > > Change CS_RESET > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:429 > > >> > > switch_core_session_run() > > (OpenZAP/2:3/49171xxxxxxx) State > > >> RESET > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:527 > > >> > > uuid_bridge_on_reset() OpenZAP/2:3/49171xxxxxxx > > CUSTOM RESET > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:533 > > >> > > uuid_bridge_on_reset() (OpenZAP/2:3/49171xxxxxxx) > > State Change > > >> > CS_RESET > > >> > > -> CS_SOFT_EXECUTE > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > >> > > switch_core_session_signal_state_change() Send signal > > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:429 > > >> > > switch_core_session_run() > > (OpenZAP/2:3/49171xxxxxxx) State > > >> RESET > > >> > going > > >> > > to sleep > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:375 > > >> > > switch_core_session_run() (OpenZAP/2:3/49171xxxxxxx) > > >> Running State > > >> > > Change CS_SOFT_EXECUTE > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:439 > > >> > > switch_core_session_run() > > (OpenZAP/2:3/49171xxxxxxx) State > > >> > SOFT_EXECUTE > > >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > > >> > channel_on_soft_execute() > > >> > > CHANNEL SOFT_EXECUTE > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > > >> > > uuid_bridge_on_soft_execute() > > OpenZAP/2:3/49171xxxxxxx CUSTOM > > >> > TRANSMIT > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:571 > > >> > > uuid_bridge_on_soft_execute() > > (OpenZAP/2:1/216xxxxx) State > > >> Change > > >> > > CS_RESET -> CS_SOFT_EXECUTE > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > >> > > switch_core_session_signal_state_change() Send signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:375 > > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > > Running State > > >> > Change > > >> > > CS_SOFT_EXECUTE > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:439 > > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > > >> SOFT_EXECUTE > > >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:517 > > >> > channel_on_soft_execute() > > >> > > CHANNEL SOFT_EXECUTE > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:545 > > >> > > uuid_bridge_on_soft_execute() OpenZAP/2:1/216xxxxx > > CUSTOM > > >> TRANSMIT > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:181 > > >> > > switch_core_standard_on_soft_execute() > > >> OpenZAP/2:1/216xxxxx Standard > > >> > > SOFT_EXECUTE > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:439 > > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > > >> SOFT_EXECUTE > > >> > > going to sleep > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:814 > > >> > > switch_ivr_multi_threaded_bridge() > > (OpenZAP/2:1/216xxxxx) > > >> State > > >> > Change > > >> > > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > >> > > switch_core_session_signal_state_change() Send signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:856 > > >> > > switch_ivr_multi_threaded_bridge() > OpenZAP/2:1/216xxxxx > > >> receive > > >> > message > > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:863 > > >> > > switch_ivr_multi_threaded_bridge() > > >> OpenZAP/2:3/49171xxxxxxx receive > > >> > > message [SWITCH_MESSAGE_INDICATE_BRIDGE] > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:667 > > >> > > switch_core_session_queue_private_event() Send signal > > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_ivr_bridge.c:907 > > >> > > switch_ivr_multi_threaded_bridge() > > (OpenZAP/2:1/216xxxxx) > > >> State > > >> > Change > > >> > > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > > >> > > 2009-01-06 17:35:35 [DEBUG] switch_core_session.c:779 > > >> > > switch_core_session_signal_state_change() Send signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:375 > > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) > > Running State > > >> > Change > > >> > > CS_EXCHANGE_MEDIA > > >> > > 2009-01-06 17:35:35 [DEBUG] > > switch_core_state_machine.c:436 > > >> > > switch_core_session_run() (OpenZAP/2:1/216xxxxx) State > > >> > EXCHANGE_MEDIA > > >> > > 2009-01-06 17:35:35 [DEBUG] mod_openzap.c:511 > > >> > > channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:667 > > >> > > switch_core_session_queue_private_event() Send signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > > message > > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > > >> > switch_ivr_parse_event() > > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > >> > > custom/warteschleife_30.wav interrupt_digit 0 ) > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_play_say.c:932 > > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > > >> channels 20ms > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_io.c:610 > > >> > > switch_core_session_write_frame() > > OpenZAP/2:1/216xxxxx receive > > >> > message > > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:226 > > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx > > receive message > > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr.c:391 > > >> > switch_ivr_parse_event() > > >> > > OpenZAP/2:3/49171xxxxxxx Command Execute > > stop_dtmf(true) > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:229 > > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx > > receive message > > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > >> > > 2009-01-06 17:35:36 [DEBUG] switch_ivr_bridge.c:231 > > >> > > audio_bridge_thread() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:667 > > >> > > switch_core_session_queue_private_event() Send signal > > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:226 > > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx > > receive message > > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:229 > > >> > > audio_bridge_thread() OpenZAP/2:3/49171xxxxxxx > > receive message > > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > >> > > 2009-01-06 17:35:37 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:3/49171xxxxxxx [BREAK] > > >> > > 2009-01-06 17:35:37 [DEBUG] switch_ivr_bridge.c:231 > > >> > > audio_bridge_thread() Send signal > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:777 > > zap_isdn_921_23() > > >> > READ 31 > > >> > > > > >> > > > >> > > > -------------------------------------------------------------------------------- > > >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7d 02 01 22 30 > > 0d a1 > > >> 05 30 > > >> > 03 02 01 > > >> > > 08 82 01 00 83 01 00] > > >> > > > > >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:320 > > zap_isdn_931_34() > > >> > Yay I got > > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > > Terminator) > > >> > > 2009-01-06 17:35:46 [DEBUG] ozmod_isdn.c:352 > > >> zap_isdn_931_34() zchan > > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > >> > > 2009-01-06 17:35:46 [CRIT] ozmod_isdn.c:760 > > zap_isdn_931_34() > > >> > Received > > >> > > unhandled message 98 (0x62) > > >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:777 > > zap_isdn_921_23() > > >> > READ 31 > > >> > > > > >> > > > >> > > > -------------------------------------------------------------------------------- > > >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7e 02 01 22 30 > > 0d a1 > > >> 05 30 > > >> > 03 02 01 > > >> > > 09 82 01 00 83 01 00] > > >> > > > > >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:320 > > zap_isdn_931_34() > > >> > Yay I got > > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > > Terminator) > > >> > > 2009-01-06 17:36:02 [DEBUG] ozmod_isdn.c:352 > > >> zap_isdn_931_34() zchan > > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > >> > > 2009-01-06 17:36:02 [CRIT] ozmod_isdn.c:760 > > zap_isdn_931_34() > > >> > Received > > >> > > unhandled message 98 (0x62) > > >> > > 2009-01-06 17:36:06 [DEBUG] switch_ivr_play_say.c:1222 > > >> > > switch_ivr_play_file() done playing file > > >> > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:229 > > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > > message > > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > >> > > 2009-01-06 17:36:07 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:07 [DEBUG] switch_ivr_bridge.c:231 > > >> > > audio_bridge_thread() Send signal > > OpenZAP/2:3/49171xxxxxxx > > >> [BREAK] > > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:667 > > >> > > switch_core_session_queue_private_event() Send signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_bridge.c:226 > > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > > message > > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr.c:391 > > >> > switch_ivr_parse_event() > > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > >> > > custom/warteschleife_30.wav interrupt_digit 0 ) > > >> > > 2009-01-06 17:36:08 [DEBUG] switch_ivr_play_say.c:932 > > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > > >> channels 20ms > > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_io.c:610 > > >> > > switch_core_session_write_frame() > > OpenZAP/2:1/216xxxxx receive > > >> > message > > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > >> > > 2009-01-06 17:36:08 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:777 > > zap_isdn_921_23() > > >> > READ 31 > > >> > > > > >> > > > >> > > > -------------------------------------------------------------------------------- > > >> > > [08 02 80 04 62 1c 18 91 a1 15 02 01 7f 02 01 22 30 > > 0d a1 > > >> 05 30 > > >> > 03 02 01 > > >> > > 0a 82 01 00 83 01 00] > > >> > > > > >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:320 > > zap_isdn_931_34() > > >> > Yay I got > > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > > Terminator) > > >> > > 2009-01-06 17:36:17 [DEBUG] ozmod_isdn.c:352 > > >> zap_isdn_931_34() zchan > > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > >> > > 2009-01-06 17:36:17 [CRIT] ozmod_isdn.c:760 > > zap_isdn_931_34() > > >> > Received > > >> > > unhandled message 98 (0x62) > > >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:777 > > zap_isdn_921_23() > > >> > READ 32 > > >> > > > > >> > > > >> > > > -------------------------------------------------------------------------------- > > >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 80 02 01 22 > > 30 0d > > >> a1 05 > > >> > 30 03 02 > > >> > > 01 0b 82 01 00 83 01 00] > > >> > > > > >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:320 > > zap_isdn_931_34() > > >> > Yay I got > > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > > Terminator) > > >> > > 2009-01-06 17:36:33 [DEBUG] ozmod_isdn.c:352 > > >> zap_isdn_931_34() zchan > > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > >> > > 2009-01-06 17:36:33 [CRIT] ozmod_isdn.c:760 > > zap_isdn_931_34() > > >> > Received > > >> > > unhandled message 98 (0x62) > > >> > > 2009-01-06 17:36:38 [DEBUG] switch_ivr_play_say.c:1222 > > >> > > switch_ivr_play_file() done playing file > > >> > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:229 > > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > > message > > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > >> > > 2009-01-06 17:36:39 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:39 [DEBUG] switch_ivr_bridge.c:231 > > >> > > audio_bridge_thread() Send signal > > OpenZAP/2:3/49171xxxxxxx > > >> [BREAK] > > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:667 > > >> > > switch_core_session_queue_private_event() Send signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_bridge.c:226 > > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > > message > > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr.c:391 > > >> > switch_ivr_parse_event() > > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 1 > > >> > > > > >> > > > >> > > > custom/dieser_berater_ist_leider_zur_zeit_nicht_erreichbar_bitte_waehlen_sie_einen_ande.wav > > >> > > interrupt_digit 0 ) > > >> > > 2009-01-06 17:36:40 [DEBUG] switch_ivr_play_say.c:932 > > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > > >> channels 20ms > > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_io.c:610 > > >> > > switch_core_session_write_frame() > > OpenZAP/2:1/216xxxxx receive > > >> > message > > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > >> > > 2009-01-06 17:36:40 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:45 [DEBUG] switch_ivr_play_say.c:1222 > > >> > > switch_ivr_play_file() done playing file > > >> > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:229 > > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > > message > > >> > > [SWITCH_MESSAGE_INDICATE_BRIDGE] > > >> > > 2009-01-06 17:36:46 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:46 [DEBUG] switch_ivr_bridge.c:231 > > >> > > audio_bridge_thread() Send signal > > OpenZAP/2:3/49171xxxxxxx > > >> [BREAK] > > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:667 > > >> > > switch_core_session_queue_private_event() Send signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_bridge.c:226 > > >> > > audio_bridge_thread() OpenZAP/2:1/216xxxxx receive > > message > > >> > > [SWITCH_MESSAGE_INDICATE_UNBRIDGE] > > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr.c:391 > > >> > switch_ivr_parse_event() > > >> > > OpenZAP/2:1/216xxxxx Command Execute read(0 4 > > >> > > > > >> > > > >> > > > custom/geben_sie_bitte_nach_dem_signalton_die_kennung_des_gewuenschten_beraters_ein_sch2.wav > > >> > > dtmfdtmf 10000 #,*) > > >> > > 2009-01-06 17:36:47 [DEBUG] switch_ivr_play_say.c:932 > > >> > > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 > > >> channels 20ms > > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_io.c:610 > > >> > > switch_core_session_write_frame() > > OpenZAP/2:1/216xxxxx receive > > >> > message > > >> > > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > > >> > > 2009-01-06 17:36:47 [DEBUG] switch_core_session.c:489 > > >> > > switch_core_session_perform_receive_message() Send > > signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:777 > > zap_isdn_921_23() > > >> > READ 32 > > >> > > > > >> > > > >> > > > -------------------------------------------------------------------------------- > > >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 81 02 01 22 > > 30 0d > > >> a1 05 > > >> > 30 03 02 > > >> > > 01 0c 82 01 00 83 01 00] > > >> > > > > >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:320 > > zap_isdn_931_34() > > >> > Yay I got > > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > > Terminator) > > >> > > 2009-01-06 17:36:48 [DEBUG] ozmod_isdn.c:352 > > >> zap_isdn_931_34() zchan > > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > >> > > 2009-01-06 17:36:48 [CRIT] ozmod_isdn.c:760 > > zap_isdn_931_34() > > >> > Received > > >> > > unhandled message 98 (0x62) > > >> > > 2009-01-06 17:37:03 [DEBUG] switch_ivr_play_say.c:1222 > > >> > > switch_ivr_play_file() done playing file > > >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:777 > > zap_isdn_921_23() > > >> > READ 32 > > >> > > > > >> > > > >> > > > -------------------------------------------------------------------------------- > > >> > > [08 02 80 04 62 1c 19 91 a1 16 02 02 00 82 02 01 22 > > 30 0d > > >> a1 05 > > >> > 30 03 02 > > >> > > 01 0d 82 01 00 83 01 00] > > >> > > > > >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:320 > > zap_isdn_931_34() > > >> > Yay I got > > >> > > an event! Type:[62] Size:[103] CRV: 4 (0x4, CTX: > > Terminator) > > >> > > 2009-01-06 17:37:04 [DEBUG] ozmod_isdn.c:352 > > >> zap_isdn_931_34() zchan > > >> > > 783bc0 (2:3) source isdn_data->channels_local_crv[0x4] > > >> > > 2009-01-06 17:37:04 [CRIT] ozmod_isdn.c:760 > > zap_isdn_931_34() > > >> > Received > > >> > > unhandled message 98 (0x62) > > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:777 > > zap_isdn_921_23() > > >> > READ 13 > > >> > > > > >> > > > >> > > > -------------------------------------------------------------------------------- > > >> > > [08 02 00 35 45 08 02 80 90 1e 02 82 88] > > >> > > > > >> > > 5) Hangup > > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:320 > > zap_isdn_931_34() > > >> > Yay I got > > >> > > an event! Type:[45] Size:[115] CRV: 53 (0x35, CTX: > > Originator) > > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:352 > > >> zap_isdn_931_34() zchan > > >> > > 77df70 (2:1) source > > isdn_data->channels_remote_crv[0x35] > > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:455 > > zap_isdn_931_34() > > >> > Changing > > >> > > state on 2:1 from UP to TERMINATING > > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:813 > > >> state_advance() 2:1 > > >> > STATE > > >> > > [TERMINATING] > > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1121 > > state_advance() > > >> > > Terminating: Direction Inbound > > >> > > 2009-01-06 17:37:12 [DEBUG] mod_openzap.c:1418 > > >> > on_clear_channel_signal() > > >> > > got clear channel sig [STOP] > > >> > > 2009-01-06 17:37:12 [NOTICE] mod_openzap.c:1437 > > >> > > on_clear_channel_signal() Hangup OpenZAP/2:1/216xxxxx > > >> > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > >> > > 2009-01-06 17:37:12 [DEBUG] switch_channel.c:1441 > > >> > > switch_channel_perform_hangup() Send signal > > >> OpenZAP/2:1/216xxxxx > > >> > [KILL] > > >> > > 2009-01-06 17:37:12 [DEBUG] switch_core_session.c:779 > > >> > > switch_core_session_signal_state_change() Send signal > > >> > > OpenZAP/2:1/216xxxxx [BREAK] > > >> > > 2009-01-06 17:37:12 [DEBUG] ozmod_isdn.c:1529 > > q931_rx_32() > > >> WRITE 5 > > >> > > > > >> > > > > >> > > > > >> > > > > >> > > > >> > _______________________________________________ > > >> > Freeswitch-users mailing list > > >> > Freeswitch-users at 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 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 > > > > > > >> > > > >> > > >> > > > > > > > >> > > > >>> > > >> > iax:guest at conference.freeswitch.org/888 > > > > >> > > >> > > > >> > googletalk:conf+888 at conference.freeswitch.org > > > > > > >> > > > >> > > >> > > > > > > > >> > > > > > >>> > > >> > pstn:213-799-1400 > > >> > > > >> > > > ------------------------------------------------------------------------ > > >> > > > >> > _______________________________________________ > > >> > Freeswitch-users mailing list > > >> > Freeswitch-users at 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 at 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 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 > > > > > > >> > > > >> > > >> iax:guest at conference.freeswitch.org/888 > > > > >> > > >> googletalk:conf+888 at conference.freeswitch.org > > > > > > >> > > > >> > > >> pstn:213-799-1400 > > >> > > > ------------------------------------------------------------------------ > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users at 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 at 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 at 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 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 > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090111/f78b9edc/attachment-0001.html From allenrlevin at yahoo.com Sun Jan 11 14:10:42 2009 From: allenrlevin at yahoo.com (Allen Levin) Date: Sun, 11 Jan 2009 14:10:42 -0800 (PST) Subject: [Freeswitch-users] x100p freeswitch Message-ID: <586214.95908.qm@web81505.mail.mud.yahoo.com> Has anyone setup an X100p card to work with freeswitch? Thanks, Allen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090111/fec14a69/attachment.html From brian at freeswitch.org Sun Jan 11 21:40:00 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 11 Jan 2009 23:40:00 -0600 Subject: [Freeswitch-users] x100p freeswitch In-Reply-To: <586214.95908.qm@web81505.mail.mud.yahoo.com> References: <586214.95908.qm@web81505.mail.mud.yahoo.com> Message-ID: <38D75861-E672-4EA2-A920-81FA06E588D7@freeswitch.org> Yes it works. Read the OpenZAP page. http://wiki.freeswitch.org/wiki/OpenZAP /b On Jan 11, 2009, at 4:10 PM, Allen Levin wrote: > > Has anyone setup an X100p card to work with freeswitch? > > Thanks, > Allen > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090111/2007b01d/attachment.html From timb0311 at hotmail.com Sun Jan 11 23:23:48 2009 From: timb0311 at hotmail.com (Tim B) Date: Mon, 12 Jan 2009 02:23:48 -0500 Subject: [Freeswitch-users] Compile Mod_Managed on Centos 5.2 (Michael Jerris) In-Reply-To: References: Message-ID: Mike, thanks for the update. Could you email me directly and let me know when you have a chance to fix this? I am trying to setup a test server and hope to get it up and running soon. Thanks. Tim > ------------------------------> > Message: 6> Date: Sat, 10 Jan 2009 00:10:15 -0500> From: "Michael Jerris" > Subject: Re: [Freeswitch-users] Compile Mod_Managed on Centos 5.2> To: freeswitch-users at lists.freeswitch.org> Message-ID:> <93769c20901092110j5a7b3078i5eaf7b6f93b12ce7 at mail.gmail.com>> Content-Type: text/plain; charset="windows-1252"> > This is broken from a change that just went in this afternoon.. I will fix> it shortly.> > Mike> _________________________________________________________________ Windows Live? Hotmail?: Chat. Store. Share. Do more with mail. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_hm_justgotbetter_explore_012009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/a815bd91/attachment.html From ahgindia308 at gmail.com Sun Jan 11 23:29:24 2009 From: ahgindia308 at gmail.com (ahgindia) Date: Sun, 11 Jan 2009 23:29:24 -0800 (PST) Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <191c3a030901100801vaf6680bicd380cf9e8610e73@mail.gmail.com> References: <21386948.post@talk.nabble.com> <191c3a030901100801vaf6680bicd380cf9e8610e73@mail.gmail.com> Message-ID: <21409714.post@talk.nabble.com> Hello, I have reported a bug at jira. http://jira.freeswitch.org/browse/FSCORE-267 Let me know the issue on this if you find any. Anthony Minessale-2 wrote: > > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Loading_FreeSWITCH_in_GDB > > please get a bt and file a jira at http://jira.freeswitch.org > > > > > On Sat, Jan 10, 2009 at 5:01 AM, ahgindia wrote: > >> >> Hi All, >> Recently I was testing the new freeswitch release 1.0.2 >> The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo CPU >> E7200 >> @ 2.53GHz processor. >> But it crashed, when there were 96 active calls in it (as can be seen >> from >> "show calls" on freeswitch cli) >> There is a dump file for it, in the folder from where i started the >> freeswitch. >> Let me know how can we know the cause of the crash. >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21386948.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21409714.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ahgindia308 at gmail.com Sun Jan 11 23:33:22 2009 From: ahgindia308 at gmail.com (ahgindia) Date: Sun, 11 Jan 2009 23:33:22 -0800 (PST) Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> References: <21386948.post@talk.nabble.com> <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> Message-ID: <21409759.post@talk.nabble.com> Hello, Here is the output of backtrace : http://pastebin.freeswitch.org/6745 Let me know if find any reason for crash in this trace. Brian West-3 wrote: > > What is the output of the backtrace? Can you include that in your > email? > > /b > > On Jan 10, 2009, at 5:01 AM, ahgindia wrote: > >> >> Hi All, >> Recently I was testing the new freeswitch release 1.0.2 >> The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo >> CPU E7200 >> @ 2.53GHz processor. >> But it crashed, when there were 96 active calls in it (as can be >> seen from >> "show calls" on freeswitch cli) >> There is a dump file for it, in the folder from where i started the >> freeswitch. >> Let me know how can we know the cause of the crash. >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21386948.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21409759.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From can_man at gmx.de Mon Jan 12 01:45:01 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Mon, 12 Jan 2009 10:45:01 +0100 Subject: [Freeswitch-users] XML lib curl - what is the best practice for directory binding? In-Reply-To: References: <20090105151604.264500@gmx.net> Message-ID: <20090112094501.225370@gmx.net> > Take a look at the wiki for this module. I have been updating it > constantly and there are a lot of new information there. > > http://wiki.freeswitch.org/wiki/Mod_xml_curl > Thank you for your answer. I got all my info from the wiki page and partly from this list, and I think the wiki page is lacking some info which would make it much easier for a beginner to understand. I am happy to add info, but currently I don't know what's the best way of doing things, so I don't see myself in a position to change the page considerably. Maybe us two could meet on IRC at one point in order to make some changes together. Cheers, Phil > > On Jan 5, 2009, at 1:16 PM, can_man at gmx.de wrote: > > > > > Hello, > > > > I have been looking into the xml curl directory binding and I would > > like to update the wiki with the accepted best practice. I have > > listed the HTTP POST request I am getting and how I respond. If > > there is a better way please let me know and I will update the wiki > > accordingly. Btw, what I have done works - so no bug hunting this > > time ;-) > > I will make a pylons webserver available in the next few days, > > starting with dialplan and directory support. > > > > Thank you, > > Phil > > > > > > At boot: > > HTTP POST request 1 > > > > [('hostname', u'voip'), ('section', u'directory'), ('tag_name', > > u''), ('key_name', u''), ('key_value', u'')] > > > > my response: > > > > > >
> >
> > > > > > I have left the response empty as I want to provide the users at > > runtime. > > > > ----------------------------------------------------------------------- > > At boot: > > HTTP POST request 2 > > > > [('hostname', u'voip'), ('section', u'directory'), ('tag_name', > > u''), ('key_name', u''), ('key_value', u'')] > > > > my response: > > > > > >
> >
> > > > > > ----------------------------------------------------------------------- > > At boot: > > HTTP POST request 3 > > > > [('hostname', u'voip'), ('section', u'directory'), ('tag_name', > > u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'), > > ('domain', u'192.168.178.22'), ('purpose', u'network-list')] > > > > my response: > > > > > >
> >
> > > > > > > > What is meant by network list here? If all the users should be > > loaded at boot time, is this the request which should get a response > > with the complete list? > > > > ---------------------------------------------------------------------- > > > > During runtime following this action: > > > > > > > >
> > > > > > > > > > > > > > > >
> > > > > > > > HTTP POST request: > > ('hostname', u'voip'), ('section', u'directory'), ('tag_name', > > u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'), > > ('mailbox', u'315'), ('key', u'id'), ('user', u'315'), ('domain', > > u'192.168.178.22'), ('ip', u'217.10.79.9') > > > > my response: > > > > > >
> > //change to your domain > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >
> > > > -- > > Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit > > allen: http://www.gmx.net/de/go/multimessenger > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a From dave at 3c.co.uk Mon Jan 12 03:49:31 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 12 Jan 2009 11:49:31 +0000 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl Message-ID: <496B2E4B.7070202@3c.co.uk> Hi all - In case anyone's interested, I've documented how we interfaced FS with Lumenvox via MRCP using FS' event socket and unicast interfaces and a bit of Perl here: http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl Three surprises: that it worked at all, that it works quite well and that it was really quite easy to do. One thing I'm looking for: has anyone written a module which attaches a bug to an audio stream and forwards the audio as RTP to a specified IP/port to just allow audio to be tapped off a call and sent somewhere else to be listened to? Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk From wasim at convergence.pk Mon Jan 12 04:05:05 2009 From: wasim at convergence.pk (Wasim Baig) Date: Mon, 12 Jan 2009 17:05:05 +0500 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <496B2E4B.7070202@3c.co.uk> References: <496B2E4B.7070202@3c.co.uk> Message-ID: On Mon, Jan 12, 2009 at 4:49 PM, David Knell wrote: In case anyone's interested, I've documented how we interfaced FS with > Lumenvox via MRCP using FS' event socket and unicast interfaces and a > bit of Perl here: > http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl > > Three surprises: that it worked at all, that it works quite well and > that it was really quite easy to do. aka the FS motto ... "it works, its works quite well, its really quite easy ... " -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/6bcb60ac/attachment.html From gmaruzz at celliax.org Mon Jan 12 04:34:19 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 12 Jan 2009 13:34:19 +0100 Subject: [Freeswitch-users] Skypiax, Skype compatible endpoint Message-ID: <7b197bef0901120434k16f5d910j9d11012ee243e674@mail.gmail.com> Ciao FreeSWITCHers, mod_skypiax is now usable, for Skype calls and finding bugs :-). Its wiki page (http://wiki.freeswitch.org/wiki/Skypiax, at the moment it's just the README files concatenated) begins like that: WHAT IS SKYPIAX This software (Skypiax) uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype. Skypiax is an endpoint (channel driver) that use the Skype client as an interface to the Skype network, and allows incoming and outgoing Skype calls from/to FreeSWITCH (that can be bridged, originated, answered, etc. as in all other endpoints, eg sofia/SIP). Think at Skypiax as similar to OpenZAP for analog lines: for each channel you need an interface (a Skype client). So, for eg, for two concurrent calls, you will need two channels, two Skype clients running on server. If your server's Skype client(s) has got the Skype credits, Skypiax works for SkypeOut calls too. You can use it from the dialplan, eg with the provided modified "default.xml" dialplan, you can call "sip:skype/remote_skypename__OR__skypeout_phonenumber" for calling via the Skype network from a SIP softphone to remote_skypename or to a phone number via SkypeOut, or you can call the "2908" extension from any phone to be bridged to the Skype Test Call). With the provided skypiax.conf.xml all incoming Skype calls will be routed to the "5000" extension, the IVR in default FreeSWITCH installation. On Linux the Skype client uses a lot of CPU. To lower its CPU consumption, you can use the Xvfb "fake" X server and (more important) the snd-dummy ALSA "fake" sound driver. Scripts are provided for this. But for a low number of channels it would works with regular X servers and ALSA drivers. On a Linux machine with 3GB ram and a quad core intel6600, we got no problem with 20 concurrent calls, and plenty of room for adding more Skypiax channels (100? not tested). On Windows, no need to do anything special, the Skype client is lighter on CPU. Skypiax is now pre-beta, but usable for testing and finding bugs :-). You can download Skypiax source code with subversion with the command: svn co http://svn.freeswitch.org/svn/freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax mod_skypiax then, follow the README file in the mod_skypiax directory. More info on skypiax: http://wiki.freeswitch.org/wiki/Skypiax http://www.celliax.org Sincerely, Giovanni Maruzzelli ========================================= Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039 From rehan at supertec.com Mon Jan 12 16:49:26 2009 From: rehan at supertec.com (Rehan Allah Wala) Date: Mon, 12 Jan 2009 17:49:26 -0700 Subject: [Freeswitch-users] Skypiax, Skype compatible endpoint In-Reply-To: <7b197bef0901120434k16f5d910j9d11012ee243e674@mail.gmail.com> References: <7b197bef0901120434k16f5d910j9d11012ee243e674@mail.gmail.com> Message-ID: <496B82A6.3574.4468654E@rehan.supertec.com> i am looking for a consulant to send me a quote to run this for me on amazon ec2 Rehan > Ciao FreeSWITCHers, > > mod_skypiax is now usable, for Skype calls and finding bugs :-). > > Its wiki page (http://wiki.freeswitch.org/wiki/Skypiax, at the moment > it's just the README files concatenated) begins like that: > > WHAT IS SKYPIAX > > This software (Skypiax) uses the Skype API but is not endorsed, > certified or otherwise approved in any way by Skype. > > Skypiax is an endpoint (channel driver) that use the Skype client as > an interface to the Skype network, and allows incoming and outgoing > Skype calls from/to FreeSWITCH (that can be bridged, originated, > answered, etc. as in all other endpoints, eg sofia/SIP). > > Think at Skypiax as similar to OpenZAP for analog lines: for each > channel you need an interface (a Skype client). So, for eg, for two > concurrent calls, you will need two channels, two Skype clients > running on server. > > If your server's Skype client(s) has got the Skype credits, Skypiax > works for SkypeOut calls too. > > You can use it from the dialplan, eg with the provided modified > "default.xml" dialplan, you can call > "sip:skype/remote_skypename__OR__skypeout_phonenumber" for calling via > the Skype network from a SIP softphone to remote_skypename or to a > phone number via SkypeOut, or you can call the "2908" extension from > any phone to be bridged to the Skype Test Call). > > With the provided skypiax.conf.xml all incoming Skype calls will be > routed to the "5000" extension, the IVR in default FreeSWITCH > installation. > > On Linux the Skype client uses a lot of CPU. To lower its CPU > consumption, you can use the Xvfb "fake" X server and (more important) > the snd-dummy ALSA "fake" sound driver. Scripts are provided for this. > But for a low number of channels it would works with regular X servers > and ALSA drivers. > > On a Linux machine with 3GB ram and a quad core intel6600, we got no > problem with 20 concurrent calls, and plenty of room for adding more > Skypiax channels (100? not tested). > > On Windows, no need to do anything special, the Skype client is lighter on CPU. > > > Skypiax is now pre-beta, but usable for testing and finding bugs :-). > > > You can download Skypiax source code with subversion with the command: > > svn co http://svn.freeswitch.org/svn/freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax > mod_skypiax > > then, follow the README file in the mod_skypiax directory. > > > More info on skypiax: > > http://wiki.freeswitch.org/wiki/Skypiax > > http://www.celliax.org > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > Company : Celliax > Website: www.celliax.org > Address : via Pierlombardo 9, 20135 Milano > Country/Territory : Italy > Business Email: gmaruzz at celliax dot org > Cell : 39-347-2665618 > Fax : 39-02-87390039 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Rehan Ahmed AllahWala Msn/Yahoo/GoogleTalk/Email: Rehan at Rehan.com http://www.supertec.com/ - Internet Telephony Solutions Http://www.DIDX.net - DID Number Market Place. Don't Remember Me ? Visit http://www.Rehan.com ~~~~~~~~~~~~~~~~~~~ "First they ignore you, then they laugh at you, then they fight you, then you win." By Gandhi. "Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi From ashley.ohq at gmail.com Mon Jan 12 00:02:22 2009 From: ashley.ohq at gmail.com (Ashley van Gerven) Date: Mon, 12 Jan 2009 19:02:22 +1100 Subject: [Freeswitch-users] SBC configuration Message-ID: <150a3aa50901120002x69967134v51b3ba0f13fac6b3@mail.gmail.com> Hi, I am interested in testing Freeswitch acting as an SBC. Is it simply a matter of configuring the dialplan correctly, using RE's so that inbound calls are just forwarded to our internal PBX and outbound calls from the PBX are forwarded to the VOIP provider? Or do I need to create an application that specifcally creates a new call and then joins the inbound and outbound calls? I haven't been able to find info on the wiki or google re. SBC setup. thanks Ash -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/1b803735/attachment.html From curly2009 at gmx.de Mon Jan 12 03:14:14 2009 From: curly2009 at gmx.de (curly2009 at gmx.de) Date: Mon, 12 Jan 2009 12:14:14 +0100 Subject: [Freeswitch-users] Openzap doesn't work Message-ID: <20090112111414.279860@gmx.net> Hello, I'm a newbie in FS. I installed and configure Openzap and Zaptel. I have a PRI Digium Wildcard TE110P. When I call a number then I get an Error! 2009-01-12 10:20:01 [CRIT] zap_io.c:804 zap_channel_open_any() All circuits are busy. 2009-01-12 10:20:01 [ERR] mod_openzap.c:910 channel_outgoing_channel() No channels available 2009-01-12 10:20:01 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [openzap] cause: [DESTINATION_OUT_OF_ORDER] What's wrong? Do you need any configurations? Thanks! Best Regards, Curly -- Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a From gservat at gmail.com Mon Jan 12 05:07:55 2009 From: gservat at gmail.com (Gonzalo Servat) Date: Mon, 12 Jan 2009 11:07:55 -0200 Subject: [Freeswitch-users] Skypiax, Skype compatible endpoint In-Reply-To: <7b197bef0901120434k16f5d910j9d11012ee243e674@mail.gmail.com> References: <7b197bef0901120434k16f5d910j9d11012ee243e674@mail.gmail.com> Message-ID: On Mon, Jan 12, 2009 at 10:34 AM, Giovanni Maruzzelli wrote: > Ciao FreeSWITCHers, > > mod_skypiax is now usable, for Skype calls and finding bugs :-). > [..snip..] Nice work Giovanni! I'll be trying it out and I'll let you know how I go. Thank you! Gonzalo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/fbb4e5e8/attachment.html From ahgindia308 at gmail.com Mon Jan 12 05:08:09 2009 From: ahgindia308 at gmail.com (ahgindia) Date: Mon, 12 Jan 2009 05:08:09 -0800 (PST) Subject: [Freeswitch-users] Freeswitch not passing more than two calls In-Reply-To: <5BA25000-CF3F-4704-B3E0-2336581E1B77@freeswitch.org> References: <21355617.post@talk.nabble.com> <87f2f3b90901081558u13d08328q97d4c998c45f81ad@mail.gmail.com> <21371461.post@talk.nabble.com> <4DD06273-247F-4494-9BB9-CCDB82963B67@freeswitch.org> <21374865.post@talk.nabble.com> <5BA25000-CF3F-4704-B3E0-2336581E1B77@freeswitch.org> Message-ID: <21414081.post@talk.nabble.com> Hello, I have tried the new freeswitch release 1.0.2. But while starting phase only, I encountered freeswitch crashes frequently. Here it is : http://www.nabble.com/Freeswitch-crashed-!!!-tt21386948.html What do you suggest now? Brian West-3 wrote: > > Are you not on svn trunk or 1.0.2? If not I would highly recommend > you update to that before we move forward. > > /b > > On Jan 9, 2009, at 9:51 AM, ahgindia wrote: > >> >> Here is the log I collected from freeswitch log messages : >> http://pastebin.freeswitch.org/6714 >> Please check this and let me know if you find anything. >> >> Is it reliable safe to use new release of freeswitch 1.0.2 on the >> production >> server? Will there be this issue in that release. Please advice me >> for the >> issue. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-not-passing-more-than-two-calls-tp21355617p21414081.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From sicfslist at gmail.com Mon Jan 12 05:14:11 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Mon, 12 Jan 2009 07:14:11 -0600 Subject: [Freeswitch-users] SBC configuration In-Reply-To: <150a3aa50901120002x69967134v51b3ba0f13fac6b3@mail.gmail.com> References: <150a3aa50901120002x69967134v51b3ba0f13fac6b3@mail.gmail.com> Message-ID: <35b355e90901120514m2cab0684t77e924c158ef9839@mail.gmail.com> Hey Ashley, When you say "SBC" what are you trying to accomplish specifically? FS should be able to accomplish what you want ... but can you provide some more details as far as what you want to accomplish and I'm sure someone here will be glad to help. SR On Mon, Jan 12, 2009 at 2:02 AM, Ashley van Gerven wrote: > Hi, > > I am interested in testing Freeswitch acting as an SBC. Is it simply a > matter of configuring the dialplan correctly, using RE's so that inbound > calls are just forwarded to our internal PBX and outbound calls from the PBX > are forwarded to the VOIP provider? > > Or do I need to create an application that specifcally creates a new call > and then joins the inbound and outbound calls? > > I haven't been able to find info on the wiki or google re. SBC setup. > > > thanks > Ash > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/bd650a32/attachment-0001.html From dave at 3c.co.uk Mon Jan 12 05:16:19 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 12 Jan 2009 13:16:19 +0000 Subject: [Freeswitch-users] SBC configuration In-Reply-To: <150a3aa50901120002x69967134v51b3ba0f13fac6b3@mail.gmail.com> References: <150a3aa50901120002x69967134v51b3ba0f13fac6b3@mail.gmail.com> Message-ID: Hi Ash, It's the former. Here's a snippet from a dialplan of ours - this takes calls with a specific prefix from a specific IP address and forwards them to a particular carrier: - you'll need something similar for each direction. Cheers -- Dave > Hi, > > I am interested in testing Freeswitch acting as an SBC. Is it simply > a matter of configuring the dialplan correctly, using RE's so that > inbound calls are just forwarded to our internal PBX and outbound > calls from the PBX are forwarded to the VOIP provider? > > Or do I need to create an application that specifcally creates a new > call and then joins the inbound and outbound calls? > > I haven't been able to find info on the wiki or google re. SBC setup. > > > thanks > Ash > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pmhshz at gmail.com Mon Jan 12 05:23:14 2009 From: pmhshz at gmail.com (shehzad p) Date: Mon, 12 Jan 2009 05:23:14 -0800 (PST) Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <21386948.post@talk.nabble.com> References: <21386948.post@talk.nabble.com> Message-ID: <21414332.post@talk.nabble.com> Hi all, I am also testing FS release 1.0.2, but I faced strange problem. When I stop freeswitch (from CLI using ... or shutdown), Freeswitch ends with showing "Segmentation fault": Below is the last 15 lines when fault occures. Sometimes this does not happen and FS shut down normally. ===================================================================================== 2009-01-12 16:52:56 [CONSOLE] switch_loadable_module.c:1244 do_shutdown() mod_esf unloaded. 2009-01-12 16:52:56 [CONSOLE] switch_core.c:1462 switch_core_destroy() Closing Event Engine. 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 switch_event_shutdown() Stopping event queue 0 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 switch_event_shutdown() Stopping event queue 1 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 switch_event_thread() Event Thread 0 Ended. 2009-01-12 16:52:56 [CONSOLE] switch_event.c:445 switch_event_shutdown() Stopping dispatch queue 0 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 switch_event_thread() Event Thread 1 Ended. 2009-01-12 16:52:56 [CONSOLE] switch_event.c:247 switch_event_dispatch_thread() Dispatch Thread 0 Ended. 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 switch_event_thread() Event Thread 2 Ended. 2009-01-12 16:52:56 [CONSOLE] switch_event.c:414 switch_core_memory_reclaim_events() Returning 23 recycled event(s) 1012 bytes 2009-01-12 16:52:56 [CONSOLE] switch_event.c:416 switch_core_memory_reclaim_events() Returning 331 recycled event header(s) 5296 bytes 2009-01-12 16:52:56 [CONSOLE] switch_core_sqldb.c:539 switch_core_sqldb_stop() Waiting for unfinished SQL transactions 2009-01-12 16:52:56 [NOTICE] switch_core_sqldb.c:199 switch_core_sql_thread() SQL thread ending 2009-01-12 16:52:56 [CONSOLE] switch_scheduler.c:303 switch_scheduler_task_thread_stop() Stopping Task Thread Segmentation fault (core dumped) ===================================================================================== What should be the cause of such crash. ahgindia wrote: > > Hi All, > Recently I was testing the new freeswitch release 1.0.2 > The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo CPU > E7200 @ 2.53GHz processor. > But it crashed, when there were 96 active calls in it (as can be seen from > "show calls" on freeswitch cli) > There is a dump file for it, in the folder from where i started the > freeswitch. > Let me know how can we know the cause of the crash. > -- View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21414332.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Prometheus001 at gmx.net Mon Jan 12 06:01:11 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 12 Jan 2009 15:01:11 +0100 Subject: [Freeswitch-users] Openzap doesn't work In-Reply-To: <20090112111414.279860@gmx.net> References: <20090112111414.279860@gmx.net> Message-ID: <496B4D27.4020605@gmx.net> I had the same problem a while ago with German PRI lines (S2M). Do you have in your openzap.conf.xml? In older Versions of openzap I used euro, in newer releases this didn't work any more, so I used q931. If this is not successfull, can you post your openzap.conf.xml and openzap.xml and the output of cat /proc/zaptel/* Best regards Peter curly2009 at gmx.de schrieb: > Hello, > > I'm a newbie in FS. > > I installed and configure Openzap and Zaptel. I have a PRI Digium Wildcard TE110P. > > When I call a number then I get an Error! > > 2009-01-12 10:20:01 [CRIT] zap_io.c:804 zap_channel_open_any() All circuits are busy. > 2009-01-12 10:20:01 [ERR] mod_openzap.c:910 channel_outgoing_channel() No channels available > 2009-01-12 10:20:01 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [openzap] cause: [DESTINATION_OUT_OF_ORDER] > > What's wrong? Do you need any configurations? > > Thanks! > Best Regards, Curly > > > From curly2009 at gmx.de Mon Jan 12 06:36:45 2009 From: curly2009 at gmx.de (=?iso-8859-1?Q?=22Franziska_R=F6hler=22?=) Date: Mon, 12 Jan 2009 15:36:45 +0100 Subject: [Freeswitch-users] Openzap doesn't work In-Reply-To: <496B4D27.4020605@gmx.net> References: <20090112111414.279860@gmx.net> <496B4D27.4020605@gmx.net> Message-ID: <20090112143645.85460@gmx.net> I use q931 too. openzap.conf.xml openzap.conf [span zt] name => OpenZap number => 1 trunk_type => e1 b-channel => 1-15, 17-31 d-channel => 16 cat /proc/zaptel/* Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) HDB3/CCS/CRC4 RECOVERING 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 HDLCFCS (In use) RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 Clear (In use) RED 25 WCT1/0/25 Clear (In use) RED 26 WCT1/0/26 Clear (In use) RED 27 WCT1/0/27 Clear (In use) RD 28 WCT1/0/28 Clear (In use) RED 29 WCT1/0/29 Clear (In use) RED 30 WCT1/0/30 Clear (In use) RED 31 WCT1/0/31 Clear (In use) RED Why is it RED? zttool shows no alarms (ok) and ztfcg -vv shows that 31 Channels to configure. > I had the same problem a while ago with German PRI lines (S2M). > > Do you have > > in your > openzap.conf.xml? > > In older Versions of openzap I used euro, in newer releases this didn't > work any more, so I used q931. > > If this is not successfull, > > can you post your openzap.conf.xml and openzap.xml > and the output of cat /proc/zaptel/* > > Best regards > Peter > > curly2009 at gmx.de schrieb: > > Hello, > > > > I'm a newbie in FS. > > > > I installed and configure Openzap and Zaptel. I have a PRI Digium > Wildcard TE110P. > > > > When I call a number then I get an Error! > > > > 2009-01-12 10:20:01 [CRIT] zap_io.c:804 zap_channel_open_any() All > circuits are busy. > > 2009-01-12 10:20:01 [ERR] mod_openzap.c:910 channel_outgoing_channel() > No channels available > > 2009-01-12 10:20:01 [ERR] switch_ivr_originate.c:926 > switch_ivr_originate() Cannot create outgoing channel of type [openzap] cause: > [DESTINATION_OUT_OF_ORDER] > > > > What's wrong? Do you need any configurations? > > > > Thanks! > > Best Regards, Curly > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Mon Jan 12 06:41:15 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 12 Jan 2009 09:41:15 -0500 Subject: [Freeswitch-users] Freeswitch not passing more than two calls In-Reply-To: <21414081.post@talk.nabble.com> References: <21355617.post@talk.nabble.com> <87f2f3b90901081558u13d08328q97d4c998c45f81ad@mail.gmail.com> <21371461.post@talk.nabble.com> <4DD06273-247F-4494-9BB9-CCDB82963B67@freeswitch.org> <21374865.post@talk.nabble.com> <5BA25000-CF3F-4704-B3E0-2336581E1B77@freeswitch.org> <21414081.post@talk.nabble.com> Message-ID: <22321C02-010A-4D49-A745-3D3FD211E6B8@jerris.com> Please follow up on this issue on jira. On Jan 12, 2009, at 8:08 AM, ahgindia wrote: > > Hello, > > I have tried the new freeswitch release 1.0.2. > But while starting phase only, I encountered freeswitch crashes > frequently. > Here it is : http://www.nabble.com/Freeswitch-crashed-!!!-tt21386948.html > > What do you suggest now? > > > Brian West-3 wrote: >> >> Are you not on svn trunk or 1.0.2? If not I would highly recommend >> you update to that before we move forward. >> >> /b >> >> On Jan 9, 2009, at 9:51 AM, ahgindia wrote: >> >>> >>> Here is the log I collected from freeswitch log messages : >>> http://pastebin.freeswitch.org/6714 >>> Please check this and let me know if you find anything. >>> >>> Is it reliable safe to use new release of freeswitch 1.0.2 on the >>> production >>> server? Will there be this issue in that release. Please advice me >>> for the >>> issue. >> >> From jmesquita at gmail.com Mon Jan 12 06:55:20 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 12 Jan 2009 12:55:20 -0200 Subject: [Freeswitch-users] XML lib curl - what is the best practice for directory binding? In-Reply-To: <20090112094501.225370@gmx.net> References: <20090105151604.264500@gmx.net> <20090112094501.225370@gmx.net> Message-ID: <5a8712120901120655o6321bae5x87ee62c8bbee5756@mail.gmail.com> Phil, I am always on IRC under jmesquita or jmesquita_ (maybe even both). I was out on vacation, thats why it took me a while longer to reply, sorry about that. Mesquita On Mon, Jan 12, 2009 at 7:45 AM, wrote: > > > > Take a look at the wiki for this module. I have been updating it > > constantly and there are a lot of new information there. > > > > http://wiki.freeswitch.org/wiki/Mod_xml_curl > > > > Thank you for your answer. I got all my info from the wiki page and partly > from this list, and I think > the wiki page is lacking some info which would make it much easier for a > beginner to understand. > I am happy to add info, but currently I don't know what's the best way of > doing things, so I don't see > myself in a position to change the page considerably. Maybe us two could > meet on IRC at one point in order > to make some changes together. > > Cheers, > Phil > > > > > > On Jan 5, 2009, at 1:16 PM, can_man at gmx.de wrote: > > > > > > > > Hello, > > > > > > I have been looking into the xml curl directory binding and I would > > > like to update the wiki with the accepted best practice. I have > > > listed the HTTP POST request I am getting and how I respond. If > > > there is a better way please let me know and I will update the wiki > > > accordingly. Btw, what I have done works - so no bug hunting this > > > time ;-) > > > I will make a pylons webserver available in the next few days, > > > starting with dialplan and directory support. > > > > > > Thank you, > > > Phil > > > > > > > > > At boot: > > > HTTP POST request 1 > > > > > > [('hostname', u'voip'), ('section', u'directory'), ('tag_name', > > > u''), ('key_name', u''), ('key_value', u'')] > > > > > > my response: > > > > > > > > >
> > >
> > > > > > > > > I have left the response empty as I want to provide the users at > > > runtime. > > > > > > ----------------------------------------------------------------------- > > > At boot: > > > HTTP POST request 2 > > > > > > [('hostname', u'voip'), ('section', u'directory'), ('tag_name', > > > u''), ('key_name', u''), ('key_value', u'')] > > > > > > my response: > > > > > > > > >
> > >
> > > > > > > > > ----------------------------------------------------------------------- > > > At boot: > > > HTTP POST request 3 > > > > > > [('hostname', u'voip'), ('section', u'directory'), ('tag_name', > > > u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'), > > > ('domain', u'192.168.178.22'), ('purpose', u'network-list')] > > > > > > my response: > > > > > > > > >
> > >
> > > > > > > > > > > > What is meant by network list here? If all the users should be > > > loaded at boot time, is this the request which should get a response > > > with the complete list? > > > > > > ---------------------------------------------------------------------- > > > > > > During runtime following this action: > > > > > > > > > > > >
> > > > > > > > > > > > > > > > > > > > > > > >
> > > > > > > > > > > > HTTP POST request: > > > ('hostname', u'voip'), ('section', u'directory'), ('tag_name', > > > u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'), > > > ('mailbox', u'315'), ('key', u'id'), ('user', u'315'), ('domain', > > > u'192.168.178.22'), ('ip', u'217.10.79.9') > > > > > > my response: > > > > > > > > >
> > > //change to your domain > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >
> > > > > > -- > > > Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit > > > allen: http://www.gmx.net/de/go/multimessenger > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL > f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/3aceed4e/attachment-0001.html From curly2009 at gmx.de Mon Jan 12 07:16:45 2009 From: curly2009 at gmx.de (curly2009 at gmx.de) Date: Mon, 12 Jan 2009 16:16:45 +0100 Subject: [Freeswitch-users] Openzap doesn't work In-Reply-To: <20090112143645.85460@gmx.net> References: <20090112111414.279860@gmx.net> <496B4D27.4020605@gmx.net> <20090112143645.85460@gmx.net> Message-ID: <20090112151645.30780@gmx.net> I use q931 too. openzap.conf.xml openzap.conf [span zt] name => OpenZap number => 1 trunk_type => e1 b-channel => 1-15, 17-31 d-channel => 16 cat /proc/zaptel/* Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) HDB3/CCS/CRC4 RECOVERING 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 HDLCFCS (In use) RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 Clear (In use) RED 25 WCT1/0/25 Clear (In use) RED 26 WCT1/0/26 Clear (In use) RED 27 WCT1/0/27 Clear (In use) RD 28 WCT1/0/28 Clear (In use) RED 29 WCT1/0/29 Clear (In use) RED 30 WCT1/0/30 Clear (In use) RED 31 WCT1/0/31 Clear (In use) RED Why is it RED? zttool shows no alarms (ok) and ztfcg -vv shows that 31 Channels to configure. > I had the same problem a while ago with German PRI lines (S2M). > > Do you have > > in your > openzap.conf.xml? > > In older Versions of openzap I used euro, in newer releases this didn't > work any more, so I used q931. > > If this is not successfull, > > can you post your openzap.conf.xml and openzap.xml > and the output of cat /proc/zaptel/* > > Best regards > Peter > > curly2009 at gmx.de schrieb: > > Hello, > > > > I'm a newbie in FS. > > > > I installed and configure Openzap and Zaptel. I have a PRI Digium > Wildcard TE110P. > > > > When I call a number then I get an Error! > > > > 2009-01-12 10:20:01 [CRIT] zap_io.c:804 zap_channel_open_any() All > circuits are busy. > > 2009-01-12 10:20:01 [ERR] mod_openzap.c:910 channel_outgoing_channel() > No channels available > > 2009-01-12 10:20:01 [ERR] switch_ivr_originate.c:926 > switch_ivr_originate() Cannot create outgoing channel of type [openzap] cause: > [DESTINATION_OUT_OF_ORDER] > > > > What's wrong? Do you need any configurations? > > > > Thanks! > > Best Regards, Curly > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger From regs at kinetix.gr Mon Jan 12 07:51:15 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Mon, 12 Jan 2009 17:51:15 +0200 Subject: [Freeswitch-users] mod_opal first unsuccessful test Message-ID: <496B66F3.9000902@kinetix.gr> Hi, I successfully compiled mod_opal using the latest svn for both opal and ptlib as Brian suggested. When I try to establish a call using h323 from my openphone client I get no audio although I can see RTP packets in both directions when I am doing a capture. Some details : I am using the 11094 revision of the FS trunk. I am using the PCMU codec. I tried dialing the default IVR (5000) and other testing extensions (freeswitch conference, echo test etc.) I tried using fast start on and off , h245 tunneling on and off, h245 in SETUP on and off. In my captures I have also noticed a strange behavior : FS sends to my client 2 "alerting" packets for no apparent reason. Could this be a cause of the problem? Had anyone any success with mod_opal lately? If yes, could you please reply quoting your config options (both on FS and on your client)? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From msc at freeswitch.org Mon Jan 12 07:50:52 2009 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 12 Jan 2009 07:50:52 -0800 Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <21414332.post@talk.nabble.com> References: <21386948.post@talk.nabble.com> <21414332.post@talk.nabble.com> Message-ID: <428196FD-5E22-4745-BE8D-5C4AFFC3B2CF@freeswitch.org> Could you please do a backtrace and post it to a pastebin? If in Linux do this: gdb /path/to/freeswitch /path/to/corefile -MC Sent from my iPhone On Jan 12, 2009, at 5:23 AM, shehzad p wrote: > > Hi all, > I am also testing FS release 1.0.2, but I faced strange problem. > When I stop freeswitch (from CLI using ... or shutdown), Freeswitch > ends > with showing "Segmentation fault": > Below is the last 15 lines when fault occures. Sometimes this does not > happen and FS shut down normally. > > === > === > === > === > === > ====================================================================== > 2009-01-12 16:52:56 [CONSOLE] switch_loadable_module.c:1244 > do_shutdown() > mod_esf unloaded. > 2009-01-12 16:52:56 [CONSOLE] switch_core.c:1462 switch_core_destroy() > Closing Event Engine. > 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 > switch_event_shutdown() > Stopping event queue 0 > 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 > switch_event_shutdown() > Stopping event queue 1 > 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 > switch_event_thread() Event > Thread 0 Ended. > 2009-01-12 16:52:56 [CONSOLE] switch_event.c:445 > switch_event_shutdown() > Stopping dispatch queue 0 > 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 > switch_event_thread() Event > Thread 1 Ended. > 2009-01-12 16:52:56 [CONSOLE] switch_event.c:247 > switch_event_dispatch_thread() Dispatch Thread 0 Ended. > 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 > switch_event_thread() Event > Thread 2 Ended. > 2009-01-12 16:52:56 [CONSOLE] switch_event.c:414 > switch_core_memory_reclaim_events() Returning 23 recycled event(s) > 1012 > bytes > 2009-01-12 16:52:56 [CONSOLE] switch_event.c:416 > switch_core_memory_reclaim_events() Returning 331 recycled event > header(s) > 5296 bytes > 2009-01-12 16:52:56 [CONSOLE] switch_core_sqldb.c:539 > switch_core_sqldb_stop() Waiting for unfinished SQL transactions > 2009-01-12 16:52:56 [NOTICE] switch_core_sqldb.c:199 > switch_core_sql_thread() SQL thread ending > 2009-01-12 16:52:56 [CONSOLE] switch_scheduler.c:303 > switch_scheduler_task_thread_stop() Stopping Task Thread > Segmentation fault (core dumped) > === > === > === > === > === > ====================================================================== > > What should be the cause of such crash. > > > ahgindia wrote: >> >> Hi All, >> Recently I was testing the new freeswitch release 1.0.2 >> The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo CPU >> E7200 @ 2.53GHz processor. >> But it crashed, when there were 96 active calls in it (as can be >> seen from >> "show calls" on freeswitch cli) >> There is a dump file for it, in the folder from where i started the >> freeswitch. >> Let me know how can we know the cause of the crash. >> > > > > -- > View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21414332.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Mon Jan 12 08:02:32 2009 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 12 Jan 2009 08:02:32 -0800 Subject: [Freeswitch-users] Openzap doesn't work In-Reply-To: <20090112143645.85460@gmx.net> References: <20090112111414.279860@gmx.net> <496B4D27.4020605@gmx.net> <20090112143645.85460@gmx.net> Message-ID: <7AEB3C13-131F-45D3-B2E1-28BB8F07F28D@freeswitch.org> This looks like a zaptel issue. Do we have any zaptel users familiar with this issue? If not you should probably check on the asterisk list for help. Once zaptel is working we can then see about getting openzap up. -MC Sent from my iPhone On Jan 12, 2009, at 6:36 AM, "Franziska R?hler" wrote: > I use q931 too. > > openzap.conf.xml > > > > > > > > > > > > > > > > > > > > openzap.conf > > [span zt] > name => OpenZap > number => 1 > trunk_type => e1 > b-channel => 1-15, 17-31 > d-channel => 16 > > cat /proc/zaptel/* > Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) HDB3/ > CCS/CRC4 RECOVERING > > 1 WCT1/0/1 Clear (In use) RED > 2 WCT1/0/2 Clear (In use) RED > 3 WCT1/0/3 Clear (In use) RED > 4 WCT1/0/4 Clear (In use) RED > 5 WCT1/0/5 Clear (In use) RED > 6 WCT1/0/6 Clear (In use) RED > 7 WCT1/0/7 Clear (In use) RED > 8 WCT1/0/8 Clear (In use) RED > 9 WCT1/0/9 Clear (In use) RED > 10 WCT1/0/10 Clear (In use) RED > 11 WCT1/0/11 Clear (In use) RED > 12 WCT1/0/12 Clear (In use) RED > 13 WCT1/0/13 Clear (In use) RED > 14 WCT1/0/14 Clear (In use) RED > 15 WCT1/0/15 Clear (In use) RED > 16 WCT1/0/16 HDLCFCS (In use) RED > 17 WCT1/0/17 Clear (In use) RED > 18 WCT1/0/18 Clear (In use) RED > 19 WCT1/0/19 Clear (In use) RED > 20 WCT1/0/20 Clear (In use) RED > 21 WCT1/0/21 Clear (In use) RED > 22 WCT1/0/22 Clear (In use) RED > 23 WCT1/0/23 Clear (In use) RED > 24 WCT1/0/24 Clear (In use) RED > 25 WCT1/0/25 Clear (In use) RED > 26 WCT1/0/26 Clear (In use) RED > 27 WCT1/0/27 Clear (In use) RD > 28 WCT1/0/28 Clear (In use) RED > 29 WCT1/0/29 Clear (In use) RED > 30 WCT1/0/30 Clear (In use) RED > 31 WCT1/0/31 Clear (In use) RED > > Why is it RED? zttool shows no alarms (ok) and ztfcg -vv shows that > 31 Channels to configure. > > > >> I had the same problem a while ago with German PRI lines (S2M). >> >> Do you have >> >> in your >> openzap.conf.xml? >> >> In older Versions of openzap I used euro, in newer releases this >> didn't >> work any more, so I used q931. >> >> If this is not successfull, >> >> can you post your openzap.conf.xml and openzap.xml >> and the output of cat /proc/zaptel/* >> >> Best regards >> Peter >> >> curly2009 at gmx.de schrieb: >>> Hello, >>> >>> I'm a newbie in FS. >>> >>> I installed and configure Openzap and Zaptel. I have a PRI Digium >> Wildcard TE110P. >>> >>> When I call a number then I get an Error! >>> >>> 2009-01-12 10:20:01 [CRIT] zap_io.c:804 zap_channel_open_any() All >> circuits are busy. >>> 2009-01-12 10:20:01 [ERR] mod_openzap.c:910 >>> channel_outgoing_channel() >> No channels available >>> 2009-01-12 10:20:01 [ERR] switch_ivr_originate.c:926 >> switch_ivr_originate() Cannot create outgoing channel of type >> [openzap] cause: >> [DESTINATION_OUT_OF_ORDER] >>> >>> What's wrong? Do you need any configurations? >>> >>> Thanks! >>> Best Regards, Curly >>> >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Jan 12 08:22:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 12 Jan 2009 10:22:11 -0600 Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <21409759.post@talk.nabble.com> References: <21386948.post@talk.nabble.com> <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> <21409759.post@talk.nabble.com> Message-ID: <191c3a030901120822i31e6d40dgdc88dae43d49b05b@mail.gmail.com> please supply the entire javascript code you were executing. On Mon, Jan 12, 2009 at 1:33 AM, ahgindia wrote: > > Hello, > > Here is the output of backtrace : > http://pastebin.freeswitch.org/6745 > > Let me know if find any reason for crash in this trace. > > > Brian West-3 wrote: > > > > What is the output of the backtrace? Can you include that in your > > email? > > > > /b > > > > On Jan 10, 2009, at 5:01 AM, ahgindia wrote: > > > >> > >> Hi All, > >> Recently I was testing the new freeswitch release 1.0.2 > >> The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo > >> CPU E7200 > >> @ 2.53GHz processor. > >> But it crashed, when there were 96 active calls in it (as can be > >> seen from > >> "show calls" on freeswitch cli) > >> There is a dump file for it, in the folder from where i started the > >> freeswitch. > >> Let me know how can we know the cause of the crash. > >> -- > >> View this message in context: > >> > http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21386948.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21409759.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/b4d4bb9a/attachment-0001.html From cstomi.levlist at gmail.com Mon Jan 12 08:24:43 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Mon, 12 Jan 2009 17:24:43 +0100 Subject: [Freeswitch-users] loopback+intercept+bind_meta_app Message-ID: <496B6ECB.4070809@gmail.com> Hello, could you please tell me if I can use bind_meta_app if 1, originate a call with loopback channel 2, push into fifo 3, originate another call to handle the previous call 4 and joining them with intercept. I have problems with it I quess the issue is that this isn't a simple bridge and there is not really a and b legs. Might not. Is it possible to use dtmf bindings with intercept? If it should work, then maybe it is an issue with loopback? Thanks, Tamas From Prometheus001 at gmx.net Mon Jan 12 08:26:46 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 12 Jan 2009 17:26:46 +0100 Subject: [Freeswitch-users] Openzap doesn't work In-Reply-To: <7AEB3C13-131F-45D3-B2E1-28BB8F07F28D@freeswitch.org> References: <20090112111414.279860@gmx.net> <496B4D27.4020605@gmx.net> <20090112143645.85460@gmx.net> <7AEB3C13-131F-45D3-B2E1-28BB8F07F28D@freeswitch.org> Message-ID: <496B6F46.3080105@gmx.net> >From my experience it shows red when: - the cables are not connected - the pri lines are not synchronized yet - sometimes the telco provider has to reset the line Best regards Peter Michael S Collins schrieb: > This looks like a zaptel issue. Do we have any zaptel users familiar > with this issue? If not you should probably check on the asterisk list > for help. Once zaptel is working we can then see about getting openzap > up. > > -MC > > Sent from my iPhone > > On Jan 12, 2009, at 6:36 AM, "Franziska R?hler" > wrote: > > >> I use q931 too. >> >> openzap.conf.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> openzap.conf >> >> [span zt] >> name => OpenZap >> number => 1 >> trunk_type => e1 >> b-channel => 1-15, 17-31 >> d-channel => 16 >> >> cat /proc/zaptel/* >> Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) HDB3/ >> CCS/CRC4 RECOVERING >> >> 1 WCT1/0/1 Clear (In use) RED >> 2 WCT1/0/2 Clear (In use) RED >> 3 WCT1/0/3 Clear (In use) RED >> 4 WCT1/0/4 Clear (In use) RED >> 5 WCT1/0/5 Clear (In use) RED >> 6 WCT1/0/6 Clear (In use) RED >> 7 WCT1/0/7 Clear (In use) RED >> 8 WCT1/0/8 Clear (In use) RED >> 9 WCT1/0/9 Clear (In use) RED >> 10 WCT1/0/10 Clear (In use) RED >> 11 WCT1/0/11 Clear (In use) RED >> 12 WCT1/0/12 Clear (In use) RED >> 13 WCT1/0/13 Clear (In use) RED >> 14 WCT1/0/14 Clear (In use) RED >> 15 WCT1/0/15 Clear (In use) RED >> 16 WCT1/0/16 HDLCFCS (In use) RED >> 17 WCT1/0/17 Clear (In use) RED >> 18 WCT1/0/18 Clear (In use) RED >> 19 WCT1/0/19 Clear (In use) RED >> 20 WCT1/0/20 Clear (In use) RED >> 21 WCT1/0/21 Clear (In use) RED >> 22 WCT1/0/22 Clear (In use) RED >> 23 WCT1/0/23 Clear (In use) RED >> 24 WCT1/0/24 Clear (In use) RED >> 25 WCT1/0/25 Clear (In use) RED >> 26 WCT1/0/26 Clear (In use) RED >> 27 WCT1/0/27 Clear (In use) RD >> 28 WCT1/0/28 Clear (In use) RED >> 29 WCT1/0/29 Clear (In use) RED >> 30 WCT1/0/30 Clear (In use) RED >> 31 WCT1/0/31 Clear (In use) RED >> >> Why is it RED? zttool shows no alarms (ok) and ztfcg -vv shows that >> 31 Channels to configure. >> >> >> >> >>> I had the same problem a while ago with German PRI lines (S2M). >>> >>> Do you have >>> >>> in your >>> openzap.conf.xml? >>> >>> In older Versions of openzap I used euro, in newer releases this >>> didn't >>> work any more, so I used q931. >>> >>> If this is not successfull, >>> >>> can you post your openzap.conf.xml and openzap.xml >>> and the output of cat /proc/zaptel/* >>> >>> Best regards >>> Peter >>> >>> curly2009 at gmx.de schrieb: >>> >>>> Hello, >>>> >>>> I'm a newbie in FS. >>>> >>>> I installed and configure Openzap and Zaptel. I have a PRI Digium >>>> >>> Wildcard TE110P. >>> >>>> When I call a number then I get an Error! >>>> >>>> 2009-01-12 10:20:01 [CRIT] zap_io.c:804 zap_channel_open_any() All >>>> >>> circuits are busy. >>> >>>> 2009-01-12 10:20:01 [ERR] mod_openzap.c:910 >>>> channel_outgoing_channel() >>>> >>> No channels available >>> >>>> 2009-01-12 10:20:01 [ERR] switch_ivr_originate.c:926 >>>> >>> switch_ivr_originate() Cannot create outgoing channel of type >>> [openzap] cause: >>> [DESTINATION_OUT_OF_ORDER] >>> >>>> What's wrong? Do you need any configurations? >>>> >>>> Thanks! >>>> Best Regards, Curly >>>> >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Mon Jan 12 08:38:24 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 12 Jan 2009 17:38:24 +0100 Subject: [Freeswitch-users] Skypiax, Skype compatible endpoint In-Reply-To: <496B82A6.3574.4468654E@rehan.supertec.com> References: <7b197bef0901120434k16f5d910j9d11012ee243e674@mail.gmail.com> <496B82A6.3574.4468654E@rehan.supertec.com> Message-ID: <496B7200.2040402@gmx.net> what kind of ec2 machine is it? Linux/Distribution? Windows? best regards Peter Rehan Allah Wala schrieb: > i am looking for a consulant to send me a quote to run this for me on amazon ec2 > > Rehan > > > > >> Ciao FreeSWITCHers, >> >> mod_skypiax is now usable, for Skype calls and finding bugs :-). >> >> Its wiki page (http://wiki.freeswitch.org/wiki/Skypiax, at the moment >> it's just the README files concatenated) begins like that: >> >> WHAT IS SKYPIAX >> >> This software (Skypiax) uses the Skype API but is not endorsed, >> certified or otherwise approved in any way by Skype. >> >> Skypiax is an endpoint (channel driver) that use the Skype client as >> an interface to the Skype network, and allows incoming and outgoing >> Skype calls from/to FreeSWITCH (that can be bridged, originated, >> answered, etc. as in all other endpoints, eg sofia/SIP). >> >> Think at Skypiax as similar to OpenZAP for analog lines: for each >> channel you need an interface (a Skype client). So, for eg, for two >> concurrent calls, you will need two channels, two Skype clients >> running on server. >> >> If your server's Skype client(s) has got the Skype credits, Skypiax >> works for SkypeOut calls too. >> >> You can use it from the dialplan, eg with the provided modified >> "default.xml" dialplan, you can call >> "sip:skype/remote_skypename__OR__skypeout_phonenumber" for calling via >> the Skype network from a SIP softphone to remote_skypename or to a >> phone number via SkypeOut, or you can call the "2908" extension from >> any phone to be bridged to the Skype Test Call). >> >> With the provided skypiax.conf.xml all incoming Skype calls will be >> routed to the "5000" extension, the IVR in default FreeSWITCH >> installation. >> >> On Linux the Skype client uses a lot of CPU. To lower its CPU >> consumption, you can use the Xvfb "fake" X server and (more important) >> the snd-dummy ALSA "fake" sound driver. Scripts are provided for this. >> But for a low number of channels it would works with regular X servers >> and ALSA drivers. >> >> On a Linux machine with 3GB ram and a quad core intel6600, we got no >> problem with 20 concurrent calls, and plenty of room for adding more >> Skypiax channels (100? not tested). >> >> On Windows, no need to do anything special, the Skype client is lighter on CPU. >> >> >> Skypiax is now pre-beta, but usable for testing and finding bugs :-). >> >> >> You can download Skypiax source code with subversion with the command: >> >> svn co http://svn.freeswitch.org/svn/freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax >> mod_skypiax >> >> then, follow the README file in the mod_skypiax directory. >> >> >> More info on skypiax: >> >> http://wiki.freeswitch.org/wiki/Skypiax >> >> http://www.celliax.org >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> Company : Celliax >> Website: www.celliax.org >> Address : via Pierlombardo 9, 20135 Milano >> Country/Territory : Italy >> Business Email: gmaruzz at celliax dot org >> Cell : 39-347-2665618 >> Fax : 39-02-87390039 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > Rehan Ahmed AllahWala > Msn/Yahoo/GoogleTalk/Email: Rehan at Rehan.com > http://www.supertec.com/ - Internet Telephony Solutions > Http://www.DIDX.net - DID Number Market Place. > Don't Remember Me ? Visit http://www.Rehan.com > > ~~~~~~~~~~~~~~~~~~~ > "First they ignore you, then they laugh at you, then they fight you, then you win." > By Gandhi. > > "Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch at davidnicol.otherinbox.com Mon Jan 12 09:23:17 2009 From: freeswitch at davidnicol.otherinbox.com (freeswitch at davidnicol.otherinbox.com) Date: Mon, 12 Jan 2009 12:23:17 -0500 Subject: [Freeswitch-users] how many telephones can I drive from a linksys SPA-2100 FXS? Message-ID: <200901121723.n0CHNGJn009796@box7.911domain.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/a173280b/attachment.html From mszlazak at aol.com Mon Jan 12 09:22:58 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 12 Jan 2009 12:22:58 -0500 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <496B2E4B.7070202@3c.co.uk> References: <496B2E4B.7070202@3c.co.uk> Message-ID: <8CB42F18C78E89E-820-404@webmail-me14.sysops.aol.com> Great! I hope you will try doing Voxeo's Prophecy next as well ;-) Thanks Dave. Mark. -----Original Message----- From: David Knell To: freeswitch-users at lists.freeswitch.org Sent: Mon, 12 Jan 2009 3:49 am Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl Hi all - In case anyone's interested, I've documented how we interfaced FS with Lumenvox via MRCP using FS' event socket and unicast interfaces and a bit of Perl here: http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl Three surprises: that it worked at all, that it works quite well and that it was really quite easy to do. One thing I'm looking for: has anyone written a module which attaches a bug to an audio stream and forwards the audio as RTP to a specified IP/port to just allow audio to be tapped off a call and sent somewhere else to be listened to? Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/f031869c/attachment.html From brian at freeswitch.org Mon Jan 12 09:29:44 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 12 Jan 2009 11:29:44 -0600 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <496B2E4B.7070202@3c.co.uk> References: <496B2E4B.7070202@3c.co.uk> Message-ID: <3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org> "With FreeSWITCH not having any supported ASR at the time of writing (with the exception of PocketSphinx), we needed something to allow us to connect it to an MRCP server to test SoftIVR's ASR functionality. After a few false starts, we implemented a simple MRCP connector using the outbound socket interface, unicast and a bit of Perl." Was mod_openmrcp not enough :) We really need someone to fund the writing of mod_unimrcp. /b On Jan 12, 2009, at 5:49 AM, David Knell wrote: > Hi all - > > In case anyone's interested, I've documented how we interfaced FS with > Lumenvox via MRCP using FS' event socket and unicast interfaces and a > bit of Perl here: > http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl > > Three surprises: that it worked at all, that it works quite well and > that it was really quite easy to do. > > One thing I'm looking for: has anyone written a module which > attaches a > bug to an audio stream and forwards the audio as RTP to a specified > IP/port to just allow audio to be tapped off a call and sent somewhere > else to be listened to? > > Cheers -- > > Dave > > -- > David Knell, Director, 3C Limited > T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 > http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/ce4ec945/attachment-0001.html From mszlazak at aol.com Mon Jan 12 09:46:00 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 12 Jan 2009 12:46:00 -0500 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org> References: <496B2E4B.7070202@3c.co.uk> <3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org> Message-ID: <8CB42F4C40F8A82-820-5D0@webmail-me14.sysops.aol.com> Yup, or just get pocketsphinx "tuned" up for telephony and then no one will have to bother with ASR vendors. I believe that some speech data from a good size sample for training is needed to make it more "speaker independent" and better suited for use with phone calls. I have a list of things from the Sphinx forums that would be good to have for a telephony ready PocketSphinx. There is a "wsj" database but I don't know if that's would help?? Best. Mark. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Mon, 12 Jan 2009 9:29 am Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl "With FreeSWITCH not having any supported ASR at the time of writing (with the exception of PocketSphinx), we needed something to allow us to connect it to an MRCP server to test SoftIVR's ASR functionality. After a few false starts, we implemented a simple MRCP connector using the outbound socket interface, unicast and a bit of Perl." Was mod_openmrcp not enough :) ?We really need someone to fund the writing of mod_unimrcp. /b On Jan 12, 2009, at 5:49 AM, David Knell wrote: Hi all - In case anyone's interested, I've documented how we interfaced FS with Lumenvox via MRCP using FS' event socket and unicast interfaces and a bit of Perl here: http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl Three surprises: that it worked at all, that it works quite well and that it was really quite easy to do. One thing I'm looking for: has anyone written a module which attaches a bug to an audio stream and forwards the audio as RTP to a specified IP/port to just allow audio to be tapped off a call and sent somewhere else to be listened to? Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 ?F: 020 3002 7257 ?M: 001 415 630 3031 http://www.3c.co.uk _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/ae0a8942/attachment.html From freeswitch at davidnicol.otherinbox.com Mon Jan 12 09:50:07 2009 From: freeswitch at davidnicol.otherinbox.com (freeswitch at davidnicol.otherinbox.com) Date: Mon, 12 Jan 2009 12:50:07 -0500 Subject: [Freeswitch-users] how many telephones can I drive from a linksys SPA-2100 FXS? Message-ID: <200901121750.n0CHo6uj028005@box7.911domain.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/82e7d194/attachment.html From freeswitch-users at lists.rupa.com Mon Jan 12 09:53:46 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Mon, 12 Jan 2009 11:53:46 -0600 Subject: [Freeswitch-users] how many telephones can I drive from a linksys SPA-2100 FXS? In-Reply-To: <200901121723.n0CHNGJn009796@box7.911domain.com> References: <200901121723.n0CHNGJn009796@box7.911domain.com> Message-ID: <496B83AA.9050904@lists.rupa.com> On 1/12/2009 11:23 AM, freeswitch at davidnicol.otherinbox.com wrote: > I have just purchased one of these devices on eBay. Will it be able to > ring all four of the dumb unpowered analog telephones in my house? > Most likely, no problem if they aren't mechanical ringers. Each phone should have a REN (ringer equivalent number) and the SPA should document how many RENs it can drive. But 4 should be no problem. From brian at freeswitch.org Mon Jan 12 09:55:17 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 12 Jan 2009 11:55:17 -0600 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <8CB42F4C40F8A82-820-5D0@webmail-me14.sysops.aol.com> References: <496B2E4B.7070202@3c.co.uk> <3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org> <8CB42F4C40F8A82-820-5D0@webmail-me14.sysops.aol.com> Message-ID: Pocketsphinx works great for telephony.. just don't load 10000 word dictionary or grammar :P the pizza demo uses it.. and it works great from every phone I have tested it with... Rome wasn't built in a day and we need more people that have the skills to really build a general purpose acoustical model that works in more situations. /b On Jan 12, 2009, at 11:46 AM, mszlazak at aol.com wrote: > Yup, or just get pocketsphinx "tuned" up for telephony and then no > one will have to bother with ASR vendors. > > I believe that some speech data from a good size sample for training > is needed to make it more "speaker independent" and better suited > for use with phone calls. I have a list of things from the Sphinx > forums that would be good to have for a telephony ready > PocketSphinx. There is a "wsj" database but I don't know if that's > would help?? > > Best. Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/4260fe4f/attachment.html From dave at 3c.co.uk Mon Jan 12 10:20:34 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 12 Jan 2009 18:20:34 +0000 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org> References: <496B2E4B.7070202@3c.co.uk> <3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org> Message-ID: <496B89F2.6030002@3c.co.uk> Hi Brian, > "With FreeSWITCH not having any supported ASR at the time of writing > (with the exception of PocketSphinx), we needed something to allow us > to connect it to an MRCP server to test SoftIVR's ASR functionality. > After a few false starts, we implemented a simple MRCP connector using > the outbound socket interface, unicast and a bit of Perl." > > Was mod_openmrcp not enough :) We really need someone to fund the > writing of mod_unimrcp. mod_openmrcp is (from our testing) badly broken: it segfaults on the second call; it is (in my opinion) unnecessarily baroque and it is no longer supported. Cheers -- Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/b75ec3aa/attachment.html From brian at freeswitch.org Mon Jan 12 10:28:57 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 12 Jan 2009 12:28:57 -0600 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <496B89F2.6030002@3c.co.uk> References: <496B2E4B.7070202@3c.co.uk> <3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org> <496B89F2.6030002@3c.co.uk> Message-ID: <74545DF4-BD5C-4D3D-B96A-9BA9A2F531A7@freeswitch.org> Putting some effort behind unimrcp would be the best case at this point right? /b On Jan 12, 2009, at 12:20 PM, David Knell wrote: > mod_openmrcp is (from our testing) badly broken: it segfaults on the > second call; it is (in my opinion) unnecessarily baroque and it is > no longer supported. > > Cheers -- > > Dave From anthony.minessale at gmail.com Mon Jan 12 11:19:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 12 Jan 2009 13:19:15 -0600 Subject: [Freeswitch-users] mod_opal first unsuccessful test In-Reply-To: <496B66F3.9000902@kinetix.gr> References: <496B66F3.9000902@kinetix.gr> Message-ID: <191c3a030901121119i69066dcepdaef6e00f3567dd5@mail.gmail.com> I have only tested it with x-meeting so far. On Mon, Jan 12, 2009 at 9:51 AM, Apostolos Pantsiopoulos wrote: > Hi, > > I successfully compiled mod_opal using the latest svn for both opal > and ptlib as Brian suggested. > > When I try to establish a call using h323 from my openphone client > I get no audio although I can see RTP packets in both directions when I am > doing a capture. Some details : > > I am using the 11094 revision of the FS trunk. > I am using the PCMU codec. > I tried dialing the default IVR (5000) and other testing extensions > (freeswitch conference, echo test etc.) > I tried using fast start on and off , h245 tunneling on and off, h245 in > SETUP on and off. > > In my captures I have also noticed a strange behavior : FS sends to > my client 2 "alerting" packets > for no apparent reason. Could this be a cause of the problem? > > Had anyone any success with mod_opal lately? If yes, could you > please reply quoting your config > options (both on FS and on your client)? > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/441145c4/attachment-0001.html From gilbertandrew at me.com Mon Jan 12 12:19:25 2009 From: gilbertandrew at me.com (Andrew Gilbert) Date: Mon, 12 Jan 2009 15:19:25 -0500 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <496B2E4B.7070202@3c.co.uk> References: <496B2E4B.7070202@3c.co.uk> Message-ID: This is great. On Jan 12, 2009, at 6:49 AM, David Knell wrote: > Hi all - > > In case anyone's interested, I've documented how we interfaced FS with > Lumenvox via MRCP using FS' event socket and unicast interfaces and a > bit of Perl here: > http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl > > Three surprises: that it worked at all, that it works quite well and > that it was really quite easy to do. > > One thing I'm looking for: has anyone written a module which > attaches a > bug to an audio stream and forwards the audio as RTP to a specified > IP/port to just allow audio to be tapped off a call and sent somewhere > else to be listened to? > > Cheers -- > > Dave > > -- > David Knell, Director, 3C Limited > T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 > http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From willbelair at yahoo.com Mon Jan 12 12:34:35 2009 From: willbelair at yahoo.com (Will Smith) Date: Mon, 12 Jan 2009 12:34:35 -0800 (PST) Subject: [Freeswitch-users] outbound call, new comer Message-ID: <968078.77098.qm@web53606.mail.re2.yahoo.com> Hi, I am first time FS user, so it is a bit confused with all the setup. For inbound calls, I tried to add a voicepulse.xml in the sip_profiles/external with the following codes: ? ? ??? ??? ??? ??? ??? ??? ? ? ??? ??? ??? ??? ??? ??? ??? ? ? ------------ ?and in the conf/dialplan/default.xml file I added: ? ?? ??? ????? ????? ????? ????? ???? ?? ? ------------------ ? For inbound, I added ? ?? ??? ? ?????? ??? ?? ? ---------- ? And, if I dial 3334445555 from a softphone registered with my_sip_provider, I got the message to the voice mail of 1001 - the 1001 extension does not ring. And if from 1001, I dial some real number like 18188892345, I got the error: Invalid Gateway ... Cannot create outgoing channel of type [fosia] cause: [Invalid_number_format] ... ? ? Would someone please give me some help to set this up. I am a bit confused with these. ? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/afc6e9d0/attachment.html From kristian.kielhofner at gmail.com Mon Jan 12 13:07:21 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 12 Jan 2009 16:07:21 -0500 Subject: [Freeswitch-users] outbound call, new comer In-Reply-To: <968078.77098.qm@web53606.mail.re2.yahoo.com> References: <968078.77098.qm@web53606.mail.re2.yahoo.com> Message-ID: <2d9149cd0901121307veeb1f34s788739e940dbbb60@mail.gmail.com> In West Philadelphia born and raised... Voicepulse seems to be picky about number format. Trying doing full E.164 (+1). Also, make sure your realm is correct. What does a SIP debug look like? On 1/12/09, Will Smith wrote: > > > Hi, > I am first time FS user, so it is a bit confused with all the setup. For inbound calls, I tried to add a voicepulse.xml in the sip_profiles/external with the following codes: > > > > > > > > > > > > > > > > > > > > > > ------------ > and in the conf/dialplan/default.xml file I added: > > > > > > > > > > > > > ------------------ > > For inbound, I added > > > > > > > > ---------- > > And, if I dial 3334445555 from a softphone registered with my_sip_provider, I got the message to the voice mail of 1001 - the 1001 extension does not ring. > And if from 1001, I dial some real number like 18188892345, I got the error: Invalid Gateway ... Cannot create outgoing channel of type [fosia] cause: [Invalid_number_format] ... > > > Would someone please give me some help to set this up. I am a bit confused with these. > > Thank you > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From willbelair at yahoo.com Mon Jan 12 13:21:14 2009 From: willbelair at yahoo.com (Will Smith) Date: Mon, 12 Jan 2009 13:21:14 -0800 (PST) Subject: [Freeswitch-users] outbound call, new comer In-Reply-To: <2d9149cd0901121307veeb1f34s788739e940dbbb60@mail.gmail.com> Message-ID: <318886.13039.qm@web53604.mail.re2.yahoo.com> Forgive me, I don't know how to turn on the SIP debug mode. This is what it say from FS command line: 2009-01-13 16:26:46 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->18187188288in context default 2009-01-13 16:26:47 [ERR] mod_sofia.c:2341 sofia_outgoing_channel() Invalid Gateway 2009-01-13 16:26:47 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2009-01-13 16:26:47 [ERR] switch_ivr_originate.c:1110 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2009-01-13 16:26:47 [INFO] mod_dptools.c:1891 audio_bridge_function() Originate Failed.? Cause: INVALID_NUMBER_FORMAT 2009-01-13 16:26:47 [NOTICE] mod_dptools.c:1918 audio_bridge_function() Hangup sofia/internal/1000 at 192.168.2.104 [CS_EXECUTE] [INVALID_NUMBER_FORMAT] 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 163 (sofia/internal/1000 at 192.168.2.104) Ended 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/internal/1000 at 192.168.2.104 [CS_HANGUP] --- On Mon, 1/12/09, Kristian Kielhofner wrote: From: Kristian Kielhofner Subject: Re: [Freeswitch-users] outbound call, new comer To: freeswitch-users at lists.freeswitch.org Date: Monday, January 12, 2009, 1:07 PM In West Philadelphia born and raised... Voicepulse seems to be picky about number format. Trying doing full E.164 (+1). Also, make sure your realm is correct. What does a SIP debug look like? On 1/12/09, Will Smith wrote: > > > Hi, > I am first time FS user, so it is a bit confused with all the setup. For inbound calls, I tried to add a voicepulse.xml in the sip_profiles/external with the following codes: > > > > > > > > > > > > > > > > > > > > > > ------------ > and in the conf/dialplan/default.xml file I added: > > > > > > > > > > > > > ------------------ > > For inbound, I added > > > > > > > > ---------- > > And, if I dial 3334445555 from a softphone registered with my_sip_provider, I got the message to the voice mail of 1001 - the 1001 extension does not ring. > And if from 1001, I dial some real number like 18188892345, I got the error: Invalid Gateway ... Cannot create outgoing channel of type [fosia] cause: [Invalid_number_format] ... > > > Would someone please give me some help to set this up. I am a bit confused with these. > > Thank you > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/d399969f/attachment.html From brian at freeswitch.org Mon Jan 12 13:29:16 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 12 Jan 2009 15:29:16 -0600 Subject: [Freeswitch-users] outbound call, new comer In-Reply-To: <318886.13039.qm@web53604.mail.re2.yahoo.com> References: <318886.13039.qm@web53604.mail.re2.yahoo.com> Message-ID: <6849BC86-78E9-465C-963F-6EE9FECBC5A6@freeswitch.org> TPORT_LOG=1 ./freeswitch /b On Jan 12, 2009, at 3:21 PM, Will Smith wrote: > Forgive me, > I don't know how to turn on the SIP debug mode. This is what it say > from FS command line: > > 2009-01-13 16:26:46 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 1000->18187188288in context default > 2009-01-13 16:26:47 [ERR] mod_sofia.c:2341 sofia_outgoing_channel() > Invalid Gateway > 2009-01-13 16:26:47 [NOTICE] mod_sofia.c:2540 > sofia_outgoing_channel() Close Channel N/A [CS_NEW] > 2009-01-13 16:26:47 [ERR] switch_ivr_originate.c:1110 > switch_ivr_originate() Cannot create outgoing channel of type > [sofia] cause: [INVALID_NUMBER_FORMAT] > 2009-01-13 16:26:47 [INFO] mod_dptools.c:1891 > audio_bridge_function() Originate Failed. Cause: > INVALID_NUMBER_FORMAT > 2009-01-13 16:26:47 [NOTICE] mod_dptools.c:1918 > audio_bridge_function() Hangup sofia/internal/1000 at 192.168.2.104 > [CS_EXECUTE] [INVALID_NUMBER_FORMAT] > 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:956 > switch_core_session_thread() Session 163 (sofia/internal/1000 at 192.168.2.104 > ) Ended > 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:958 > switch_core_session_thread() Close Channel sofia/internal/1000 at 192.168.2.104 > [CS_HANGUP] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/ffaa646e/attachment-0001.html From kristian.kielhofner at gmail.com Mon Jan 12 13:42:31 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 12 Jan 2009 16:42:31 -0500 Subject: [Freeswitch-users] outbound call, new comer In-Reply-To: <6849BC86-78E9-465C-963F-6EE9FECBC5A6@freeswitch.org> References: <318886.13039.qm@web53604.mail.re2.yahoo.com> <6849BC86-78E9-465C-963F-6EE9FECBC5A6@freeswitch.org> Message-ID: <2d9149cd0901121342g4374198dk5e05e8adeaca1713@mail.gmail.com> I always just use ngrep. As long as you aren't using TLS it works quite well: ngrep -d [device] -q -W byline SIP udp port 5060 Of course you can update your BPF syntax if you wish. On 1/12/09, Brian West wrote: > > TPORT_LOG=1 ./freeswitch > > > /b > > > > On Jan 12, 2009, at 3:21 PM, Will Smith wrote: > > > Forgive me, > I don't know how to turn on the SIP debug mode. This is what it say from FS command line: > > 2009-01-13 16:26:46 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->18187188288in context default > 2009-01-13 16:26:47 [ERR] mod_sofia.c:2341 sofia_outgoing_channel() Invalid Gateway > 2009-01-13 16:26:47 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close Channel N/A [CS_NEW] > 2009-01-13 16:26:47 [ERR] switch_ivr_originate.c:1110 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] > 2009-01-13 16:26:47 [INFO] mod_dptools.c:1891 audio_bridge_function() Originate Failed. Cause: INVALID_NUMBER_FORMAT > 2009-01-13 16:26:47 [NOTICE] mod_dptools.c:1918 audio_bridge_function() Hangup sofia/internal/1000 at 192.168.2.104 [CS_EXECUTE] [INVALID_NUMBER_FORMAT] > 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 163 (sofia/internal/1000 at 192.168.2.104) Ended > 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/internal/1000 at 192.168.2.104 [CS_HANGUP] > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From willbelair at yahoo.com Mon Jan 12 13:53:49 2009 From: willbelair at yahoo.com (Will Smith) Date: Mon, 12 Jan 2009 13:53:49 -0800 (PST) Subject: [Freeswitch-users] outbound call, new comer In-Reply-To: <2d9149cd0901121342g4374198dk5e05e8adeaca1713@mail.gmail.com> Message-ID: <734575.74278.qm@web53609.mail.re2.yahoo.com> This is what I found in the log file. Also, if from command line I type: originate sofia/external/355 at my_sip_provider.com 2009?? , it will ring my extension and play some music (which? is defined in the 2009.xml file) Thank you for all your help -------------------------------- 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [show_info] destination_number(18187188234) =~ /^9992$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [video_record] destination_number(18187188234) =~ /^9993$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [video_playback] destination_number(18187188234) =~ /^9994$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [delay_echo] destination_number(18187188234) =~ /^9995$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [echo] destination_number(18187188234) =~ /^9996$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [milliwatt] destination_number(18187188234) =~ /^9997$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [tone_stream] destination_number(18187188234) =~ /^9998$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [hold_music] destination_number(18187188234) =~ /^9999$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [qtioutbound] destination_number(18187188234) =~ /^(1{0,1}\d{10})$/ 2009-01-13 13:01:58 [DEBUG] switch_core_state_machine.c:100 switch_core_standard_on_routing() (sofia/internal/1000 at 192.168.2.104) State Change CS_ROUTING -> CS_EXECUTE 2009-01-13 13:01:58 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/1000 at 192.168.2.104 [BREAK] ---------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/c50d9868/attachment.html From mszlazak at aol.com Mon Jan 12 15:09:46 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 12 Jan 2009 18:09:46 -0500 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: References: <496B2E4B.7070202@3c.co.uk><3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org><8CB42F4C40F8A82-820-5D0@webmail-me14.sysops.aol.com> Message-ID: <8CB4321FE9C775B-464-99A@WEBMAIL-DF10.sysops.aol.com> That's not the opinion of Nickolay S. from the Sphinx forums. He didn't think it was telephony ready but you implied something similar in a past email. Also, I got a similar impression with the pizza demo as it came with FS. Instead I tried Voxeo's Prophecy as per your recommendation and found it worked better. As I understand it, pocketsphinx and sphinx (3 & 4) are very good but need adapting and training for there various uses.? So, why bother with LumenVox, Voxeo, Nuance, etc if one could get pocketsphinx working better since it's already integrated with FS? -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Mon, 12 Jan 2009 9:55 am Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl Pocketsphinx works great for telephony.. just don't load 10000 word dictionary or grammar :P ?the pizza demo uses it.. and it works great from every phone I have tested it with... Rome wasn't built in a day and we need more people that have the skills to really build a general purpose acoustical model that works in more situations.? /b On Jan 12, 2009, at 11:46 AM, mszlazak at aol.com wrote: Yup, or just get pocketsphinx "tuned" up for telephony and then no one will have to bother with ASR vendors. I believe that some speech data from a good size sample for training is needed to make it more "speaker independent" and better suited for use with phone calls. I have a list of things from the Sphin x forums that would be good to have for a telephony ready PocketSphinx. There is a "wsj" database but I don't know if that's would help?? Best. Mark. = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/6eb6827f/attachment.html From brian at freeswitch.org Mon Jan 12 15:21:24 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 12 Jan 2009 17:21:24 -0600 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <8CB4321FE9C775B-464-99A@WEBMAIL-DF10.sysops.aol.com> References: <496B2E4B.7070202@3c.co.uk><3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org><8CB42F4C40F8A82-820-5D0@webmail-me14.sysops.aol.com> <8CB4321FE9C775B-464-99A@WEBMAIL-DF10.sysops.aol.com> Message-ID: <42B494CA-C9E1-49C5-916E-8447EAFBB508@freeswitch.org> Maybe for NON english speakers it doesn't do well but for my tests and needs it does excellent. Sphinx isn't ready thats for sure.. but PocketSphinx does great. I have PocketSphinx doing voice dial by name directory on very common and simple names. If you adapt it it can get much better. But have you called AT&T lately? I have no idea what they use but OMG it sucks... you say "NO" it doesn't understand you.. you say your account number .. it doesn't understand you... you scream curse words at it and it will take you to an agent so they can get you to the right place. Its aweful. Pocketsphinx has performed better than that on my testing. /b On Jan 12, 2009, at 5:09 PM, mszlazak at aol.com wrote: > That's not the opinion of Nickolay S. from the Sphinx forums. He > didn't think it was telephony ready but you implied something > similar in a past email. Also, I got a similar impression with the > pizza demo as it came with FS. Instead I tried Voxeo's Prophecy as > per your recommendation and found it worked better. As I understand > it, pocketsphinx and sphinx (3 & 4) are very good but need adapting > and training for there various uses. > > So, why bother with LumenVox, Voxeo, Nuance, etc if one could get > pocketsphinx working better since it's already integrated with FS? From anthony.minessale at gmail.com Mon Jan 12 15:22:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 12 Jan 2009 17:22:31 -0600 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <8CB4321FE9C775B-464-99A@WEBMAIL-DF10.sysops.aol.com> References: <496B2E4B.7070202@3c.co.uk> <3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org> <8CB42F4C40F8A82-820-5D0@webmail-me14.sysops.aol.com> <8CB4321FE9C775B-464-99A@WEBMAIL-DF10.sysops.aol.com> Message-ID: <191c3a030901121522j10538c6bq8f1b511c232dd7cb@mail.gmail.com> You should be careful who you ask and who's opinion you weigh against. Just because you go ask some other guy something does not make it gospel. These are the basic facts: ASR/TTS is one of the many areas where we have done a huge amount of pro bono work to get people interested but there is only so much we can do as demo's and examples. We chose to add pocketsphinx because it's free and gives people a way to test. We don't have any monitary support from any of the commerical ASR/TTS providers nor do we have any support from any users who intend to improve the ASR/TTS support. Therefore given the endless hours we spend working on FS we have to decide what to work on and what to put aside. We are currently looking at trying get unimrcp working but again, nobody is willing to pay so the timeline is extended to our spare time. We want to support every ASR/TTS interface we can because it helps to expand our flexablilty but it simply will take a while. On Mon, Jan 12, 2009 at 5:09 PM, wrote: > That's not the opinion of Nickolay S. from the Sphinx forums. He didn't > think it was telephony ready but you implied something similar in a past > email. Also, I got a similar impression with the pizza demo as it came with > FS. Instead I tried Voxeo's Prophecy as per your recommendation and found it > worked better. As I understand it, pocketsphinx and sphinx (3 & 4) are very > good but need adapting and training for there various uses. > So, why bother with LumenVox, Voxeo, Nuance, etc if one could get > pocketsphinx working better since it's already integrated with FS? > > > > -----Original Message----- > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Sent: Mon, 12 Jan 2009 9:55 am > Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl > > Pocketsphinx works great for telephony.. just don't load 10000 word > dictionary or grammar :P the pizza demo uses it.. and it works great from > every phone I have tested it with... Rome wasn't built in a day and we need > more people that have the skills to really build a general purpose > acoustical model that works in more situations. > /b > On Jan 12, 2009, at 11:46 AM, mszlazak at aol.com wrote: > > Yup, or just get pocketsphinx "tuned" up for telephony and then no one will > have to bother with ASR vendors. > > I believe that some speech data from a good size sample for training is > needed to make it more "speaker independent" and better suited for use with > phone calls. I have a list of things from the Sphinx forums that would be > good to have for a telephony ready PocketSphinx. There is a "wsj" database > but I don't know if that's would help?? > > Best. Mark. > > > = > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options > /freeswitch-users http://www.freeswitch.org > > > ------------------------------ > *A Good Credit Score is 700 or Above. See yours in just 2 easy steps! > * > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/7e39a26c/attachment-0001.html From mileskeaton at gmail.com Mon Jan 12 18:47:53 2009 From: mileskeaton at gmail.com (Miles Keaton) Date: Mon, 12 Jan 2009 18:47:53 -0800 Subject: [Freeswitch-users] recommend PSTN / SIP trunking? Message-ID: <59b2d39b0901121847t3edd80f7ha7cf0720e80166da@mail.gmail.com> I'm using Junction Networks (http://pstn.junctionnetworks.com/) for SIP trunking of phone numbers, and I'm happy with them. Their servers are in NYC. Now I'm trying to find a similar company whose servers are based in LA or San Diego. Any recommendations? From AFShin9 at gmail.com Mon Jan 12 19:18:48 2009 From: AFShin9 at gmail.com (Seysan) Date: Mon, 12 Jan 2009 20:18:48 -0700 Subject: [Freeswitch-users] recommend PSTN / SIP trunking? In-Reply-To: <59b2d39b0901121847t3edd80f7ha7cf0720e80166da@mail.gmail.com> References: <59b2d39b0901121847t3edd80f7ha7cf0720e80166da@mail.gmail.com> Message-ID: Do you mean you want DIDs in LA and San Diego Area? On Mon, Jan 12, 2009 at 7:47 PM, Miles Keaton wrote: > I'm using Junction Networks (http://pstn.junctionnetworks.com/) for > SIP trunking of phone numbers, and I'm happy with them. Their servers > are in NYC. > > Now I'm trying to find a similar company whose servers are based in LA > or San Diego. > > Any recommendations? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090112/c77ad020/attachment.html From daldworth at teliax.com Mon Jan 12 19:39:34 2009 From: daldworth at teliax.com (David Aldworth) Date: Mon, 12 Jan 2009 20:39:34 -0700 Subject: [Freeswitch-users] recommend PSTN / SIP trunking? In-Reply-To: <59b2d39b0901121847t3edd80f7ha7cf0720e80166da@mail.gmail.com> References: <59b2d39b0901121847t3edd80f7ha7cf0720e80166da@mail.gmail.com> Message-ID: Hi Miles - Check out http://www.teliax.com Our LA POP is in 1 Wilshire (624 S. Grand). We also have a POP in NYC, Atlanta and Denver. If you have any questions let me know. Or you can give us a call. Cheers, David On Jan 12, 2009, at 7:47 PM, Miles Keaton wrote: > I'm using Junction Networks (http://pstn.junctionnetworks.com/) for > SIP trunking of phone numbers, and I'm happy with them. Their servers > are in NYC. > > Now I'm trying to find a similar company whose servers are based in LA > or San Diego. > > Any recommendations? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mszlazak at aol.com Mon Jan 12 22:55:36 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 13 Jan 2009 01:55:36 -0500 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <42B494CA-C9E1-49C5-916E-8447EAFBB508@freeswitch.org> References: <496B2E4B.7070202@3c.co.uk><3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org><8CB42F4C40F8A82-820-5D0@webmail-me14.sysops.aol.com><8CB4321FE9C775B-464-99A@WEBMAIL-DF10.sysops.aol.com> <42B494CA-C9E1-49C5-916E-8447EAFBB508@freeswitch.org> Message-ID: <8CB436312A08329-80C-1B5D@MBLK-M24.sysops.aol.com> "My god" I would LOVE it if this is really the case and would praise pocketsphinx (PS) and FS to no end. But my experience has been different. First, I tried the pizza demo with a soft phone and later by outside phone calls to my Linksys 3102 pstn-to-voip gateway. Second, I tried these two set-ups again but with Voxeo's Prophecy ASR. Both are as is and by this I mean there was no training of PocketSphinx just running the pizza demo and with Prophecy there is no training because it can't be trained. Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I couldn't use it at a pizza join. Also, I get a much better experience when calling LumenVox and trying their pizza demo. Now, maybe Prophecy is the type of asr that doesn't require hours of training to make it speaker independent. I know that the Sphinx family are the types of ASR that do need this. So, if there is some settings for adaptation of Pocketsphinx for speaker independence then are they turned on? ? How many hours of calls to a business should an owner expect before PocketSphinx gets good enough not to scare customers away? If there are many hours needed then I could see using another ASR in the mean time, recording their calls and feeding the audio to Pocketsphinx for training, then switching to Pocketspinx once it's "tuned up." At least this way a business doesn't have to deal with a "virgin" pocketsphinx. Mark -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Mon, 12 Jan 2009 3:21 pm Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl Maybe for NON english speakers it doesn't do well but for my tests and needs it does excellent. Sphinx isn't ready thats for sure.. but PocketSphinx does great. I have PocketSphinx doing voice dial by name directory on very common and simple names. If you adapt it it can get much better. But have you called AT&T lately? I have no idea what they use but OMG it sucks... you say "NO" it doesn't understand you.. you say your account number .. it doesn't understand you... you scream curse words at it and it will take you to an agent so they can get you to the right place. Its aweful. Pocketsphinx has performed better than that on my testing. /b On Jan 12, 2009, at 5:09 PM, mszlazak at aol.com wrote: > That's not the opinion of Nickolay S. from the Sphinx forums. He > didn't think it was telephony ready but you implied something > similar in a past email. Also, I got a similar impression with the > pizza demo as it came with FS. Instead I tried Voxeo's Prophecy as > per your recommendation and found it worked better. As I understand > it, pocketsphinx and sphinx (3 & 4) are very good but need adapting > and training for there various uses. > > So, why bother with LumenVox, Voxeo, Nuance, etc if one could get > pocketsphinx working better since it's already integrated with FS? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/660a9212/attachment.html From ahgindia308 at gmail.com Tue Jan 13 01:48:44 2009 From: ahgindia308 at gmail.com (ahgindia) Date: Tue, 13 Jan 2009 01:48:44 -0800 (PST) Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <191c3a030901120822i31e6d40dgdc88dae43d49b05b@mail.gmail.com> References: <21386948.post@talk.nabble.com> <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> <21409759.post@talk.nabble.com> <191c3a030901120822i31e6d40dgdc88dae43d49b05b@mail.gmail.com> Message-ID: <21432304.post@talk.nabble.com> Hi, I cannot supply whole code for security purpose. But I can provide some code snippet for the portion you wish. Here is the overview of the code: -> When a call reaches to our server, we will authenticate that call based on originator ip, -> Then we will call a procedure for route lookup, based on the number called. So we will find a terminator who can terminate this number. -> We will send a call to terminator with the number called, and then bridge the originator with the terminator. -> Finally, we insert a CDR for the call with all the details of the caller and terminator and such other things. So, what portion of code should I provide you? Thanks for your cooperation. Anthony Minessale-2 wrote: > > please supply the entire javascript code you were executing. > > > On Mon, Jan 12, 2009 at 1:33 AM, ahgindia wrote: > >> >> Hello, >> >> Here is the output of backtrace : >> http://pastebin.freeswitch.org/6745 >> >> Let me know if find any reason for crash in this trace. >> >> >> Brian West-3 wrote: >> > >> > What is the output of the backtrace? Can you include that in your >> > email? >> > >> > /b >> > >> > On Jan 10, 2009, at 5:01 AM, ahgindia wrote: >> > >> >> >> >> Hi All, >> >> Recently I was testing the new freeswitch release 1.0.2 >> >> The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo >> >> CPU E7200 >> >> @ 2.53GHz processor. >> >> But it crashed, when there were 96 active calls in it (as can be >> >> seen from >> >> "show calls" on freeswitch cli) >> >> There is a dump file for it, in the folder from where i started the >> >> freeswitch. >> >> Let me know how can we know the cause of the crash. >> >> -- >> >> View this message in context: >> >> >> http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21386948.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at 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 at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21409759.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21432304.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ahgindia308 at gmail.com Tue Jan 13 01:56:11 2009 From: ahgindia308 at gmail.com (ahgindia) Date: Tue, 13 Jan 2009 01:56:11 -0800 (PST) Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <191c3a030901120822i31e6d40dgdc88dae43d49b05b@mail.gmail.com> References: <21386948.post@talk.nabble.com> <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> <21409759.post@talk.nabble.com> <191c3a030901120822i31e6d40dgdc88dae43d49b05b@mail.gmail.com> Message-ID: <21432408.post@talk.nabble.com> Hi, I cannot supply whole code for security purpose. But I can provide some code snippet for the portion you wish. Here is the overview of the code: -> When a call reaches to our server, we will authenticate that call based on originator ip, -> Then we will call a procedure for route lookup, based on the number called. So we will find a terminator who can terminate this number. -> We will send a call to terminator with the number called, and then bridge the originator with the terminator. -> Finally, we insert a CDR for the call with all the details of the caller and terminator and such other things. So, what portion of code should I provide you? Thanks for your cooperation. Anthony Minessale-2 wrote: > > please supply the entire javascript code you were executing. > > > On Mon, Jan 12, 2009 at 1:33 AM, ahgindia wrote: > >> >> Hello, >> >> Here is the output of backtrace : >> http://pastebin.freeswitch.org/6745 >> >> Let me know if find any reason for crash in this trace. >> >> >> Brian West-3 wrote: >> > >> > What is the output of the backtrace? Can you include that in your >> > email? >> > >> > /b >> > >> > On Jan 10, 2009, at 5:01 AM, ahgindia wrote: >> > >> >> >> >> Hi All, >> >> Recently I was testing the new freeswitch release 1.0.2 >> >> The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo >> >> CPU E7200 >> >> @ 2.53GHz processor. >> >> But it crashed, when there were 96 active calls in it (as can be >> >> seen from >> >> "show calls" on freeswitch cli) >> >> There is a dump file for it, in the folder from where i started the >> >> freeswitch. >> >> Let me know how can we know the cause of the crash. >> >> -- >> >> View this message in context: >> >> >> http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21386948.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at 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 at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21409759.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21432408.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ahgindia308 at gmail.com Tue Jan 13 02:29:22 2009 From: ahgindia308 at gmail.com (ahgindia) Date: Tue, 13 Jan 2009 02:29:22 -0800 (PST) Subject: [Freeswitch-users] Re ad application not working as intended In-Reply-To: <191c3a030901100843j7de39a8dkb47642185de2b90b@mail.gmail.com> References: <21386369.post@talk.nabble.com> <654BDDF5-750D-4851-90A9-FEF9E1C8DAA8@freeswitch.org> <191c3a030901100843j7de39a8dkb47642185de2b90b@mail.gmail.com> Message-ID: <21432868.post@talk.nabble.com> The issue here was that when we a minimum of 0 or 1 DTMF input from user and hangs up. If we change that minimum to 2 or more, it correctly accepts the DTMF from user upto maximum specified limit and then hang up. Also there is an issue with minimum limit of DTMF input. If we keep a mimimum limit to 4, still the user is able to input two DTMF and press #, instead of minimum required DTMF input from user. Is this a bug or am I missing something. Anthony Minessale-2 wrote: > > escape the $ in ${res} with a \ so it will not be evaluated by the > dialplan. > > > > > On Sat, Jan 10, 2009 at 9:53 AM, Brian West wrote: > >> You need to use execute_extension so that ${res} will be expanded. >> FreeSWITCH doesn't expand vars as it goes like asterisk does. >> >> Easy way to see the var is to use the info app. >> /b >> >> Sent from my iPhone >> >> On Jan 10, 2009, at 3:34 AM, ahgindia wrote: >> >> > >> > Hi all, >> > I want have DTMF input from user without answering the call (i.e. in >> > session >> > progress). >> > I used Read application for this, but it is not working as it is >> > intended to >> > be. >> > It is only accepting one DTMF from the user and the channel gets >> > hang up. >> > Am I missing something? >> > Here is the dialplan that I used for this: >> > >> > >> > The phrase application is used to check user input. >> > -- >> > View this message in context: >> http://www.nabble.com/Read-application-not-working-as-intended-tp21386369p21386369.html >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Read-application-not-working-as-intended-tp21386369p21432868.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From pmhshz at gmail.com Tue Jan 13 02:48:49 2009 From: pmhshz at gmail.com (shehzad p) Date: Tue, 13 Jan 2009 02:48:49 -0800 (PST) Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <428196FD-5E22-4745-BE8D-5C4AFFC3B2CF@freeswitch.org> References: <21386948.post@talk.nabble.com> <21414332.post@talk.nabble.com> <428196FD-5E22-4745-BE8D-5C4AFFC3B2CF@freeswitch.org> Message-ID: <21433120.post@talk.nabble.com> Please find the output of bt from below pastebin link: http://pastebin.freeswitch.org/6757 Thanks, pms Michael S Collins wrote: > > Could you please do a backtrace and post it to a pastebin? If in Linux > do this: > gdb /path/to/freeswitch /path/to/corefile > > -MC > > Sent from my iPhone > > On Jan 12, 2009, at 5:23 AM, shehzad p wrote: > >> >> Hi all, >> I am also testing FS release 1.0.2, but I faced strange problem. >> When I stop freeswitch (from CLI using ... or shutdown), Freeswitch >> ends >> with showing "Segmentation fault": >> Below is the last 15 lines when fault occures. Sometimes this does not >> happen and FS shut down normally. >> >> === >> === >> === >> === >> === >> ====================================================================== >> 2009-01-12 16:52:56 [CONSOLE] switch_loadable_module.c:1244 >> do_shutdown() >> mod_esf unloaded. >> 2009-01-12 16:52:56 [CONSOLE] switch_core.c:1462 switch_core_destroy() >> Closing Event Engine. >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 >> switch_event_shutdown() >> Stopping event queue 0 >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 >> switch_event_shutdown() >> Stopping event queue 1 >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 >> switch_event_thread() Event >> Thread 0 Ended. >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:445 >> switch_event_shutdown() >> Stopping dispatch queue 0 >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 >> switch_event_thread() Event >> Thread 1 Ended. >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:247 >> switch_event_dispatch_thread() Dispatch Thread 0 Ended. >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 >> switch_event_thread() Event >> Thread 2 Ended. >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:414 >> switch_core_memory_reclaim_events() Returning 23 recycled event(s) >> 1012 >> bytes >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:416 >> switch_core_memory_reclaim_events() Returning 331 recycled event >> header(s) >> 5296 bytes >> 2009-01-12 16:52:56 [CONSOLE] switch_core_sqldb.c:539 >> switch_core_sqldb_stop() Waiting for unfinished SQL transactions >> 2009-01-12 16:52:56 [NOTICE] switch_core_sqldb.c:199 >> switch_core_sql_thread() SQL thread ending >> 2009-01-12 16:52:56 [CONSOLE] switch_scheduler.c:303 >> switch_scheduler_task_thread_stop() Stopping Task Thread >> Segmentation fault (core dumped) >> === >> === >> === >> === >> === >> ====================================================================== >> >> What should be the cause of such crash. >> >> >> ahgindia wrote: >>> >>> Hi All, >>> Recently I was testing the new freeswitch release 1.0.2 >>> The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo CPU >>> E7200 @ 2.53GHz processor. >>> But it crashed, when there were 96 active calls in it (as can be >>> seen from >>> "show calls" on freeswitch cli) >>> There is a dump file for it, in the folder from where i started the >>> freeswitch. >>> Let me know how can we know the cause of the crash. >>> >> >> >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21414332.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21433120.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From regs at kinetix.gr Tue Jan 13 03:46:04 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 13 Jan 2009 13:46:04 +0200 Subject: [Freeswitch-users] mod_opal first unsuccessful test In-Reply-To: <024701c97510$cb9a1890$62ce49b0$@jongbloed@bigpond.com> References: <496B66F3.9000902@kinetix.gr> <191c3a030901121119o688c722er745106b014367569@mail.gmail.com> <024701c97510$cb9a1890$62ce49b0$@jongbloed@bigpond.com> Message-ID: <496C7EFC.8060401@kinetix.gr> Hi, Yes, openPhone is working with my soundcard. I am using it every day for testing purposes. I use the 1.8.1 version. Is there a newer version that uses OPAL? I didn't know that. Where can I get it from? Robert Jongbloed wrote: > > Hi guys, > > > > I was using the OpenPhone that you build with OPAL for my testing. So > that is identical (I think) to you. > > > > I have not (yet) do any third party client testing. > > > > Two ALERTING messages are fine, perfectly legal and OPAL can handle it. > > > > You say you can see the RTP packets flowing so that implies that the > mod_opal is actually working, so let's look somewhere else. Have you > confirmed that OpenPhone is using the sound card correctly? Made a > call between two machines JUST using OpenPhone for example? > > > > > > Robert Jongbloed > > OPAL/OpenH323/PTLib Architect and Co-founder. > > > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Tuesday, 13 January 2009 6:20 AM > *To:* Robert Jongbloed > *Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test > > > > Heh, > what client are you using in your tests that are working? > > ---------- Forwarded message ---------- > From: *Apostolos Pantsiopoulos* > > Date: Mon, Jan 12, 2009 at 9:51 AM > Subject: [Freeswitch-users] mod_opal first unsuccessful test > To: freeswitch-users at lists.freeswitch.org > > > > Hi, > > I successfully compiled mod_opal using the latest svn for both opal > and ptlib as Brian suggested. > > When I try to establish a call using h323 from my openphone client > I get no audio although I can see RTP packets in both directions when I am > doing a capture. Some details : > > I am using the 11094 revision of the FS trunk. > I am using the PCMU codec. > I tried dialing the default IVR (5000) and other testing extensions > (freeswitch conference, echo test etc.) > I tried using fast start on and off , h245 tunneling on and off, h245 in > SETUP on and off. > > In my captures I have also noticed a strange behavior : FS sends to > my client 2 "alerting" packets > for no apparent reason. Could this be a cause of the problem? > > Had anyone any success with mod_opal lately? If yes, could you > please reply quoting your config > options (both on FS and on your client)? > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/8f1c84ef/attachment.html From regs at kinetix.gr Tue Jan 13 04:04:30 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 13 Jan 2009 14:04:30 +0200 Subject: [Freeswitch-users] mod_opal first unsuccessful test In-Reply-To: <496C7EFC.8060401@kinetix.gr> References: <496B66F3.9000902@kinetix.gr> <191c3a030901121119o688c722er745106b014367569@mail.gmail.com> <024701c97510$cb9a1890$62ce49b0$@jongbloed@bigpond.com> <496C7EFC.8060401@kinetix.gr> Message-ID: <496C834E.9070608@kinetix.gr> I also tried using Ekiga - which is OPAL based - and got the same behavior. No audio - although I can see RTP packets. Apostolos Pantsiopoulos wrote: > Hi, > > Yes, openPhone is working with my soundcard. I am using it > every day for testing purposes. I use the 1.8.1 version. Is there a newer > version that uses OPAL? I didn't know that. Where can I get it from? > > Robert Jongbloed wrote: >> >> Hi guys, >> >> >> >> I was using the OpenPhone that you build with OPAL for my testing. So >> that is identical (I think) to you. >> >> >> >> I have not (yet) do any third party client testing. >> >> >> >> Two ALERTING messages are fine, perfectly legal and OPAL can handle it. >> >> >> >> You say you can see the RTP packets flowing so that implies that the >> mod_opal is actually working, so let's look somewhere else. Have you >> confirmed that OpenPhone is using the sound card correctly? Made a >> call between two machines JUST using OpenPhone for example? >> >> >> >> >> >> Robert Jongbloed >> >> OPAL/OpenH323/PTLib Architect and Co-founder. >> >> >> >> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] >> *Sent:* Tuesday, 13 January 2009 6:20 AM >> *To:* Robert Jongbloed >> *Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test >> >> >> >> Heh, >> what client are you using in your tests that are working? >> >> ---------- Forwarded message ---------- >> From: *Apostolos Pantsiopoulos* > > >> Date: Mon, Jan 12, 2009 at 9:51 AM >> Subject: [Freeswitch-users] mod_opal first unsuccessful test >> To: freeswitch-users at lists.freeswitch.org >> >> >> >> Hi, >> >> I successfully compiled mod_opal using the latest svn for both opal >> and ptlib as Brian suggested. >> >> When I try to establish a call using h323 from my openphone client >> I get no audio although I can see RTP packets in both directions when >> I am >> doing a capture. Some details : >> >> I am using the 11094 revision of the FS trunk. >> I am using the PCMU codec. >> I tried dialing the default IVR (5000) and other testing extensions >> (freeswitch conference, echo test etc.) >> I tried using fast start on and off , h245 tunneling on and off, h245 in >> SETUP on and off. >> >> In my captures I have also noticed a strange behavior : FS sends to >> my client 2 "alerting" packets >> for no apparent reason. Could this be a cause of the problem? >> >> Had anyone any success with mod_opal lately? If yes, could you >> please reply quoting your config >> options (both on FS and on your client)? >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/1f4c5206/attachment-0001.html From anthony.minessale at gmail.com Tue Jan 13 06:00:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 13 Jan 2009 08:00:31 -0600 Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <21432304.post@talk.nabble.com> References: <21386948.post@talk.nabble.com> <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> <21409759.post@talk.nabble.com> <191c3a030901120822i31e6d40dgdc88dae43d49b05b@mail.gmail.com> <21432304.post@talk.nabble.com> Message-ID: <191c3a030901130600m46d6348dlb9ab1d859d47b321@mail.gmail.com> So I can supply you with 250 thousand lines of C code that make your application possible. but you are not willing to show me the silly js code that may be the cause of your crash? What security purposes are you kidding? just rename any sensitive data to something else or stop using js because without seeing the script code that's all I can tell you as the solution to your problem. On Tue, Jan 13, 2009 at 3:48 AM, ahgindia wrote: > > Hi, > > I cannot supply whole code for security purpose. > But I can provide some code snippet for the portion you wish. > Here is the overview of the code: > -> When a call reaches to our server, we will authenticate that call based > on originator ip, > -> Then we will call a procedure for route lookup, based on the number > called. So we will find a terminator who can terminate this number. > -> We will send a call to terminator with the number called, and then > bridge > the originator with the terminator. > -> Finally, we insert a CDR for the call with all the details of the caller > and terminator and such other things. > > So, what portion of code should I provide you? > Thanks for your cooperation. > > > Anthony Minessale-2 wrote: > > > > please supply the entire javascript code you were executing. > > > > > > On Mon, Jan 12, 2009 at 1:33 AM, ahgindia wrote: > > > >> > >> Hello, > >> > >> Here is the output of backtrace : > >> http://pastebin.freeswitch.org/6745 > >> > >> Let me know if find any reason for crash in this trace. > >> > >> > >> Brian West-3 wrote: > >> > > >> > What is the output of the backtrace? Can you include that in your > >> > email? > >> > > >> > /b > >> > > >> > On Jan 10, 2009, at 5:01 AM, ahgindia wrote: > >> > > >> >> > >> >> Hi All, > >> >> Recently I was testing the new freeswitch release 1.0.2 > >> >> The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo > >> >> CPU E7200 > >> >> @ 2.53GHz processor. > >> >> But it crashed, when there were 96 active calls in it (as can be > >> >> seen from > >> >> "show calls" on freeswitch cli) > >> >> There is a dump file for it, in the folder from where i started the > >> >> freeswitch. > >> >> Let me know how can we know the cause of the crash. > >> >> -- > >> >> View this message in context: > >> >> > >> > http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21386948.html > >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> >> > >> >> > >> >> _______________________________________________ > >> >> Freeswitch-users mailing list > >> >> Freeswitch-users at 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 at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21409759.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at hotmail.com < > MSN%3Aanthony_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 < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21432304.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/6a62182b/attachment.html From anthony.minessale at gmail.com Tue Jan 13 06:01:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 13 Jan 2009 08:01:08 -0600 Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <21432304.post@talk.nabble.com> References: <21386948.post@talk.nabble.com> <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> <21409759.post@talk.nabble.com> <191c3a030901120822i31e6d40dgdc88dae43d49b05b@mail.gmail.com> <21432304.post@talk.nabble.com> Message-ID: <191c3a030901130601r78b55fd8y7ff14f22099abd34@mail.gmail.com> in the future please start your own thread. Also do not report bugs to the mailing list, use http://jira.freeswitch.org On Tue, Jan 13, 2009 at 3:48 AM, ahgindia wrote: > > Hi, > > I cannot supply whole code for security purpose. > But I can provide some code snippet for the portion you wish. > Here is the overview of the code: > -> When a call reaches to our server, we will authenticate that call based > on originator ip, > -> Then we will call a procedure for route lookup, based on the number > called. So we will find a terminator who can terminate this number. > -> We will send a call to terminator with the number called, and then > bridge > the originator with the terminator. > -> Finally, we insert a CDR for the call with all the details of the caller > and terminator and such other things. > > So, what portion of code should I provide you? > Thanks for your cooperation. > > > Anthony Minessale-2 wrote: > > > > please supply the entire javascript code you were executing. > > > > > > On Mon, Jan 12, 2009 at 1:33 AM, ahgindia wrote: > > > >> > >> Hello, > >> > >> Here is the output of backtrace : > >> http://pastebin.freeswitch.org/6745 > >> > >> Let me know if find any reason for crash in this trace. > >> > >> > >> Brian West-3 wrote: > >> > > >> > What is the output of the backtrace? Can you include that in your > >> > email? > >> > > >> > /b > >> > > >> > On Jan 10, 2009, at 5:01 AM, ahgindia wrote: > >> > > >> >> > >> >> Hi All, > >> >> Recently I was testing the new freeswitch release 1.0.2 > >> >> The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo > >> >> CPU E7200 > >> >> @ 2.53GHz processor. > >> >> But it crashed, when there were 96 active calls in it (as can be > >> >> seen from > >> >> "show calls" on freeswitch cli) > >> >> There is a dump file for it, in the folder from where i started the > >> >> freeswitch. > >> >> Let me know how can we know the cause of the crash. > >> >> -- > >> >> View this message in context: > >> >> > >> > http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21386948.html > >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> >> > >> >> > >> >> _______________________________________________ > >> >> Freeswitch-users mailing list > >> >> Freeswitch-users at 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 at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21409759.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at hotmail.com < > MSN%3Aanthony_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 < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21432304.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/bd91836e/attachment-0001.html From anthony.minessale at gmail.com Tue Jan 13 06:01:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 13 Jan 2009 08:01:57 -0600 Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <21433120.post@talk.nabble.com> References: <21386948.post@talk.nabble.com> <21414332.post@talk.nabble.com> <428196FD-5E22-4745-BE8D-5C4AFFC3B2CF@freeswitch.org> <21433120.post@talk.nabble.com> Message-ID: <191c3a030901130601m7f94fd15ra7c05d35b6408497@mail.gmail.com> please remove FS src and dest dir from your machine and recompile fresh from scratch. On Tue, Jan 13, 2009 at 4:48 AM, shehzad p wrote: > > > Please find the output of bt from below pastebin link: > http://pastebin.freeswitch.org/6757 > > Thanks, > pms > > Michael S Collins wrote: > > > > Could you please do a backtrace and post it to a pastebin? If in Linux > > do this: > > gdb /path/to/freeswitch /path/to/corefile > > > > -MC > > > > Sent from my iPhone > > > > On Jan 12, 2009, at 5:23 AM, shehzad p wrote: > > > >> > >> Hi all, > >> I am also testing FS release 1.0.2, but I faced strange problem. > >> When I stop freeswitch (from CLI using ... or shutdown), Freeswitch > >> ends > >> with showing "Segmentation fault": > >> Below is the last 15 lines when fault occures. Sometimes this does not > >> happen and FS shut down normally. > >> > >> === > >> === > >> === > >> === > >> === > >> ====================================================================== > >> 2009-01-12 16:52:56 [CONSOLE] switch_loadable_module.c:1244 > >> do_shutdown() > >> mod_esf unloaded. > >> 2009-01-12 16:52:56 [CONSOLE] switch_core.c:1462 switch_core_destroy() > >> Closing Event Engine. > >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 > >> switch_event_shutdown() > >> Stopping event queue 0 > >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 > >> switch_event_shutdown() > >> Stopping event queue 1 > >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 > >> switch_event_thread() Event > >> Thread 0 Ended. > >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:445 > >> switch_event_shutdown() > >> Stopping dispatch queue 0 > >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 > >> switch_event_thread() Event > >> Thread 1 Ended. > >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:247 > >> switch_event_dispatch_thread() Dispatch Thread 0 Ended. > >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 > >> switch_event_thread() Event > >> Thread 2 Ended. > >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:414 > >> switch_core_memory_reclaim_events() Returning 23 recycled event(s) > >> 1012 > >> bytes > >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:416 > >> switch_core_memory_reclaim_events() Returning 331 recycled event > >> header(s) > >> 5296 bytes > >> 2009-01-12 16:52:56 [CONSOLE] switch_core_sqldb.c:539 > >> switch_core_sqldb_stop() Waiting for unfinished SQL transactions > >> 2009-01-12 16:52:56 [NOTICE] switch_core_sqldb.c:199 > >> switch_core_sql_thread() SQL thread ending > >> 2009-01-12 16:52:56 [CONSOLE] switch_scheduler.c:303 > >> switch_scheduler_task_thread_stop() Stopping Task Thread > >> Segmentation fault (core dumped) > >> === > >> === > >> === > >> === > >> === > >> ====================================================================== > >> > >> What should be the cause of such crash. > >> > >> > >> ahgindia wrote: > >>> > >>> Hi All, > >>> Recently I was testing the new freeswitch release 1.0.2 > >>> The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo CPU > >>> E7200 @ 2.53GHz processor. > >>> But it crashed, when there were 96 active calls in it (as can be > >>> seen from > >>> "show calls" on freeswitch cli) > >>> There is a dump file for it, in the folder from where i started the > >>> freeswitch. > >>> Let me know how can we know the cause of the crash. > >>> > >> > >> > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21414332.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21433120.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/09dc290a/attachment.html From anthony.minessale at gmail.com Tue Jan 13 06:10:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 13 Jan 2009 08:10:11 -0600 Subject: [Freeswitch-users] Re ad application not working as intended In-Reply-To: <21432868.post@talk.nabble.com> References: <21386369.post@talk.nabble.com> <654BDDF5-750D-4851-90A9-FEF9E1C8DAA8@freeswitch.org> <191c3a030901100843j7de39a8dkb47642185de2b90b@mail.gmail.com> <21432868.post@talk.nabble.com> Message-ID: <191c3a030901130610h54afc1adxbcbb7f5e94e3f053@mail.gmail.com> Are you using code >= 1.0.2 ? There were a few changes to the read app and it has been confirmed to work properly. If you fall below min digits the app will exit anyway and the var "read_result" will be set to "failure" On Tue, Jan 13, 2009 at 4:29 AM, ahgindia wrote: > > The issue here was that when we a minimum of 0 or 1 DTMF input from user > and > hangs up. > If we change that minimum to 2 or more, it correctly accepts the DTMF from > user upto maximum specified limit and then hang up. > Also there is an issue with minimum limit of DTMF input. > > If we keep a mimimum limit to 4, still the user is able to input two DTMF > and press #, instead of minimum required DTMF input from user. > Is this a bug or am I missing something. > > > Anthony Minessale-2 wrote: > > > > escape the $ in ${res} with a \ so it will not be evaluated by the > > dialplan. > > > > > > > > > > On Sat, Jan 10, 2009 at 9:53 AM, Brian West > wrote: > > > >> You need to use execute_extension so that ${res} will be expanded. > >> FreeSWITCH doesn't expand vars as it goes like asterisk does. > >> > >> Easy way to see the var is to use the info app. > >> /b > >> > >> Sent from my iPhone > >> > >> On Jan 10, 2009, at 3:34 AM, ahgindia wrote: > >> > >> > > >> > Hi all, > >> > I want have DTMF input from user without answering the call (i.e. in > >> > session > >> > progress). > >> > I used Read application for this, but it is not working as it is > >> > intended to > >> > be. > >> > It is only accepting one DTMF from the user and the channel gets > >> > hang up. > >> > Am I missing something? > >> > Here is the dialplan that I used for this: > >> > > >> > > >> > The phrase application is used to check user input. > >> > -- > >> > View this message in context: > >> > http://www.nabble.com/Read-application-not-working-as-intended-tp21386369p21386369.html > >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at 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 at 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 at hotmail.com < > MSN%3Aanthony_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 < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Read-application-not-working-as-intended-tp21386369p21432868.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/53761bda/attachment-0001.html From sellsky at mail.ru Tue Jan 13 03:07:09 2009 From: sellsky at mail.ru (sellsky at mail.ru) Date: Tue, 13 Jan 2009 14:07:09 +0300 Subject: [Freeswitch-users] mod_java under MS Windows - is it possible? Message-ID: <108017052.20090113140709@mail.ru> Hello, I'm new to FreeSwitch, and now I'm trying to set it up. The problem is, I want to use mod_java under MS Windows. I uncomment this line for mod_java in modules.conf.xml, and I got an error "Error Loading module W:\FreeSWITCH\mod\mod_java.dll". Yes, there is no such dll. Next I tried to compile FreeSwitch myself, and I found there is no MSVC project file for mod_java. I sure can't deal with it myself. Is it possible to make mod_java.dll? -- Best regards, sellsky mailto:sellsky at mail.ru From intralanman at freeswitch.org Tue Jan 13 08:10:41 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 13 Jan 2009 16:10:41 +0000 Subject: [Freeswitch-users] SBC configuration In-Reply-To: References: <150a3aa50901120002x69967134v51b3ba0f13fac6b3@mail.gmail.com> Message-ID: <496CBD01.6030406@freeswitch.org> David Knell wrote: > Hi Ash, > > It's the former. Here's a snippet from a dialplan of ours - this > takes calls with a specific prefix from a specific IP address and > forwards them to a particular carrier: > > > > i'd personally use the acl stuff here instead of manually putting regexes in a dialplan to match ip addresses. > > > > > > and depending on what exactly you want your sbc to do, lcr here might work better for you. -Ray -------------- next part -------------- A non-text attachment was scrubbed... Name: intralanman.vcf Type: text/x-vcard Size: 209 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/771469da/attachment.vcf From paulh at instruments.com Tue Jan 13 08:18:32 2009 From: paulh at instruments.com (Paul Herring) Date: Tue, 13 Jan 2009 10:18:32 -0600 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: References: Message-ID: What would it take to put a budget together to for this project? Date: Tue, 13 Jan 2009 01:55:36 -0500 From: mszlazak at aol.com Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl To: freeswitch-users at lists.freeswitch.org Message-ID: <8CB436312A08329-80C-1B5D at MBLK-M24.sysops.aol.com> Content-Type: text/plain; charset="us-ascii" "My god" I would LOVE it if this is really the case and would praise pocketsphinx (PS) and FS to no end. But my experience has been different. First, I tried the pizza demo with a soft phone and later by outside phone calls to my Linksys 3102 pstn-to-voip gateway. Second, I tried these two set-ups again but with Voxeo's Prophecy ASR. Both are as is and by this I mean there was no training of PocketSphinx just running the pizza demo and with Prophecy there is no training because it can't be trained. Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I couldn't use it at a pizza join. Also, I get a much better experience when calling LumenVox and trying their pizza demo. Now, maybe Prophecy is the type of asr that doesn't require hours of training to make it speaker independent. I know that the Sphinx family are the types of ASR that do need this. So, if there is some settings for adaptation of Pocketsphinx for speaker independence then are they turned on? ? How many hours of calls to a business should an owner expect before PocketSphinx gets good enough not to scare customers away? If there are many hours needed then I could see using another ASR in the mean time, recording their calls and feeding the audio to Pocketsphinx for training, then switching to Pocketspinx once it's "tuned up." At least this way a business doesn't have to deal with a "virgin" pocketsphinx. Mark -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul Herring.vcf Type: text/x-vcard Size: 540 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/7aaccea7/attachment.vcf From regs at kinetix.gr Tue Jan 13 08:45:45 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 13 Jan 2009 18:45:45 +0200 Subject: [Freeswitch-users] Gateways vs Directory Users Message-ID: <496CC539.6060900@kinetix.gr> I think one of the most difficult ideas to grasp when someone migrates to FS from other soft-switches is the definition of endpoints (in the generic meaning of the word) i.e. other gateways that your FS gateway will exchange traffic with. From what I understood from the wiki : a user (directory) is an entitiy that defines our subscribers (with registration or not) a gateway is another VoIP gateway that we use to send traffic to (provider) and sometimes receive traffic from (correct me If I am wrong this far) Now, I have the following scenario : I have an FS box that will exhchange traffic with certain gateways. Sometimes a gateway will act as a client (customer) or a provider, depending on the traffic direction. 1st approach : The first plan I tried, was to use the directory to "define" these other gateways since they are clients. I turned off registrations in my SIP profile (because I don't need any), I wrote an xml file for each gateway in the directory with the appropriate cidr and that was it. I then realized that I could not use my definitions in the users directory so that I can send a call to that client. In order to do so, I had to define that particular IP as a gateway in my SIP profile section. 2nd approach : I forgot all about the directory thing (after all I needed not any mailboxes or registrations) and tried to implement the same scenario by just using gateways. I declared the gateways in my SIP profile and was able to dial out to them just by using : which was fine. But when I got an inbound call from that gateway I realized (1) that I had to define this gateway in an acl and (2) the dialplan used (context) was always the inbound one since there is no way of telling a gateway which context to use when accepting a call. I found a way in the wiki that suggests using DIDs to accomplish that, but I find it this to be not a very elegant solution because I don't want to use prefixes or DIDs to accomplish my goal. I want to be able to define a gateway using a gateway name (label) of my choice and an IP address, so when I come to the point of having 100+ gateways i won't have to search wich DID to use for a new one. 3rd approach : Use a combination of the above. Declare a gateway that receives and sends traffic twice. Once as a user in the directory and once in the gateways section of my profile. Does all these sound correct to you? Am I missing something? Which approach is the best? Please send me your feedback. Inline comments would be ideal. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From anthony.minessale at gmail.com Tue Jan 13 08:58:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 13 Jan 2009 10:58:08 -0600 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <8CB436312A08329-80C-1B5D@MBLK-M24.sysops.aol.com> References: <496B2E4B.7070202@3c.co.uk> <3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org> <8CB42F4C40F8A82-820-5D0@webmail-me14.sysops.aol.com> <8CB4321FE9C775B-464-99A@WEBMAIL-DF10.sysops.aol.com> <42B494CA-C9E1-49C5-916E-8447EAFBB508@freeswitch.org> <8CB436312A08329-80C-1B5D@MBLK-M24.sysops.aol.com> Message-ID: <191c3a030901130858n763b7662p529f55375e94409d@mail.gmail.com> I'm missing something, What is your point exactly? I *just* explained that we want to support unimrcp so you can use Prophecy if you wish so we get it, there is no need to continue to complain. I tried to tell you that we are short on time and we are trying our best. We had openmrcp and the devloper discontinued the project. Now we need to get rid of it and switch to unimrcp. If you recall, you called us for consulting, then spent an hour on the phone gathering free information then proceeded to get all kinds of free help on this list using the free software we have made available to you. It's great that Prophecy is the only place you want to spend any money and I encourage you to do so we can connect you with Voxeo any time. But what else exactly do you want from us? You may want to factor in that your limited experience and particular requirements contribute to your trouble setting everything up so clearly the pocketsphinx route is not for you. (You are only the 2nd person to try it on windows for instance) I keep reading all of your emails and I am trying to understand exactly what you want from us. On Tue, Jan 13, 2009 at 12:55 AM, wrote: > "My god" I would LOVE it if this is really the case and would praise > pocketsphinx (PS) and FS to no end. But my experience has been different. > > First, I tried the pizza demo with a soft phone and later by outside phone > calls to my Linksys 3102 pstn-to-voip gateway. > Second, I tried these two set-ups again but with Voxeo's Prophecy ASR. > > Both are as is and by this I mean there was no training of PocketSphinx > just running the pizza demo and with Prophecy there is no training because > it can't be trained. > > Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I > couldn't use it at a pizza join. Also, I get a much better experience when > calling LumenVox and trying their pizza demo. > > Now, maybe Prophecy is the type of asr that doesn't require hours of > training to make it speaker independent. I know that the Sphinx family are > the types of ASR that do need this. > > So, if there is some settings for adaptation of Pocketsphinx for speaker > independence then are they turned on? > > How many hours of calls to a business should an owner expect before > PocketSphinx gets good enough not to scare customers away? > > If there are many hours needed then I could see using another ASR in the > mean time, recording their calls and feeding the audio to Pocketsphinx for > training, then switching to Pocketspinx once it's "tuned up." At least this > way a business doesn't have to deal with a "virgin" pocketsphinx. > > Mark > > -----Original Message----- > From: Brian West > To: freeswitch-users at lists.freeswitch.org > Sent: Mon, 12 Jan 2009 3:21 pm > Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl > > Maybe for NON english speakers it doesn't do well but for my tests and > > needs it does excellent. Sphinx isn't ready thats for sure.. but > > PocketSphinx does great. > > > I have PocketSphinx doing voice dial by name directory on very common > > and simple names. If you adapt it it can get much better. But have > > you called AT&T lately? I have no idea what they use but OMG it > > sucks... you say "NO" it doesn't understand you.. you say your account > > number .. it doesn't understand you... you scream curse words at it > > and it will take you to an agent so they can get you to the right > > place. Its aweful. Pocketsphinx has performed better than that on my > > testing. > > > /b > > > > On Jan 12, 2009, at 5:09 PM, mszlazak at aol.com wrote: > > > > That's not the opinion of Nickolay S. from the Sphinx forums. He > > > didn't think it was telephony ready but you implied something > > > similar in a past email. Also, I got a similar impression with the > > > pizza demo as it came with FS. Instead I tried Voxeo's Prophecy as > > > per your recommendation and found it worked better. As I understand > > > it, pocketsphinx and sphinx (3 & 4) are very good but need adapting > > > and training for there various uses. > > > > > > So, why bother with LumenVox, Voxeo, Nuance, etc if one could get > > > pocketsphinx working better since it's already integrated with FS? > > > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > *A Good Credit Score is 700 or Above. See yours in just 2 easy steps! > * > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/03e80f3a/attachment-0001.html From anthony.minessale at gmail.com Tue Jan 13 09:00:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 13 Jan 2009 11:00:50 -0600 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: References: Message-ID: <191c3a030901130900s7177ab6did2c58c2f14d2d05d@mail.gmail.com> Maybe Arsen can chime in. He is the author of unimrcp and the previous openmrcp. I would assume it would be a few k to hire some developers to work on the module. The new library is working so it should be straightforward. I am willing to help answer questions for whomever wants to code it and we will even try to get the module started but we need to add it to a very large todo list. On Tue, Jan 13, 2009 at 10:18 AM, Paul Herring wrote: > What would it take to put a budget together to for this project? > > > Date: Tue, 13 Jan 2009 01:55:36 -0500 > From: mszlazak at aol.com > Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl > To: freeswitch-users at lists.freeswitch.org > Message-ID: <8CB436312A08329-80C-1B5D at MBLK-M24.sysops.aol.com> > Content-Type: text/plain; charset="us-ascii" > > > "My god" I would LOVE it if this is really the case and would praise > pocketsphinx (PS) and FS to no end. But my experience has been different. > > First, I tried the pizza demo with a soft phone and later by outside phone > calls to my Linksys 3102 pstn-to-voip gateway. > Second, I tried these two set-ups again but with Voxeo's Prophecy ASR. > > Both are as is and by this I mean there was no training of PocketSphinx > just > running the pizza demo and with Prophecy there is no training because it > can't be trained. > > Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I > couldn't use it at a pizza join. Also, I get a much better experience when > calling LumenVox and trying their pizza demo. > > Now, maybe Prophecy is the type of asr that doesn't require hours of > training to make it speaker independent. I know that the Sphinx family are > the types of ASR that do need this. > > So, if there is some settings for adaptation of Pocketsphinx for speaker > independence then are they turned on? > ? > How many hours of calls to a business should an owner expect before > PocketSphinx gets good enough not to scare customers away? > > If there are many hours needed then I could see using another ASR in the > mean time, recording their calls and feeding the audio to Pocketsphinx for > training, then switching to Pocketspinx once it's "tuned up." At least this > way a business doesn't have to deal with a "virgin" pocketsphinx. > > > > Mark > > > > > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/267062ec/attachment.html From anthony.minessale at gmail.com Tue Jan 13 09:12:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 13 Jan 2009 11:12:32 -0600 Subject: [Freeswitch-users] Gateways vs Directory Users In-Reply-To: <496CC539.6060900@kinetix.gr> References: <496CC539.6060900@kinetix.gr> Message-ID: <191c3a030901130912l294f83f9h7962f188672c9374@mail.gmail.com> What you are doing is correct. In FS gateways only represent a set of credentials that the profile uses to authenticate to another sip UA The actual UA is the sofia profile and the gateways are assumed identities to deal with challenges. Also to manage everything from the same place, you can put the tag inside the tag in the directory then tell sofia to parse the whole domain for gateways in the profile in section of your sofia profile add this will tell FS to open that domain in the user directory and check every user for and parse them all. On Tue, Jan 13, 2009 at 10:45 AM, Apostolos Pantsiopoulos wrote: > I think one of the most difficult ideas to grasp when someone > migrates to FS from other soft-switches is the definition of > endpoints (in the generic meaning of the word) i.e. other gateways > that your FS gateway will exchange traffic with. From what I > understood from the wiki : > > a user (directory) is an entitiy that defines our subscribers (with > registration or not) > > a gateway is another VoIP gateway that we use to send traffic to > (provider) and sometimes receive traffic from > > (correct me If I am wrong this far) > > Now, I have the following scenario : > > I have an FS box that will exhchange traffic with certain gateways. > Sometimes a gateway will act as a client (customer) or a provider, > depending on the traffic direction. > > 1st approach : > > The first plan I tried, was to use the directory to "define" these > other gateways since they are clients. I turned off registrations > in my SIP profile (because I don't need any), I wrote an xml file > for each gateway in the directory with the appropriate cidr and that > was it. > > I then realized that I could not use my definitions in the users directory > so that I can send a call to that client. In order to do so, I had to > define > that particular IP as a gateway in my SIP profile section. > > 2nd approach : > > I forgot all about the directory thing (after all I needed not any > mailboxes or registrations) and tried to implement the same scenario > by just using gateways. I declared the gateways in my SIP profile and > was able to dial out to them just by using : > > > > which was fine. But when I got an inbound call from that gateway > I realized (1) that I had to define this gateway in an acl and (2) the > dialplan used (context) was always the inbound one since there is no > way of telling a gateway which context to use when accepting a call. > I found a way in the wiki that suggests using DIDs to accomplish that, > but I find it this to be not a very elegant solution because I don't > want to use > prefixes or DIDs to accomplish my goal. I want to be able to define a > gateway using a gateway name (label) of my choice and an IP address, > so when I come to the point of having 100+ gateways i won't > have to search wich DID to use for a new one. > > 3rd approach : > > Use a combination of the above. Declare a gateway that receives and > sends traffic twice. Once as a > user in the directory and once in the gateways section of my profile. > > Does all these sound correct to you? Am I missing something? Which > approach is the best? > Please send me your feedback. Inline comments would be ideal. > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/4b552910/attachment.html From info at freeswitch.es Tue Jan 13 09:14:08 2009 From: info at freeswitch.es (info at freeswitch.es) Date: Tue, 13 Jan 2009 12:14:08 -0500 (COT) Subject: [Freeswitch-users] Nokia 6300i Message-ID: <1231866848.3790@freeswitch.es> Hi, i try to register my nokia 6300i sip client to freeswitch. The sip client connect to freeswitch but don't work well. There is a problem with sip NOTIFY. This is the log: nta: resent NOTIFY (109814666) to udp/190.90.122.161:4301 tport_pend(0x70dc60): pending 0x7b3cd0 for udp/209.20.72.171:5060 (already 3) nta_outgoing_timer: 1/4 resent, 0/4 tout, 0/1 term, 0/5 free nta: timer set next to 1270 ms nta: timer E fired, retransmit NOTIFY (109814655) tport_release(0x70dc60): 0x2aaab0123b20 by 0x2aaab0123ed0 with (nil) tport_tsend(0x70dc60) tpn = udp/190.90.122.161:4301 tport_resolve addrinfo = 190.90.122.161:4301 tport_by_addrinfo(0x70dc60): not found by name udp/190.90.122.161:4301 tport_vsend returned 893 nta: resent NOTIFY (109814655) to udp/190.90.122.161:4301 tport_pend(0x70dc60): pending 0x2aaab0123b20 for udp/209.20.72.171:5060 (already 3) nta_outgoing_timer: 1/4 resent, 0/4 tout, 0/1 term, 0/5 free nta: timer set next to 184 ms nta: timer K fired, terminate NOTIFY (109814654) outgoing_reclaim_all((nil), (nil), 0x422d2e30) nta_outgoing_timer: 0/4 resent, 0/4 tout, 1/1 term, 1/5 free nta: timer set next to 420 ms nta: timer E fired, retransmit NOTIFY (109814665) tport_release(0x70dc60): 0x2aaabc002b40 by 0x2aaabc0102f0 with (nil) tport_tsend(0x70dc60) tpn = udp/190.90.122.161:4301 tport_resolve addrinfo = 190.90.122.161:4301 tport_by_addrinfo(0x70dc60): not found by name udp/190.90.122.161:4301 tport_vsend returned 893 nta: resent NOTIFY (109814665) to udp/190.90.122.161:4301 tport_pend(0x70dc60): pending 0x2aaabc002b40 for udp/209.20.72.171:5060 (already 3) nta_outgoing_timer: 1/4 resent, 0/4 tout, 0/0 term, 0/4 free nta: timer set next to 1810 ms I put this line in 1000.xml (../conf/directory/default)for the NAT purpose On the freeswitch server all ports are opened With Zoiper client all work fine This is the NAT configuration on nokia sip client: Any idea? Thank's From regs at kinetix.gr Tue Jan 13 09:42:34 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 13 Jan 2009 19:42:34 +0200 Subject: [Freeswitch-users] Gateways vs Directory Users In-Reply-To: <191c3a030901130912l294f83f9h7962f188672c9374@mail.gmail.com> References: <496CC539.6060900@kinetix.gr> <191c3a030901130912l294f83f9h7962f188672c9374@mail.gmail.com> Message-ID: <496CD28A.6080100@kinetix.gr> In the http://wiki.freeswitch.org/wiki/Clarification:gateways we can read : "What I gather from this is that if you only want certain extensions to be registered with your voip provider when a specific user registers with freeswitch you should define gateways in the directory section rather than in the sofia configuration. Conversely, if you always want an extension registered with a provider you would define the gateway as part of the sip profile." So If I understand the above statement correctly the gateways that are defined within a user tag are not necessarily the gateways of the specific user (i.e. he/she physically owns them) , but gateways that our FS may use to terminate calls of this specific user. What I also understand from the above statement is that these gateways get registered (or are generally accessible) only when that particular user registers to our FS. Is this correct? Anthony Minessale wrote: > What you are doing is correct. > > In FS gateways only represent a set of credentials that the profile > uses to authenticate to another sip UA > The actual UA is the sofia profile and the gateways are assumed > identities to deal with challenges. > > Also to manage everything from the same place, > you can put the tag inside the tag in the directory then > tell sofia to parse the whole domain for gateways in the profile > > in section of your sofia profile > add > > > this will tell FS to open that domain in the user directory and check > every user > for and parse them all. > > > > > > > On Tue, Jan 13, 2009 at 10:45 AM, Apostolos Pantsiopoulos > > wrote: > > I think one of the most difficult ideas to grasp when someone > migrates to FS from other soft-switches is the definition of > endpoints (in the generic meaning of the word) i.e. other gateways > that your FS gateway will exchange traffic with. From what I > understood from the wiki : > > a user (directory) is an entitiy that defines our subscribers (with > registration or not) > > a gateway is another VoIP gateway that we use to send traffic to > (provider) and sometimes receive traffic from > > (correct me If I am wrong this far) > > Now, I have the following scenario : > > I have an FS box that will exhchange traffic with certain gateways. > Sometimes a gateway will act as a client (customer) or a provider, > depending on the traffic direction. > > 1st approach : > > The first plan I tried, was to use the directory to "define" these > other gateways since they are clients. I turned off registrations > in my SIP profile (because I don't need any), I wrote an xml file > for each gateway in the directory with the appropriate cidr and that > was it. > > I then realized that I could not use my definitions in the users > directory > so that I can send a call to that client. In order to do so, I had to > define > that particular IP as a gateway in my SIP profile section. > > 2nd approach : > > I forgot all about the directory thing (after all I needed not any > mailboxes or registrations) and tried to implement the same scenario > by just using gateways. I declared the gateways in my SIP profile and > was able to dial out to them just by using : > > > > which was fine. But when I got an inbound call from that gateway > I realized (1) that I had to define this gateway in an acl and (2) the > dialplan used (context) was always the inbound one since there is no > way of telling a gateway which context to use when accepting a call. > I found a way in the wiki that suggests using DIDs to accomplish that, > but I find it this to be not a very elegant solution because I don't > want to use > prefixes or DIDs to accomplish my goal. I want to be able to define a > gateway using a gateway name (label) of my choice and an IP address, > so when I come to the point of having 100+ gateways i won't > have to search wich DID to use for a new one. > > 3rd approach : > > Use a combination of the above. Declare a gateway that receives and > sends traffic twice. Once as a > user in the directory and once in the gateways section of my profile. > > Does all these sound correct to you? Am I missing something? Which > approach is the best? > Please send me your feedback. Inline comments would be ideal. > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/6f059e84/attachment-0001.html From anthony.minessale at gmail.com Tue Jan 13 10:02:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 13 Jan 2009 12:02:35 -0600 Subject: [Freeswitch-users] Gateways vs Directory Users In-Reply-To: <496CD28A.6080100@kinetix.gr> References: <496CC539.6060900@kinetix.gr> <191c3a030901130912l294f83f9h7962f188672c9374@mail.gmail.com> <496CD28A.6080100@kinetix.gr> Message-ID: <191c3a030901131002r201a125bo43762faf145884ee@mail.gmail.com> when you tell sofia to parse a domain and it finds gateways in a user the gateways belong to the sofia profile so there is no effective difference, the main reason for gateways in users is for a feature that allows you to register all gateways in a user when you get an inbound reg but there are other benefits in the form of where you manage your gateways etc. On Tue, Jan 13, 2009 at 11:42 AM, Apostolos Pantsiopoulos wrote: > In the http://wiki.freeswitch.org/wiki/Clarification:gateways we can read > : > > "What I gather from this is that if you only want certain extensions to be > registered with your voip provider when a specific user registers with > freeswitch you should define gateways in the directory section rather than > in the sofia configuration. Conversely, if you always want an extension > registered with a provider you would define the gateway as part of the sip > profile." > > So If I understand the above statement correctly the gateways that are > defined within a user tag are not necessarily the gateways of the specific > user (i.e. he/she physically owns them) , > but gateways that our FS may use to terminate calls of this specific user. > > What I also understand from the above statement is that these gateways > get registered (or are generally accessible) only when that particular user > > registers to our FS. Is this correct? > > > Anthony Minessale wrote: > > What you are doing is correct. > > In FS gateways only represent a set of credentials that the profile uses to > authenticate to another sip UA > The actual UA is the sofia profile and the gateways are assumed identities > to deal with challenges. > > Also to manage everything from the same place, > you can put the tag inside the tag in the directory then > tell sofia to parse the whole domain for gateways in the profile > > in section of your sofia profile > add > > > this will tell FS to open that domain in the user directory and check every > user > for and parse them all. > > > > > > > On Tue, Jan 13, 2009 at 10:45 AM, Apostolos Pantsiopoulos > wrote: > >> I think one of the most difficult ideas to grasp when someone >> migrates to FS from other soft-switches is the definition of >> endpoints (in the generic meaning of the word) i.e. other gateways >> that your FS gateway will exchange traffic with. From what I >> understood from the wiki : >> >> a user (directory) is an entitiy that defines our subscribers (with >> registration or not) >> >> a gateway is another VoIP gateway that we use to send traffic to >> (provider) and sometimes receive traffic from >> >> (correct me If I am wrong this far) >> >> Now, I have the following scenario : >> >> I have an FS box that will exhchange traffic with certain gateways. >> Sometimes a gateway will act as a client (customer) or a provider, >> depending on the traffic direction. >> >> 1st approach : >> >> The first plan I tried, was to use the directory to "define" these >> other gateways since they are clients. I turned off registrations >> in my SIP profile (because I don't need any), I wrote an xml file >> for each gateway in the directory with the appropriate cidr and that >> was it. >> >> I then realized that I could not use my definitions in the users directory >> so that I can send a call to that client. In order to do so, I had to >> define >> that particular IP as a gateway in my SIP profile section. >> >> 2nd approach : >> >> I forgot all about the directory thing (after all I needed not any >> mailboxes or registrations) and tried to implement the same scenario >> by just using gateways. I declared the gateways in my SIP profile and >> was able to dial out to them just by using : >> >> >> >> which was fine. But when I got an inbound call from that gateway >> I realized (1) that I had to define this gateway in an acl and (2) the >> dialplan used (context) was always the inbound one since there is no >> way of telling a gateway which context to use when accepting a call. >> I found a way in the wiki that suggests using DIDs to accomplish that, >> but I find it this to be not a very elegant solution because I don't >> want to use >> prefixes or DIDs to accomplish my goal. I want to be able to define a >> gateway using a gateway name (label) of my choice and an IP address, >> so when I come to the point of having 100+ gateways i won't >> have to search wich DID to use for a new one. >> >> 3rd approach : >> >> Use a combination of the above. Declare a gateway that receives and >> sends traffic twice. Once as a >> user in the directory and once in the gateways section of my profile. >> >> Does all these sound correct to you? Am I missing something? Which >> approach is the best? >> Please send me your feedback. Inline comments would be ideal. >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/e4f9843b/attachment.html From mszlazak at aol.com Tue Jan 13 10:08:09 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 13 Jan 2009 13:08:09 -0500 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <191c3a030901130858n763b7662p529f55375e94409d@mail.gmail.com> References: <496B2E4B.7070202@3c.co.uk><3A34A469-1AC3-4226-A42B-2D7F02682ED3@freeswitch.org><8CB42F4C40F8A82-820-5D0@webmail-me14.sysops.aol.com><8CB4321FE9C775B-464-99A@WEBMAIL-DF10.sysops.aol.com><42B494CA-C9E1-49C5-916E-8447EAFBB508@freeswitch.org><8CB436312A08329-80C-1B5D@MBLK-M24.sysops.aol.com> <191c3a030901130858n763b7662p529f55375e94409d@mail.gmail.com> Message-ID: <8CB43C1069AD8FB-16DC-11AB@WEBMAIL-DZ19.sysops.aol.com> Apparently you did. I was responding to comments/claims of another poster by relating my experiences and wishing that PS in FS would preform better. -----Original Message----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Tue, 13 Jan 2009 8:58 am Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl I'm missing something, What is your point exactly? I *just* explained that we want to support unimrcp so you can use Prophecy if you wish so we get it, there is no need to continue to complain.? I tried to tell you that we are short on time and we are trying our best. We had openmrcp and the devloper discontinued the project.? Now we need to get rid of it and switch to unimrcp. If you recall, you called us for consulting, then spent an hour on the phone gathering free information then proceeded to get all kinds of free help on this list using the free software we have made available to you.? It's great that Prophecy is the only place you want to spend any money and I encourage you to do so we can connect you with Voxeo any time.? But what else exactly do you want from us? You may want to factor in that your limited experience and particular requirements contribute to your trouble setting everything up so clearly the pocketsphinx route is not for you.? (You are only the 2nd person to try it on windows for instance) I keep reading all of your emails and I am trying to understand exactly what you want from us. On Tue, Jan 13, 2009 at 12:55 AM, wrote: "My god" I would LOVE it if this is really the case and would praise pocketsphinx (PS) and FS to no end. But my experience has been different. First, I tried the pizza demo with a soft phone and later by outside phone calls to my Linksys 3102 pstn-to-voip gateway. Second, I tried these two set-ups again but with Voxeo's Prophecy ASR. Both are as is and by this I mean there was no training of PocketSphinx just running the pizza demo and with Prophecy there is no training because it can't be trained. Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I couldn't use it at a pizza join. Also, I get a much better experience when calling LumenVox and trying their pizza demo. Now, maybe Prophecy is the type of asr that doesn't require hours of training to make it speaker independent. I know that the Sphinx family are the types of ASR that do need this. So, if there is some settings for adaptation of Pocketsphinx for speaker independence then are they turned on? ? How many hours of calls to a business should an owner expect before PocketSphinx gets good enough not to scare customers away? If there are many hours needed then I could see using another ASR in the mean time, recording their calls and feeding the audio to Pocketsphinx for training, then switching to Pocketspinx once it's "tuned up." At least this way a business doesn't have to deal with a "virgin" pocketsphinx. Mark -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Mon, 12 Jan 2009 3:21 pm Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl Maybe for NON english speakers it doesn't do well but for my tests and needs it does excellent. Sphinx isn't ready thats for sure.. but PocketSphinx does great. I have PocketSphinx doing voice dial by name directory on very common and simple names. If you adapt it it can get much better. But have you called AT&T lately? I have no idea what they use but OMG it sucks... you say "NO" it doesn't understand you.. you say your account number .. it doesn't understand you... you scream curse words at it and it will take you to an agent so they can get you to the right place. Its aweful. Pocketsphinx has performed better than that on my testing. /b On Jan 12, 2009, at 5:09 PM, mszlazak at aol.com wrote: > That's not the opinion of Nickolay S. from the Sphinx forums. He > didn't think it was telephony ready but you implied something > similar in a past email. Also, I got a similar impression with the > pizza demo as it came with FS. Instead I tried Voxeo's Prophecy as > per your recommendation and found it worked better. As I understand > it, pocketsphinx and sphinx (3 & 4) are very good but need adapting > and training for there various uses. > > So, why bother with LumenVox, Voxeo, Nuance, etc if one could get > pocketsphinx working better since it's already integrated with FS? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org A Good Credit Score is 700 or Above. See yours in just 2 easy steps! _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/3ee12f42/attachment-0001.html From mszlazak at aol.com Tue Jan 13 10:31:39 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 13 Jan 2009 13:31:39 -0500 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: References: Message-ID: <8CB43C44ECD2203-C54-50B@webmail-me20.sysops.aol.com> Hi Paul, If you mean fixing up pocketsphinx (ps) for telephony instead of or in addition to working on unimrcp then this is the site of the person who created ps and he may have some advice. http://www.cs.cmu.edu/~dhuggins/ Also, this was a post from the sphinx forums for adapting pocketsphinx for telephony. http://sourceforge.net/forum/message.php?msg_id=5621913 I don't know how accurate it is but if accurate then here is that post to give you some of the issues involved: ----------------- Well, there are issues in both the decoder and the interface with the telephony application. ? First about the decoder, pocketsphinx right now is the most supported and most feature-reach decoder of the family, but in general it's still oriented on the embedded devices. For telephony applications you probably need to extend it a lot. The features that are currently missing are probably: ? * Out-of-box support for multiple recognizers (probably more a freeswitch issue and a model training issue, for example we have no free male/female model). ? ? * Speaker clustering. ? ? * Automatic VTLN estimation from pitch (This looks simple). ? ? * Good endpointer. ? ? * Discriminative training support in SphinxTrain (Huge task). ? * Good and clean support for a garbage model to be able to filter out out of grammar words. ? * Embedded RASTA extraction and RASTA model training. ? * Advanced features extraction ? Another issue is dialog tracking and understanding. CMU folks are doing work on dialog systems, for example Raven is available ? http://www.ravenclaw-olympus.org/systems_overview.html ? It would be worth to look on it and try to integrate it into freepbx. Decoder will need to support combined language model. As well as you'll need a component for postprocessing. The postprocessing includes disfluency removal, text normalization, text boundary detection. Integration with nltk probably useful for sense extraction. ? If you need more details on any of the above, feel free to ask. ------------------- -----Original Message----- From: Paul Herring To: freeswitch-users at lists.freeswitch.org Sent: Tue, 13 Jan 2009 8:18 am Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl What would it take to put a budget together to for this project? Date: Tue, 13 Jan 2009 01:55:36 -0500 From: mszlazak at aol.com Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl To: freeswitch-users at lists.freeswitch.org Message-ID: <8CB436312A08329-80C-1B5D at MBLK-M24.sysops.aol.com> Content-Type: text/plain; charset="us-ascii" "My god" I would LOVE it if this is really the case and would praise pocketsphinx (PS) and FS to no end. But my experience has been different. First, I tried the pizza demo with a soft phone and later by outside phone calls to my Linksys 3102 pstn-to-voip gateway. Second, I tried these two set-ups again but with Voxeo's Prophecy ASR. Both are as is and by this I mean there was no training of PocketSphinx just running the pizza demo and with Prophecy there is no training because it can't be trained. Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I couldn't use it at a pizza join. Also, I get a much better experience when calling LumenVox and trying their pizza demo. Now, maybe Prophecy is the type of asr that doesn't require hours of training to make it speaker independent. I know that the Sphinx family are the types of ASR that do need this. So, if there is some settings for adaptation of Pocketsphinx for speaker independence then are they turned on? ? How many hours of calls to a business should an owner expect before PocketSphinx gets good enough not to scare customers away? If there are many hours needed then I could see using another ASR in the mean time, recording their calls and feeding the audio to Pocketsphinx for training, then switching to Pocketspinx once it's "tuned up." At least this way a business doesn't have to deal with a "virgin" pocketsphinx. Mark -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/b8487ad9/attachment.html From brian at freeswitch.org Tue Jan 13 10:40:39 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 13 Jan 2009 12:40:39 -0600 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <8CB43C44ECD2203-C54-50B@webmail-me20.sysops.aol.com> References: <8CB43C44ECD2203-C54-50B@webmail-me20.sysops.aol.com> Message-ID: Well I also want to have someone take pocketsphinx and flite and build an opensource speech server and maybe gain some momentum to improve it. btw pocketsphinx supports jsgf, I need to update mod_pocketsphinx to do that but I want to work with DHD to figure out how to load the dictionary once... and just load grammar files moving forward. /b On Jan 13, 2009, at 12:31 PM, mszlazak at aol.com wrote: > Hi Paul, > > If you mean fixing up pocketsphinx (ps) for telephony instead of or > in addition to working on unimrcp then this is the site of the > person who created ps and he may have some advice. > > http://www.cs.cmu.edu/~dhuggins/ > > Also, this was a post from the sphinx forums for adapting > pocketsphinx for telephony. > > http://sourceforge.net/forum/message.php?msg_id=5621913 > > I don't know how accurate it is but if accurate then here is that > post to give you some of the issues involved: > > ----------------- > Well, there are issues in both the decoder and the interface with the > telephony application. > > First about the decoder, pocketsphinx right now is the most supported > and most feature-reach decoder of the family, but in general it's > still > oriented on the embedded devices. For telephony applications you > probably need to extend it a lot. The features that are currently > missing are probably: > > * Out-of-box support for multiple recognizers (probably more a > freeswitch > issue and a model training issue, for example we have no free > male/female model). > > * Speaker clustering. > > * Automatic VTLN estimation from pitch (This looks simple). > > * Good endpointer. > > * Discriminative training support in SphinxTrain (Huge task). > > * Good and clean support for a garbage model to be able to filter out > out of grammar words. > > * Embedded RASTA extraction and RASTA model training. > > * Advanced features extraction > > Another issue is dialog tracking and understanding. CMU folks are > doing > work on dialog systems, for example Raven is available > > http://www.ravenclaw-olympus.org/systems_overview.html > > It would be worth to look on it and try to integrate it into > freepbx. Decoder will need to support combined language model. As well > as you'll need a component for postprocessing. The postprocessing > includes > disfluency removal, text normalization, text boundary detection. > Integration > with nltk probably useful for sense extraction. > > If you need more details on any of the above, feel free to ask. > ------------------- > > > > > > -----Original Message----- > From: Paul Herring > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 13 Jan 2009 8:18 am > Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl > > What would it take to put a budget together to for this project? > > > > > > Date: Tue, 13 Jan 2009 01:55:36 -0500 > > From: mszlazak at aol.com > > Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <8CB436312A08329-80C-1B5D at MBLK-M24.sysops.aol.com> > > Content-Type: text/plain; charset="us-ascii" > > > > > > "My god" I would LOVE it if this is really the case and would praise > > pocketsphinx (PS) and FS to no end. But my experience has been > different. > > > > First, I tried the pizza demo with a soft phone and later by outside > phone > > calls to my Linksys 3102 pstn-to-voip gateway. > > Second, I tried these two set-ups again but with Voxeo's Prophecy ASR. > > > > Both are as is and by this I mean there was no training of > PocketSphinx just > > running the pizza demo and with Prophecy there is no training > because it > > can't be trained. > > > > Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I > > couldn't use it at a pizza join. Also, I get a much better > experience when > > calling LumenVox and trying their pizza demo. > > > > Now, maybe Prophecy is the type of asr that doesn't require hours of > > training to make it speaker independent. I know that the Sphinx > family are > > the types of ASR that do need this. > > > > So, if there is some settings for adaptation of Pocketsphinx for > speaker > > independence then are they turned on? > > ? > > How many hours of calls to a business should an owner expect before > > PocketSphinx gets good enough not to scare customers away? > > > > If there are many hours needed then I could see using another ASR in > the > > mean time, recording their calls and feeding the audio to > Pocketsphinx for > > training, then switching to Pocketspinx once it's "tuned up." At > least this > > way a business doesn't have to deal with a "virgin" pocketsphinx. > > > > > > > > Mark > > > > > > > > > > > > -- > > This message has been scanned for viruses and > > dangerous content by MailScanner, and is > > believed to be clean. > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > A Good Credit Score is 700 or Above. See yours in just 2 easy steps! > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/cf6884b9/attachment.html From anthony.minessale at gmail.com Tue Jan 13 10:59:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 13 Jan 2009 12:59:40 -0600 Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl In-Reply-To: <8CB43C44ECD2203-C54-50B@webmail-me20.sysops.aol.com> References: <8CB43C44ECD2203-C54-50B@webmail-me20.sysops.aol.com> Message-ID: <191c3a030901131059ra3fdcdcs41a24a470bef52e9@mail.gmail.com> We already have been collaborating with the maintainer of pocketsphinx for more than a year now, that's why we have a mod_pocketsphinx. His latest releases are based on our feedback so we are already doing that. On Tue, Jan 13, 2009 at 12:31 PM, wrote: > Hi Paul, > > If you mean fixing up pocketsphinx (ps) for telephony instead of or in > addition to working on unimrcp then this is the site of the person who > created ps and he may have some advice. > > http://www.cs.cmu.edu/~dhuggins/ > > Also, this was a post from the sphinx forums for adapting pocketsphinx for > telephony. > > http://sourceforge.net/forum/message.php?msg_id=5621913 > > I don't know how accurate it is but if accurate then here is that post to > give you some of the issues involved: > > ----------------- > Well, there are issues in both the decoder and the interface with the > telephony application. > > First about the decoder, pocketsphinx right now is the most supported > and most feature-reach decoder of the family, but in general it's still > oriented on the embedded devices. For telephony applications you > probably need to extend it a lot. The features that are currently > missing are probably: > > * Out-of-box support for multiple recognizers (probably more a freeswitch > issue and a model training issue, for example we have no free > male/female model). > > * Speaker clustering. > > * Automatic VTLN estimation from pitch (This looks simple). > > * Good endpointer. > > * Discriminative training support in SphinxTrain (Huge task). > > * Good and clean support for a garbage model to be able to filter out > out of grammar words. > > * Embedded RASTA extraction and RASTA model training. > > * Advanced features extraction > > Another issue is dialog tracking and understanding. CMU folks are doing > work on dialog systems, for example Raven is available > > http://www.ravenclaw-olympus.org/systems_overview.html > > It would be worth to look on it and try to integrate it into > freepbx. Decoder will need to support combined language model. As well > as you'll need a component for postprocessing. The postprocessing includes > disfluency removal, text normalization, text boundary detection. > Integration > with nltk probably useful for sense extraction. > > If you need more details on any of the above, feel free to ask. > ------------------- > > > > > > -----Original Message----- > From: Paul Herring > To: freeswitch-users at lists.freeswitch.org > Sent: Tue, 13 Jan 2009 8:18 am > Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl > > What would it take to put a budget together to for this project? > > > > Date: Tue, 13 Jan 2009 01:55:36 -0500 > > From: mszlazak at aol.com > > Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <8CB436312A08329-80C-1B5D at MBLK-M24.sysops.aol.com> > > Content-Type: text/plain; charset="us-ascii" > > > > "My god" I would LOVE it if this is really the case and would praise > > pocketsphinx (PS) and FS to no end. But my experience has been different. > > > First, I tried the pizza demo with a soft phone and later by outside phone > > calls to my Linksys 3102 pstn-to-voip gateway. > > Second, I tried these two set-ups again but with Voxeo's Prophecy ASR. > > > Both are as is and by this I mean there was no training of PocketSphinx just > > running the pizza demo and with Prophecy there is no training because it > > can't be trained. > > > Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I > > couldn't use it at a pizza join. Also, I get a much better experience when > > calling LumenVox and trying their pizza demo. > > > Now, maybe Prophecy is the type of asr that doesn't require hours of > > training to make it speaker independent. I know that the Sphinx family are > > the types of ASR that do need this. > > > So, if there is some settings for adaptation of Pocketsphinx for speaker > > independence then are they turned on? > > ? > > How many hours of calls to a business should an owner expect before > > PocketSphinx gets good enough not to scare customers away? > > > If there are many hours needed then I could see using another ASR in the > > mean time, recording their calls and feeding the audio to Pocketsphinx for > > training, then switching to Pocketspinx once it's "tuned up." At least this > > way a business doesn't have to deal with a "virgin" pocketsphinx. > > > > > Mark > > > > > > > > -- > > This message has been scanned for viruses and > > dangerous content by MailScanner, and is > > believed to be clean. > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > *A Good Credit Score is 700 or Above. See yours in just 2 easy steps! > * > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090113/c99e169e/attachment-0001.html From krice at suspicious.org Tue Jan 13 18:24:22 2009 From: krice at suspicious.org (Ken Rice) Date: Tue, 13 Jan 2009 20:24:22 -0600 Subject: [Freeswitch-users] Announcing mod_easyroute Message-ID: Greetings! I would like to announce the inclusion of mod_easyroute in FreeSwitch. mod_easyroute is a DID routing engine that uses an ODBC data source to figure out where to route calls. It uses a gateway concept and a number concept with a relation of one gateway to many numbers. Numbers are looked up in the database and a dial string and other useful information is returned into channel variables including a full dial string that can be passed to the bridge function for sending the call out. This module is not really targeted at the PBX user but at the ITSP community as an example of what can be done in C using the ODBC extensions built into the FreeSwitch core. See http://wiki.freeswitch.org/wiki/Mod_easyroute or http://www.freeswitch.org/node/158 for more information. Please note: if you are wanting to play with mod_easyroute, please ensure you have FreeSwitch compiled with ODBC support enabled (see http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core for ODBC setup information) If you are wondering why I would release such a thing, I have had this code sitting around for a bit now, a friend of mine needed the ability to route DIDs to different SIP based Application Servers for his business, so I wrote this for him, over the course of the past several months several people have approached me needing something similar and I re-used this code a few times for various consulting projects. A few weeks ago one of the guys in the IRC channel mentions he was about to write something like this for one of his clients, I mentioned I had this and he offered to "bounty" this functionality if I would release it. So here it is guys have at it. Thanks to the guys that sent me a few bucks for this code! Now to get a quick plug in for myself. I'm a consultant that works mainly with ITSPs and VoIP related projects, If you find yourself in need of some professional services to assist you in customizing or deploying FreeSwitch for your business, drop me an email off list and I will get back to you. With the growth that's happened with FreeSwitch up til now, I can't wait to see what 2009 will bring! Ken Rice Krice rmktek com Ken4VoIP AIM From william.suffill at gmail.com Tue Jan 13 18:52:11 2009 From: william.suffill at gmail.com (William Suffill) Date: Tue, 13 Jan 2009 21:52:11 -0500 Subject: [Freeswitch-users] Announcing mod_easyroute In-Reply-To: References: Message-ID: <6b65470d0901131852y2408735cl7ce6cbaf13f730f2@mail.gmail.com> Looks like a solid contribution as always Ken. I agree it should be an interesting year with the way things are shaping up. Glad to see someone pointing out how to get ODBC to work. =) Where was all this info a few hours ago? Oh well one way to learn I guess. Thanks again Ken. -- William From scott.ellis at novatex.com.au Wed Jan 14 02:03:22 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Wed, 14 Jan 2009 21:03:22 +1100 Subject: [Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call Message-ID: <496DB86A.2000708@novatex.com.au> I have an inbound call via OpenZap, when I attempt to bridge to a SIP extension, I get the ring tone (inbound line) up until the bridge fails (for timeout or do not disturb). At this point the call is answered and then my dial plan moves on to attempt another bridge to different extensions. So I no longer have the ring tone for the person dialing in. The call can still be answered and everything works ok, but I would rather not answer the call until someone actually picks up. Failing that simulating a ring tone would be good enough. Have searched around, but at a bit of a loss on how to dothis. Any suggestions greatly appreciated. Scott From my dialplan From avmanansala at gmail.com Wed Jan 14 03:14:32 2009 From: avmanansala at gmail.com (Lito Manansala) Date: Wed, 14 Jan 2009 19:14:32 +0800 Subject: [Freeswitch-users] Announcing mod_easyroute In-Reply-To: <6b65470d0901131852y2408735cl7ce6cbaf13f730f2@mail.gmail.com> References: <6b65470d0901131852y2408735cl7ce6cbaf13f730f2@mail.gmail.com> Message-ID: <8e0176e60901140314u7d6bda7dlc3fda4a484f4e2f0@mail.gmail.com> this is a killer most likely for sw billing developers. another breakthrough... On Wed, Jan 14, 2009 at 10:52 AM, William Suffill wrote: > Looks like a solid contribution as always Ken. I agree it should be an > interesting year with the way things are shaping up. > Glad to see someone pointing out how to get ODBC to work. =) > > Where was all this info a few hours ago? Oh well one way to learn I > guess. Thanks again Ken. > > -- William > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- /Lito -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/b6e63ae7/attachment.html From scott.ellis at novatex.com.au Wed Jan 14 03:53:40 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Wed, 14 Jan 2009 22:53:40 +1100 Subject: [Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call (partial call log added) In-Reply-To: <496DB86A.2000708@novatex.com.au> References: <496DB86A.2000708@novatex.com.au> Message-ID: <496DD244.9070903@novatex.com.au> > I have an inbound call via OpenZap, when I attempt to bridge to a SIP > extension, I get the ring tone (inbound line) up until the bridge fails > (for timeout or do not disturb). At this point the call is answered and > then my dial plan moves on to attempt another bridge to different > extensions. So I no longer have the ring tone for the person dialing in. > The call can still be answered and everything works ok, but I would > rather not answer the call until someone actually picks up. Failing that > simulating a ring tone would be good enough. > > Have searched around, but at a bit of a loss on how to dothis. > > Any suggestions greatly appreciated. > > Scott > > From my dialplan > > > > > > > > > > data="{leg_timeout=30}sofia/$${domain}/500"/> > > > > > > > > > data="${group_call(ringgroup2@${domain_name})"/> > > > > > data="${group_call(everyone@${domain_name})"/> > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > A call log 009-01-14 22:47:10 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][2:1] STATE [IDLE] 2009-01-14 22:47:12 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][2:1] STATE [IDLE] 2009-01-14 22:47:13 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][2:1] STATE [IDLE] 2009-01-14 22:47:13 [NOTICE] switch_ivr_originate.c:206 check_per_channel_timeouts() Hangup sofia/internal/500 [CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT] 2009-01-14 22:47:13 [DEBUG] switch_channel.c:1517 switch_channel_perform_hangup() Send signal sofia/internal/500 [KILL] 2009-01-14 22:47:13 [DEBUG] switch_core_session.c:810 switch_core_session_signal_state_change() Send signal sofia/internal/500 [BREAK] 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (sofia/internal/500) State CONSUME_MEDIA going to sleep 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/500) Running State Change CS_HANGUP 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/500) State HANGUP 2009-01-14 22:47:13 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/500 hanging up, cause: ALLOTTED_TIMEOUT 2009-01-14 22:47:13 [DEBUG] mod_sofia.c:351 sofia_on_hangup() Sending CANCEL to sofia/internal/500 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/500 Standard HANGUP, cause: ALLOTTED_TIMEOUT 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/500) State HANGUP going to sleep 2009-01-14 22:47:13 [DEBUG] switch_core_session.c:942 switch_core_session_thread() Session 403 (sofia/internal/500) Locked, Waiting on external entities 2009-01-14 22:47:13 [DEBUG] switch_ivr_originate.c:1705 switch_ivr_originate() Originate Resulted in Error Cause: 602 [ALLOTTED_TIMEOUT] 2009-01-14 22:47:13 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 403 (sofia/internal/500) Ended 2009-01-14 22:47:13 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/500 [CS_HANGUP] 2009-01-14 22:47:13 [DEBUG] switch_ivr.c:59 switch_ivr_sleep() OpenZAP/2:1/2 receive message [PROGRESS] 2009-01-14 22:47:13 [DEBUG] mod_openzap.c:785 channel_receive_message_fxo() Changing state on 2:1 from IDLE to UP 2009-01-14 22:47:13 [DEBUG] switch_core_session.c:513 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/2 [BREAK] 2009-01-14 22:47:13 [NOTICE] switch_ivr.c:59 switch_ivr_sleep() Ring-Ready OpenZAP/2:1/2! 2009-01-14 22:47:13 [NOTICE] switch_ivr.c:59 switch_ivr_sleep() Pre-Answer OpenZAP/2:1/2! 2009-01-14 22:47:13 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/2:1/2 receive message [AUDIO_SYNC] 2009-01-14 22:47:13 [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: ALLOTTED_TIMEOUT 2009-01-14 22:47:13 [DEBUG] mod_dptools.c:1930 audio_bridge_function() Continue on fail [true]: Cause: ALLOTTED_TIMEOUT 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/2:1/2 Execute set(call_timeout=15) 2009-01-14 22:47:13 [DEBUG] mod_dptools.c:699 set_function() OpenZAP/2:1/2 SET [call_timeout]=[15] 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/2:1/2 Execute set(continue_on_fail=true) 2009-01-14 22:47:13 [DEBUG] mod_dptools.c:699 set_function() OpenZAP/2:1/2 SET [continue_on_fail]=[true] 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/2:1/2 Execute set(hangup_after_bridge=true) 2009-01-14 22:47:13 [DEBUG] mod_dptools.c:699 set_function() OpenZAP/2:1/2 SET [hangup_after_bridge]=[true] 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/2:1/2 Execute ring_ready() 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/2:1/2 Execute bridge({leg_timeout=10}sofia/10.0.0.9/520) 2009-01-14 22:47:13 [DEBUG] switch_ivr_originate.c:783 switch_ivr_originate() variable string 0 = [leg_timeout=10] 2009-01-14 22:47:13 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/520 [15657cd4-e231-11dd-9df1-2bc8b5f74e14] 2009-01-14 22:47:13 [DEBUG] mod_sofia.c:2518 sofia_outgoing_channel() (sofia/internal/520) State Change CS_NEW -> CS_INIT 2009-01-14 22:47:13 [DEBUG] switch_core_session.c:810 switch_core_session_signal_state_change() Send signal sofia/internal/520 [BREAK] 2009-01-14 22:47:13 [DEBUG] switch_ivr_originate.c:1179 switch_ivr_originate() sofia/internal/520 Setting leg timeout to 10 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/520) Running State Change CS_INIT 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/internal/520) State INIT 2009-01-14 22:47:13 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/520 SOFIA INIT 2009-01-14 22:47:13 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/520) State Change CS_INIT -> CS_ROUTING 2009-01-14 22:47:13 [DEBUG] switch_core_session.c:810 switch_core_session_signal_state_change() Send signal sofia/internal/520 [BREAK] 2009-01-14 22:47:13 [DEBUG] sofia.c:2573 sofia_handle_sip_i_state() Channel sofia/internal/520 entering state [calling] 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/internal/520) State INIT going to sleep 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/520) Running State Change CS_ROUTING 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/internal/520) State ROUTING 2009-01-14 22:47:13 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/520 SOFIA ROUTING 2009-01-14 22:47:13 [DEBUG] switch_ivr_originate.c:62 originate_on_routing() (sofia/internal/520) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-01-14 22:47:13 [DEBUG] switch_core_session.c:810 switch_core_session_signal_state_change() Send signal sofia/internal/520 [BREAK] 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/internal/520) State ROUTING going to sleep 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/520) Running State Change CS_CONSUME_MEDIA 2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (sofia/internal/520) State CONSUME_MEDIA 2009-01-14 22:47:13 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][2:1] STATE [UP] 2009-01-14 22:47:13 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for UP 2009-01-14 22:47:13 [DEBUG] mod_openzap.c:1165 on_fxo_signal() got FXO sig 2:1 [UP] 2009-01-14 22:47:13 [NOTICE] mod_openzap.c:1192 on_fxo_signal() Channel [OpenZAP/2:1/2] has been answered From anthony.minessale at gmail.com Wed Jan 14 05:39:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 14 Jan 2009 07:39:20 -0600 Subject: [Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call In-Reply-To: <496DB86A.2000708@novatex.com.au> References: <496DB86A.2000708@novatex.com.au> Message-ID: <191c3a030901140539y4f8ffa20nefdc23209c1b0df7@mail.gmail.com> Have a look here: http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones On Wed, Jan 14, 2009 at 4:03 AM, Scott Ellis wrote: > I have an inbound call via OpenZap, when I attempt to bridge to a SIP > extension, I get the ring tone (inbound line) up until the bridge fails > (for timeout or do not disturb). At this point the call is answered and > then my dial plan moves on to attempt another bridge to different > extensions. So I no longer have the ring tone for the person dialing in. > The call can still be answered and everything works ok, but I would > rather not answer the call until someone actually picks up. Failing that > simulating a ring tone would be good enough. > > Have searched around, but at a bit of a loss on how to dothis. > > Any suggestions greatly appreciated. > > Scott > > From my dialplan > > > > > > > > > > data="{leg_timeout=30}sofia/$${domain}/500"/> > > > > > > > > > data="${group_call(ringgroup2@${domain_name})"/> > > > > > data="${group_call(everyone@${domain_name})"/> > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/eb51af64/attachment-0001.html From tomasborrella at gmail.com Wed Jan 14 03:54:02 2009 From: tomasborrella at gmail.com (=?ISO-8859-1?Q?Tom=E1s?=) Date: Wed, 14 Jan 2009 12:54:02 +0100 Subject: [Freeswitch-users] No caller/called ID received (Wildcard X101P) Message-ID: <46336fde0901140354q5a9ba05cx505ead321de165cc@mail.gmail.com> Hi all, I'm a new FreeSwitch user and this is my first email to the list. I'm trying to configure my Home PBX with a Wildcard X101P (configured as FXO) and I have a problem receiving the caller/called ID from PSTN. This is the content of file "openzap.conf": [span zt] name => OpenZAP number => 1 fxo-channel => 1 And this is the content of file "openzap.conf.xml": As you can see the param "enable-callerid" is set to "true", but when I received and incoming call, FreeSwitch doesn't get neither the caller number nor the called number (instead of my home number, I receive a number 1, as can be seen on the following log of an incoming call): 2009-01-14 20:43:12 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/1 [941d6234-e273-11dd-bcdf-89190a30fe61] 2009-01-14 20:43:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP->1 in context default 2009-01-14 20:43:12 [NOTICE] switch_ivr.c:1255 switch_ivr_session_transfer() Transfer OpenZAP/1:1/1 to enum[1 at default] 2009-01-14 20:43:16 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting 2009-01-14 20:43:16 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup OpenZAP/1:1/1 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 1 (OpenZAP/1:1/1 ) Ended 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/1 [CS_HANGUP] 2009-01-14 20:43:23 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/1 [9af53280-e273-11dd-bcdf-89190a30fe61] Any idea of what's the problem? Thank you very much in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/61ebfbb7/attachment.html From anthony.minessale at gmail.com Wed Jan 14 06:05:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 14 Jan 2009 08:05:06 -0600 Subject: [Freeswitch-users] No caller/called ID received (Wildcard X101P) In-Reply-To: <46336fde0901140354q5a9ba05cx505ead321de165cc@mail.gmail.com> References: <46336fde0901140354q5a9ba05cx505ead321de165cc@mail.gmail.com> Message-ID: <191c3a030901140605g54b3b669r4091e4ebb2723005@mail.gmail.com> number => 1 This value should be set to the DID of the FXO line. That way when a call hits FS it will go to that extension in the dialplan. This is unrelated to callerid, it's the destination not the source. If the line has caller-id it will also be available when it's collected after the 2nd ring. On Wed, Jan 14, 2009 at 5:54 AM, Tom?s wrote: > Hi all, > > I'm a new FreeSwitch user and this is my first email to the list. > > I'm trying to configure my Home PBX with a Wildcard X101P (configured as > FXO) and I have a problem receiving the caller/called ID from PSTN. > > This is the content of file "openzap.conf": > > [span zt] > name => OpenZAP > number => 1 > fxo-channel => 1 > > And this is the content of file "openzap.conf.xml": > > > > > > > > > > > > > > > > > > As you can see the param "enable-callerid" is set to "true", but when I > received and incoming call, FreeSwitch doesn't get neither the caller number > nor the called number (instead of my home number, I receive a number 1, as > can be seen on the following log of an incoming call): > > 2009-01-14 20:43:12 [NOTICE] switch_channel.c:565 switch_channel_set_name() > New Channel OpenZAP/1:1/1 [941d6234-e273-11dd-bcdf-89190a30fe61] > 2009-01-14 20:43:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing OpenZAP->1 in context default > 2009-01-14 20:43:12 [NOTICE] switch_ivr.c:1255 > switch_ivr_session_transfer() Transfer OpenZAP/1:1/1 to enum[1 at default] > 2009-01-14 20:43:16 [INFO] switch_core_state_machine.c:122 > switch_core_standard_on_routing() No Route, Aborting > 2009-01-14 20:43:16 [NOTICE] switch_core_state_machine.c:123 > switch_core_standard_on_routing() Hangup OpenZAP/1:1/1 [CS_ROUTING] > [NO_ROUTE_DESTINATION] > 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 1 (OpenZAP/1:1/1 ) Ended > 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel OpenZAP/1:1/1 [CS_HANGUP] > 2009-01-14 20:43:23 [NOTICE] switch_channel.c:565 switch_channel_set_name() > New Channel OpenZAP/1:1/1 [9af53280-e273-11dd-bcdf-89190a30fe61] > > Any idea of what's the problem? > > Thank you very much in advance. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/fc915dab/attachment.html From alex at sinapticode.ro Wed Jan 14 06:18:31 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Wed, 14 Jan 2009 16:18:31 +0200 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE problem Message-ID: <1231942711.4911.7.camel@gathern.lan> Hi, I want to know exactly what does this hangup_cause means: "NORMAL_TEMPORARY_FAILURE". I'm receiving lots of those. Is the SIP provider to blame, or is my setup? I took a look at the sip communication (Wireshark/tcpdump), and I couldn't find a response from the sip provider that matches. All I got was "487 Request Cancelled" and "486 Busy Here". Thanks, From mike at jerris.com Wed Jan 14 06:18:38 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 14 Jan 2009 09:18:38 -0500 Subject: [Freeswitch-users] No caller/called ID received (Wildcard X101P) In-Reply-To: <191c3a030901140605g54b3b669r4091e4ebb2723005@mail.gmail.com> References: <46336fde0901140354q5a9ba05cx505ead321de165cc@mail.gmail.com> <191c3a030901140605g54b3b669r4091e4ebb2723005@mail.gmail.com> Message-ID: <82719D59-5275-4FEC-B655-9A4CFB724D69@jerris.com> I noticed tonegroup=es. What country are you in and do you know what method they use to do dtmf. Most likely we need a small tweak to set the dtmf method for your country. Mike On Jan 14, 2009, at 9:05 AM, Anthony Minessale wrote: > > number => 1 > > This value should be set to the DID of the FXO line. > That way when a call hits FS it will go to that extension in the > dialplan. > This is unrelated to callerid, it's the destination not the source. > > If the line has caller-id it will also be available when it's > collected after the 2nd ring. > > > > On Wed, Jan 14, 2009 at 5:54 AM, Tom?s > wrote: > Hi all, > > I'm a new FreeSwitch user and this is my first email to the list. > > I'm trying to configure my Home PBX with a Wildcard X101P > (configured as FXO) and I have a problem receiving the caller/called > ID from PSTN. > > This is the content of file "openzap.conf": > > [span zt] > name => OpenZAP > number => 1 > fxo-channel => 1 > > And this is the content of file "openzap.conf.xml": > > > > > > > > > > > > > > > > > > As you can see the param "enable-callerid" is set to "true", but > when I received and incoming call, FreeSwitch doesn't get neither > the caller number nor the called number (instead of my home number, > I receive a number 1, as can be seen on the following log of an > incoming call): > > 2009-01-14 20:43:12 [NOTICE] switch_channel.c:565 > switch_channel_set_name() New Channel OpenZAP/1:1/1 [941d6234- > e273-11dd-bcdf-89190a30fe61] > 2009-01-14 20:43:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing OpenZAP->1 in context default > 2009-01-14 20:43:12 [NOTICE] switch_ivr.c:1255 > switch_ivr_session_transfer() Transfer OpenZAP/1:1/1 to > enum[1 at default] > 2009-01-14 20:43:16 [INFO] switch_core_state_machine.c:122 > switch_core_standard_on_routing() No Route, Aborting > 2009-01-14 20:43:16 [NOTICE] switch_core_state_machine.c:123 > switch_core_standard_on_routing() Hangup OpenZAP/1:1/1 [CS_ROUTING] > [NO_ROUTE_DESTINATION] > 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 1 (OpenZAP/1:1/1 ) Ended > 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel OpenZAP/1:1/1 [CS_HANGUP] > 2009-01-14 20:43:23 [NOTICE] switch_channel.c:565 > switch_channel_set_name() New Channel OpenZAP/1:1/1 [9af53280- > e273-11dd-bcdf-89190a30fe61] > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/e2eea3fc/attachment.html From mike at jerris.com Wed Jan 14 06:23:54 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 14 Jan 2009 09:23:54 -0500 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE problem In-Reply-To: <1231942711.4911.7.camel@gathern.lan> References: <1231942711.4911.7.camel@gathern.lan> Message-ID: Sip cause code to Q.850 cause code translations can be found in RFC4497 section 8.4.4. FreeSWITCH uses Q.850 codes internally so you will typically see those in the logs. We do pass the sip cause codes across a sip to sip bridge. Mike On Jan 14, 2009, at 9:18 AM, Alexandru Nedelcu wrote: > Hi, > > I want to know exactly what does this hangup_cause means: > "NORMAL_TEMPORARY_FAILURE". I'm receiving lots of those. > > Is the SIP provider to blame, or is my setup? > > I took a look at the sip communication (Wireshark/tcpdump), and I > couldn't find a response from the sip provider that matches. All I got > was "487 Request Cancelled" and "486 Busy Here". > > Thanks, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/86c852b0/attachment-0001.html From tomasborrella at gmail.com Wed Jan 14 06:38:35 2009 From: tomasborrella at gmail.com (=?ISO-8859-1?Q?Tom=E1s?=) Date: Wed, 14 Jan 2009 15:38:35 +0100 Subject: [Freeswitch-users] No caller/called ID received (Wildcard X101P) In-Reply-To: <82719D59-5275-4FEC-B655-9A4CFB724D69@jerris.com> References: <46336fde0901140354q5a9ba05cx505ead321de165cc@mail.gmail.com> <191c3a030901140605g54b3b669r4091e4ebb2723005@mail.gmail.com> <82719D59-5275-4FEC-B655-9A4CFB724D69@jerris.com> Message-ID: <46336fde0901140638l5f960600ra0b26e05f52b85e8@mail.gmail.com> Hi, Anthony, I think that's my problem, when I receive a call from the PSTN, FS receive number 1 instead of my house number and I don't know why. Michael, I live in Spain, Is it not "es" the tonegroup I should use? Thank you very much. On Wed, Jan 14, 2009 at 3:18 PM, Michael Jerris wrote: > I noticed tonegroup=es. What country are you in and do you know what > method they use to do dtmf. Most likely we need a small tweak to set the > dtmf method for your country. > Mike > > On Jan 14, 2009, at 9:05 AM, Anthony Minessale wrote: > > > number => 1 > > This value should be set to the DID of the FXO line. > That way when a call hits FS it will go to that extension in the dialplan. > This is unrelated to callerid, it's the destination not the source. > > If the line has caller-id it will also be available when it's collected > after the 2nd ring. > > > > On Wed, Jan 14, 2009 at 5:54 AM, Tom?s wrote: > >> Hi all, >> >> I'm a new FreeSwitch user and this is my first email to the list. >> >> I'm trying to configure my Home PBX with a Wildcard X101P (configured as >> FXO) and I have a problem receiving the caller/called ID from PSTN. >> >> This is the content of file "openzap.conf": >> >> [span zt] >> name => OpenZAP >> number => 1 >> fxo-channel => 1 >> >> And this is the content of file "openzap.conf.xml": >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> As you can see the param "enable-callerid" is set to "true", but when I >> received and incoming call, FreeSwitch doesn't get neither the caller number >> nor the called number (instead of my home number, I receive a number 1, as >> can be seen on the following log of an incoming call): >> >> 2009-01-14 20:43:12 [NOTICE] switch_channel.c:565 >> switch_channel_set_name() New Channel OpenZAP/1:1/1 >> [941d6234-e273-11dd-bcdf-89190a30fe61] >> 2009-01-14 20:43:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing OpenZAP->1 in context default >> 2009-01-14 20:43:12 [NOTICE] switch_ivr.c:1255 >> switch_ivr_session_transfer() Transfer OpenZAP/1:1/1 to enum[1 at default] >> 2009-01-14 20:43:16 [INFO] switch_core_state_machine.c:122 >> switch_core_standard_on_routing() No Route, Aborting >> 2009-01-14 20:43:16 [NOTICE] switch_core_state_machine.c:123 >> switch_core_standard_on_routing() Hangup OpenZAP/1:1/1 [CS_ROUTING] >> [NO_ROUTE_DESTINATION] >> 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:960 >> switch_core_session_thread() Session 1 (OpenZAP/1:1/1 ) Ended >> 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:962 >> switch_core_session_thread() Close Channel OpenZAP/1:1/1 [CS_HANGUP] >> 2009-01-14 20:43:23 [NOTICE] switch_channel.c:565 >> switch_channel_set_name() New Channel OpenZAP/1:1/1 >> [9af53280-e273-11dd-bcdf-89190a30fe61] >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/1c12276b/attachment.html From edpimentl at gmail.com Wed Jan 14 06:44:03 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 14 Jan 2009 09:44:03 -0500 Subject: [Freeswitch-users] Announcing mod_easyroute In-Reply-To: <8e0176e60901140314u7d6bda7dlc3fda4a484f4e2f0@mail.gmail.com> References: <6b65470d0901131852y2408735cl7ce6cbaf13f730f2@mail.gmail.com> <8e0176e60901140314u7d6bda7dlc3fda4a484f4e2f0@mail.gmail.com> Message-ID: <9dc4a1670901140644h2eb56ffcla06929e348508da1@mail.gmail.com> Excellent E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/a36d3710/attachment.html From anthony.minessale at gmail.com Wed Jan 14 07:10:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 14 Jan 2009 09:10:02 -0600 Subject: [Freeswitch-users] No caller/called ID received (Wildcard X101P) In-Reply-To: <46336fde0901140638l5f960600ra0b26e05f52b85e8@mail.gmail.com> References: <46336fde0901140354q5a9ba05cx505ead321de165cc@mail.gmail.com> <191c3a030901140605g54b3b669r4091e4ebb2723005@mail.gmail.com> <82719D59-5275-4FEC-B655-9A4CFB724D69@jerris.com> <46336fde0901140638l5f960600ra0b26e05f52b85e8@mail.gmail.com> Message-ID: <191c3a030901140710q7d18f80fo4848f28824179405@mail.gmail.com> like i said, in openzap.conf change number => 1 to number => and you will not see the call arriving at ext 1 anymore this has nothing to do with caller id. We so far have only tested the caller id code in US so if es uses the same one as uk or jp we may have to add some more code. Can you press F8 on the FS cli to turn on max debug and reproduce an inbound call and post the log to http://pastebin.freeswitch.org On Wed, Jan 14, 2009 at 8:38 AM, Tom?s wrote: > Hi, > > Anthony, I think that's my problem, when I receive a call from the PSTN, FS > receive number 1 instead of my house number and I don't know why. > > Michael, I live in Spain, Is it not "es" the tonegroup I should use? > > Thank you very much. > > On Wed, Jan 14, 2009 at 3:18 PM, Michael Jerris wrote: > >> I noticed tonegroup=es. What country are you in and do you know what >> method they use to do dtmf. Most likely we need a small tweak to set the >> dtmf method for your country. >> Mike >> >> On Jan 14, 2009, at 9:05 AM, Anthony Minessale wrote: >> >> >> number => 1 >> >> This value should be set to the DID of the FXO line. >> That way when a call hits FS it will go to that extension in the dialplan. >> This is unrelated to callerid, it's the destination not the source. >> >> If the line has caller-id it will also be available when it's collected >> after the 2nd ring. >> >> >> >> On Wed, Jan 14, 2009 at 5:54 AM, Tom?s wrote: >> >>> Hi all, >>> >>> I'm a new FreeSwitch user and this is my first email to the list. >>> >>> I'm trying to configure my Home PBX with a Wildcard X101P (configured as >>> FXO) and I have a problem receiving the caller/called ID from PSTN. >>> >>> This is the content of file "openzap.conf": >>> >>> [span zt] >>> name => OpenZAP >>> number => 1 >>> fxo-channel => 1 >>> >>> And this is the content of file "openzap.conf.xml": >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> As you can see the param "enable-callerid" is set to "true", but when I >>> received and incoming call, FreeSwitch doesn't get neither the caller number >>> nor the called number (instead of my home number, I receive a number 1, as >>> can be seen on the following log of an incoming call): >>> >>> 2009-01-14 20:43:12 [NOTICE] switch_channel.c:565 >>> switch_channel_set_name() New Channel OpenZAP/1:1/1 >>> [941d6234-e273-11dd-bcdf-89190a30fe61] >>> 2009-01-14 20:43:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >>> Processing OpenZAP->1 in context default >>> 2009-01-14 20:43:12 [NOTICE] switch_ivr.c:1255 >>> switch_ivr_session_transfer() Transfer OpenZAP/1:1/1 to enum[1 at default] >>> 2009-01-14 20:43:16 [INFO] switch_core_state_machine.c:122 >>> switch_core_standard_on_routing() No Route, Aborting >>> 2009-01-14 20:43:16 [NOTICE] switch_core_state_machine.c:123 >>> switch_core_standard_on_routing() Hangup OpenZAP/1:1/1 [CS_ROUTING] >>> [NO_ROUTE_DESTINATION] >>> 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 1 (OpenZAP/1:1/1 ) Ended >>> 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel OpenZAP/1:1/1 [CS_HANGUP] >>> 2009-01-14 20:43:23 [NOTICE] switch_channel.c:565 >>> switch_channel_set_name() New Channel OpenZAP/1:1/1 >>> [9af53280-e273-11dd-bcdf-89190a30fe61] >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/f004bc12/attachment.html From regs at kinetix.gr Wed Jan 14 07:15:19 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Wed, 14 Jan 2009 17:15:19 +0200 Subject: [Freeswitch-users] ipauth - directory Message-ID: <496E0187.8090805@kinetix.gr> I noticed an "ip=" setting in the brian.xml sample file. The comments state that this is used for ipauth (IP based authentication?) What exactly is this setting. I cannot find anything in the wiki about it. Does it replace the use of the + ACL mechanism for IP authentication? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From brian at freeswitch.org Wed Jan 14 07:21:35 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 14 Jan 2009 09:21:35 -0600 Subject: [Freeswitch-users] ipauth - directory In-Reply-To: <496E0187.8090805@kinetix.gr> References: <496E0187.8090805@kinetix.gr> Message-ID: <23DEEC73-4DC7-4056-902F-D74DD6644385@freeswitch.org> cidr= and the domains acl in acl.conf.xml then apply that ACL to the sofia profile. /b On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: > I noticed an "ip=" setting in the brian.xml sample file. > The comments state that this is used for ipauth (IP based > authentication?) > > What exactly is this setting. I cannot find anything in the wiki > about it. > Does it replace the use of the > > + ACL > > mechanism for IP authentication? From regs at kinetix.gr Wed Jan 14 07:36:43 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Wed, 14 Jan 2009 17:36:43 +0200 Subject: [Freeswitch-users] ipauth - directory In-Reply-To: <23DEEC73-4DC7-4056-902F-D74DD6644385@freeswitch.org> References: <496E0187.8090805@kinetix.gr> <23DEEC73-4DC7-4056-902F-D74DD6644385@freeswitch.org> Message-ID: <496E068B.6050404@kinetix.gr> Yes I know that. But what does the "ip=" setting do? Brian West wrote: > cidr= and the domains acl in acl.conf.xml then apply that ACL to the > sofia profile. > > /b > > On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: > > >> I noticed an "ip=" setting in the brian.xml sample file. >> The comments state that this is used for ipauth (IP based >> authentication?) >> >> What exactly is this setting. I cannot find anything in the wiki >> about it. >> Does it replace the use of the >> >> + ACL >> >> mechanism for IP authentication? >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/a785b938/attachment.html From jon at radel.com Wed Jan 14 07:42:49 2009 From: jon at radel.com (Jon Radel) Date: Wed, 14 Jan 2009 10:42:49 -0500 Subject: [Freeswitch-users] No caller/called ID received (Wildcard X101P) In-Reply-To: <46336fde0901140638l5f960600ra0b26e05f52b85e8@mail.gmail.com> References: <46336fde0901140354q5a9ba05cx505ead321de165cc@mail.gmail.com> <191c3a030901140605g54b3b669r4091e4ebb2723005@mail.gmail.com> <82719D59-5275-4FEC-B655-9A4CFB724D69@jerris.com> <46336fde0901140638l5f960600ra0b26e05f52b85e8@mail.gmail.com> Message-ID: <496E07F9.8040400@radel.com> Tom?s wrote: > Hi, > > Anthony, I think that's my problem, when I receive a call from the PSTN, > FS receive number 1 instead of my house number and I don't know why. If you use SIP trunking or something like an ISDN-PRI line, the number the call is to is delivered as part of the signaling, which is only way to make use of many phone numbers on a single physical circuit or connection. When you put a POTS line into an FXO port, there is no such information provided, as there is only one number on the line. (Leaving aside various schemes found in some countries such as using different ring patterns to indicate different numbers having been called.) So, as Anthony keeps pointing out, if you want FS to know the number of the line plugged into the FXO port, you have to configure it yourself. --Jon Radel -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3283 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/14cfa11f/attachment.bin From kokoska.rokoska at post.cz Wed Jan 14 08:05:16 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 14 Jan 2009 17:05:16 +0100 Subject: [Freeswitch-users] mod_sofia: NAT-ping & RPID bounties Message-ID: <496E0D3C.9050408@post.cz> Hi all, I have just post two bounties (NAT-ping and RPID changes). So I hope someone may be interested :-) BTW: Both changes have to be (and stay in the future) integral part of FreeSWITCH code. Best regards, kokoska.rokoska From Prometheus001 at gmx.net Wed Jan 14 09:44:51 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 14 Jan 2009 18:44:51 +0100 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] Message-ID: <496E2493.2000207@gmx.net> 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 From Prometheus001 at gmx.net Wed Jan 14 10:20:43 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 14 Jan 2009 19:20:43 +0100 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <496E2493.2000207@gmx.net> References: <496E2493.2000207@gmx.net> Message-ID: <496E2CFB.3040603@gmx.net> Some more info: Now I cannot get any incoming call through OpenZAP. "oz dump 1" and "oz dump 2" show that all channels are DOWN. After reload of mod_openzap everythings works again. I receive the following messages for each incoming call: 2009-01-14 19:07:29 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 12 -------------------------------------------------------------------------------- [08 02 00 19 4d 08 05 82 e6 33 30 33] 2009-01-14 19:07:29 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[4d] Size:[110] CRV: 25 (0x19, CTX: Originator) 2009-01-14 19:07:29 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 75ea20 (1:1) source isdn_data->channels_remote_crv[0x19] 2009-01-14 19:07:29 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() got clear channel sig [STOP] 2009-01-14 19:07:29 [DEBUG] ozmod_isdn.c:440 zap_isdn_931_34() Received Release in state DOWN, requested hangup for channel 1:0 2009-01-14 19:07:29 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 9 -------------------------------------------------------------------------------- [08 02 80 19 5a 08 02 82 e6] 2009-01-14 19:07:29 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse error [1] [Q931E_NO_ERROR] Peter P GMX schrieb: > 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From klaus.teller at gmx.net Wed Jan 14 12:49:02 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Wed, 14 Jan 2009 21:49:02 +0100 Subject: [Freeswitch-users] Internal vs. External Call, How to detect answered call? Message-ID: <20090114204902.106450@gmx.net> Hi, I'm unable to detect whether a call was answered or not. Using a custom Java API to connect to Freeswitch Socket Interface, i observed that: 1) session.originate("sofia/internal/1003%192.168.50.94") does block until the callee picks up while 2) session.originate("sofia/gateway/sip.gafachi.com/14156782222") does not block and returns +OK (and the channel identifier) even when the callee doesn't pick up. I was thinking that the channel Id is returned only when the callee picks up. Since this is apparently not the case, what is the definitive way to know if the remote user picked up? Thanks, Klaus. -- Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a From brian at freeswitch.org Wed Jan 14 12:58:26 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 14 Jan 2009 14:58:26 -0600 Subject: [Freeswitch-users] SVN Commits now tweet! Message-ID: <7E1D5336-9B2B-4CD0-9A00-C35F18BE4938@freeswitch.org> http://twitter.com/freeswitchsvn Check it out. ;) /b From msc at freeswitch.org Wed Jan 14 13:07:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Jan 2009 13:07:45 -0800 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <496E2493.2000207@gmx.net> References: <496E2493.2000207@gmx.net> Message-ID: <87f2f3b90901141307m3d4e0e60hce337ad36b121369@mail.gmail.com> 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 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kokoska.rokoska at post.cz Wed Jan 14 13:32:03 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Wed, 14 Jan 2009 22:32:03 +0100 Subject: [Freeswitch-users] Registrar performance Message-ID: <496E59D3.7070006@post.cz> Hi all, could someone be so kind and share some data about concurrently registered users at FreeSWITCH from production usage? I have done few sipp testing and fall in doubt... BTW: I use xml_curl for directory and the data are served through: 1. apache, php, mysql 2. lighttpd, "c" binary, mysql and in both cases the performance is near the same - well, let say, not differs dramatically. When run kamailio with usrloc db mode 3 (no cache, all in db) on the same machine, performance is very good, so I hope the uderlaying system is setted-up properly. Thanks a lot! Best regards, kokoska.rokoska From ajlong at worldlink.net Wed Jan 14 14:44:48 2009 From: ajlong at worldlink.net (Adam Long) Date: Wed, 14 Jan 2009 17:44:48 -0500 Subject: [Freeswitch-users] Using mod_managed Linux/Mono 2.02 Message-ID: <00e401c97699$b4935c80$1dba1580$@net> 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 at 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 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/20090114/6e728b72/attachment.html From Miroslav.Hostinsky at sitronicsts.com Wed Jan 14 14:51:07 2009 From: Miroslav.Hostinsky at sitronicsts.com (Hostinsky Miroslav) Date: Wed, 14 Jan 2009 23:51:07 +0100 Subject: [Freeswitch-users] ilbc & alaw transcoding not working correctly? Message-ID: Hello all!, I am using freeswitch with Nokia E65 phone. When phone uses iLBC codec and I am trying "monkey demo" in the example IVR sound is choppy. When using alaw/ulaw there is no problem, it works OK. IVR prompts work OK with G711 and iLBC, there is no problem. So probably freeswitch is not able properly transcode between iLBC and G711 (when calling to external SIP peer?). Machine is powerful but it runs inside Xen (but load is minimal, I think this is not problem). So, what do you think? Where is problem? Thank you! -- Miro -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/d5d41879/attachment.html From brian at freeswitch.org Wed Jan 14 15:14:09 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 14 Jan 2009 17:14:09 -0600 Subject: [Freeswitch-users] ilbc & alaw transcoding not working correctly? In-Reply-To: References: Message-ID: Please collect the sip traces pcaps and details and open a jira. /b On Jan 14, 2009, at 4:51 PM, Hostinsky Miroslav wrote: > Hello all!, > > I am using freeswitch with Nokia E65 phone. When phone uses iLBC > codec and I am trying "monkey demo" in the example IVR sound is > choppy. When using alaw/ulaw there is no problem, it works OK. IVR > prompts work OK with G711 and iLBC, there is no problem. So probably > freeswitch is not able properly transcode between iLBC and G711 > (when calling to external SIP peer?). > > Machine is powerful but it runs inside Xen (but load is minimal, I > think this is not problem). > > So, what do you think? Where is problem? > > Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/739f74af/attachment.html From scott.ellis at novatex.com.au Wed Jan 14 15:24:15 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Thu, 15 Jan 2009 10:24:15 +1100 Subject: [Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call In-Reply-To: <191c3a030901140539y4f8ffa20nefdc23209c1b0df7@mail.gmail.com> References: <496DB86A.2000708@novatex.com.au> <191c3a030901140539y4f8ffa20nefdc23209c1b0df7@mail.gmail.com> Message-ID: <496E741F.4020208@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/b0ceaf48/attachment-0001.html From timb0311 at hotmail.com Wed Jan 14 17:13:27 2009 From: timb0311 at hotmail.com (Tim B) Date: Wed, 14 Jan 2009 20:13:27 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 Message-ID: Got mod_managed compiled and installed. Now it isn't loading. See below... 1) Donwloaded fresh from SVN 2) Compiled... and installed.. OK [root at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig [root at phone2 mod_managed]# make [root at phone2 mod_managed]# make install 3) Added to modules.conf.xml : 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_012009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/6a5facdc/attachment.html From mgg at giagnocavo.net Wed Jan 14 21:34:20 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 15 Jan 2009 00:34:20 -0500 Subject: [Freeswitch-users] Using mod_managed Linux/Mono 2.02 In-Reply-To: <00e401c97699$b4935c80$1dba1580$@net> References: <00e401c97699$b4935c80$1dba1580$@net> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670235BBB97F@mse17be1.mse17.exchange.ms> 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long Sent: Wednesday, January 14, 2009 3:45 PM To: freeswitch-users at 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 at 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 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/20090115/73ac27e4/attachment.html From scott.ellis at novatex.com.au Wed Jan 14 22:50:30 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Thu, 15 Jan 2009 17:50:30 +1100 Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent? In-Reply-To: <496EBFDE.2070305@drlake.com.au> References: <496EBFDE.2070305@drlake.com.au> Message-ID: <496EDCB6.4020802@novatex.com.au> 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 From scott.ellis at novatex.com.au Wed Jan 14 23:16:13 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Thu, 15 Jan 2009 18:16:13 +1100 Subject: [Freeswitch-users] Country specific tones - how to contribute? Message-ID: <496EE2BD.2050102@novatex.com.au> 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 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 From jason at jasonjgw.net Wed Jan 14 23:24:05 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 15 Jan 2009 18:24:05 +1100 Subject: [Freeswitch-users] Country specific tones - how to contribute? In-Reply-To: <496EE2BD.2050102@novatex.com.au> References: <496EE2BD.2050102@novatex.com.au> Message-ID: <20090115072405.GA15789@jdc.jasonjgw.net> Scott Ellis 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 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 $${variable-name} syntax. There's an ITU document referred to on the wiki with the official definitions of ringback and other tones for various countries. From juanbackson at gmail.com Wed Jan 14 23:43:20 2009 From: juanbackson at gmail.com (Juan Backson) Date: Thu, 15 Jan 2009 15:43:20 +0800 Subject: [Freeswitch-users] Changes in PlayAndGetDigits Message-ID: <27c25bc40901142343l34a3e99ftecf0df971e8e32f6@mail.gmail.com> Hi, Is there a change in the playAndGetDigits api? In the old release, 11102, my lua script is working but is not working in the latest release. The error I am getting is " Error in playAndGetDigits expected 10..10 args, got 9 ". Thanks, JB From Prometheus001 at gmx.net Thu Jan 15 00:20:18 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 15 Jan 2009 09:20:18 +0100 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <87f2f3b90901141307m3d4e0e60hce337ad36b121369@mail.gmail.com> References: <496E2493.2000207@gmx.net> <87f2f3b90901141307m3d4e0e60hce337ad36b121369@mail.gmail.com> Message-ID: <496EF1C2.8020607@gmx.net> Hello Michael, how much $$ 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 at lists.freeswitch.org/msg05792.html 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 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From pmhshz at gmail.com Thu Jan 15 00:38:04 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 15 Jan 2009 00:38:04 -0800 (PST) Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <191c3a030901130601m7f94fd15ra7c05d35b6408497@mail.gmail.com> References: <21386948.post@talk.nabble.com> <21414332.post@talk.nabble.com> <428196FD-5E22-4745-BE8D-5C4AFFC3B2CF@freeswitch.org> <21433120.post@talk.nabble.com> <191c3a030901130601m7f94fd15ra7c05d35b6408497@mail.gmail.com> Message-ID: <21473408.post@talk.nabble.com> There NO previous version of FS installed before, and FS 1.0.2 is also freshly installed. Anthony Minessale-2 wrote: > > please remove FS src and dest dir from your machine and recompile fresh > from > scratch. > > > On Tue, Jan 13, 2009 at 4:48 AM, shehzad p wrote: > >> >> >> Please find the output of bt from below pastebin link: >> http://pastebin.freeswitch.org/6757 >> >> Thanks, >> pms >> >> Michael S Collins wrote: >> > >> > Could you please do a backtrace and post it to a pastebin? If in Linux >> > do this: >> > gdb /path/to/freeswitch /path/to/corefile >> > >> > -MC >> > >> > Sent from my iPhone >> > >> > On Jan 12, 2009, at 5:23 AM, shehzad p wrote: >> > >> >> >> >> Hi all, >> >> I am also testing FS release 1.0.2, but I faced strange problem. >> >> When I stop freeswitch (from CLI using ... or shutdown), Freeswitch >> >> ends >> >> with showing "Segmentation fault": >> >> Below is the last 15 lines when fault occures. Sometimes this does not >> >> happen and FS shut down normally. >> >> >> >> === >> >> === >> >> === >> >> === >> >> === >> >> ====================================================================== >> >> 2009-01-12 16:52:56 [CONSOLE] switch_loadable_module.c:1244 >> >> do_shutdown() >> >> mod_esf unloaded. >> >> 2009-01-12 16:52:56 [CONSOLE] switch_core.c:1462 switch_core_destroy() >> >> Closing Event Engine. >> >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 >> >> switch_event_shutdown() >> >> Stopping event queue 0 >> >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 >> >> switch_event_shutdown() >> >> Stopping event queue 1 >> >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 >> >> switch_event_thread() Event >> >> Thread 0 Ended. >> >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:445 >> >> switch_event_shutdown() >> >> Stopping dispatch queue 0 >> >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 >> >> switch_event_thread() Event >> >> Thread 1 Ended. >> >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:247 >> >> switch_event_dispatch_thread() Dispatch Thread 0 Ended. >> >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 >> >> switch_event_thread() Event >> >> Thread 2 Ended. >> >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:414 >> >> switch_core_memory_reclaim_events() Returning 23 recycled event(s) >> >> 1012 >> >> bytes >> >> 2009-01-12 16:52:56 [CONSOLE] switch_event.c:416 >> >> switch_core_memory_reclaim_events() Returning 331 recycled event >> >> header(s) >> >> 5296 bytes >> >> 2009-01-12 16:52:56 [CONSOLE] switch_core_sqldb.c:539 >> >> switch_core_sqldb_stop() Waiting for unfinished SQL transactions >> >> 2009-01-12 16:52:56 [NOTICE] switch_core_sqldb.c:199 >> >> switch_core_sql_thread() SQL thread ending >> >> 2009-01-12 16:52:56 [CONSOLE] switch_scheduler.c:303 >> >> switch_scheduler_task_thread_stop() Stopping Task Thread >> >> Segmentation fault (core dumped) >> >> === >> >> === >> >> === >> >> === >> >> === >> >> ====================================================================== >> >> >> >> What should be the cause of such crash. >> >> >> >> >> >> ahgindia wrote: >> >>> >> >>> Hi All, >> >>> Recently I was testing the new freeswitch release 1.0.2 >> >>> The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo CPU >> >>> E7200 @ 2.53GHz processor. >> >>> But it crashed, when there were 96 active calls in it (as can be >> >>> seen from >> >>> "show calls" on freeswitch cli) >> >>> There is a dump file for it, in the folder from where i started the >> >>> freeswitch. >> >>> Let me know how can we know the cause of the crash. >> >>> >> >> >> >> >> >> >> >> -- >> >> View this message in context: >> >> >> http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21414332.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at 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 at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21433120.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21473408.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Prometheus001 at gmx.net Thu Jan 15 00:43:09 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 15 Jan 2009 09:43:09 +0100 Subject: [Freeswitch-users] OpenZAP hardware timers Message-ID: <496EF71D.1010008@gmx.net> Is there a way to use the hardware timers e.g. of a PRI card in fresswitch? Or other question: Is it recommended to use those if they are available? I have installed a dual PRI card, and "show timer" shows one soft timer. Best regards Peter From scott.ellis at novatex.com.au Thu Jan 15 01:13:12 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Thu, 15 Jan 2009 20:13:12 +1100 Subject: [Freeswitch-users] Country specific tones - how to contribute? In-Reply-To: <20090115072405.GA15789@jdc.jasonjgw.net> References: <496EE2BD.2050102@novatex.com.au> <20090115072405.GA15789@jdc.jasonjgw.net> Message-ID: <496EFE28.9040702@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/1c96a3c1/attachment.html From avmanansala at gmail.com Thu Jan 15 01:16:59 2009 From: avmanansala at gmail.com (Lito Manansala) Date: Thu, 15 Jan 2009 17:16:59 +0800 Subject: [Freeswitch-users] FS 1.2 Windows Error Message-ID: <8e0176e60901150116h74847c65s12433b5b6b6993d3@mail.gmail.com> Hi, Im getting error on startup when executing freeSWITCH.exe , "The procedure entry point_apr_md5 at 12 could not be located in the dynamic library libaprutil.dll" -- /Lito -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/0f1fb271/attachment.html From scott.ellis at novatex.com.au Thu Jan 15 02:17:28 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Thu, 15 Jan 2009 21:17:28 +1100 Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent? In-Reply-To: <496EDCB6.4020802@novatex.com.au> References: <496EBFDE.2070305@drlake.com.au> <496EDCB6.4020802@novatex.com.au> Message-ID: <496F0D38.5030904@novatex.com.au> After poking around in the code, it looks like if I set in openzap.conf.xml, it should skip the GET_CALLERID state, and I should get the call answered straight away. mod_openzap.c } else if (!strcasecmp(var, "enable-callerid")) { enable_callerid = val; if (zap_configure_span("analog", span, on_analog_signal, "tonemap", tonegroup, "digit_timeout", &to, "max_dialstr", &max, "hotline", hotline, "enable_callerid", enable_callerid, TAG_END) != ZAP_SUCCESS) { zap_log(ZAP_LOG_ERROR, "Error starting OpenZAP span %d\n", span_id); continue; } ozmod_analog.c else if (!strcasecmp(var, "enable_callerid")) { if (!(val = va_arg(ap, char *))) { break; } if (zap_true(val)) { flags |= ZAP_ANALOG_CALLERID; } else { flags &= ~ZAP_ANALOG_CALLERID; } and case ZAP_OOB_RING_START: { if (event->channel->type != ZAP_CHAN_TYPE_FXO) { zap_log(ZAP_LOG_ERROR, "Cannot get a RING_START event on a non-fxo channel, please check your config.\n"); zap_set_state_locked(event->channel, ZAP_CHANNEL_STATE_DOWN); goto end; } if (!event->channel->ring_count && (event->channel->state == ZAP_CHANNEL_STATE_DOWN && !zap_test_flag(event->channel, ZAP_CHANNEL_INTHREAD))) { if (zap_test_flag(analog_data, ZAP_ANALOG_CALLERID)) { zap_set_state_locked(event->channel, ZAP_CHANNEL_STATE_GET_CALLERID); } else { zap_set_state_locked(event->channel, ZAP_CHANNEL_STATE_IDLE); } event->channel->ring_count = 1; zap_mutex_unlock(event->channel->mutex); locked = 0; zap_thread_create_detached(zap_analog_channel_run, event->channel); } else { event->channel->ring_count++; } } break; 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][1:1] STATE [DOWN] 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:760 process_event() Changing state on 1:1 from DOWN to GET_CALLERID 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() ANALOG CHANNEL thread starting. 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 1:1 for GET_CALLERID 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][1:1] STATE [GET_CALLERID] 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][1:1] STATE [GET_CALLERID] 2009-01-15 20:19:45 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][1:1] STATE [GET_CALLERID] 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][1:1] STATE [GET_CALLERID] 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][1:1] STATE [GET_CALLERID] 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][1:1] STATE [GET_CALLERID] 2009-01-15 20:19:48 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][1:1] STATE [GET_CALLERID] 2009-01-15 20:19:49 [DEBUG] ozmod_analog.c:292 zap_analog_channel_run() Changing state on 1:1 from GET_CALLERID to IDLE 2009-01-15 20:19:49 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 1:1 for IDLE 2009-01-15 20:19:49 [DEBUG] mod_openzap.c:1165 on_fxo_signal() got FXO sig 1:1 [START] The code all looks right, but I am not getting what I think should happen. Anyone with any ideas? Scott Scott Ellis wrote: > 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 > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From helmut.kuper at ewetel.de Thu Jan 15 02:47:37 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 15 Jan 2009 11:47:37 +0100 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <496EF1C2.8020607@gmx.net> References: <496E2493.2000207@gmx.net> <87f2f3b90901141307m3d4e0e60hce337ad36b121369@mail.gmail.com> <496EF1C2.8020607@gmx.net> Message-ID: <496F1449.6000407@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Michael, it must not be the case here, but I had the same error, when incomming calles used a wrong numbering plan resp not the one, FS expected. Just a hint. regards Helmut Am 15.01.2009 09:20, schrieb Peter P GMX: > 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 at lists.freeswitch.org/msg05792.html > 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 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklvFEkACgkQ4tZeNddg3dxitgCeIgNS+qUwYQ0ypc1KyXjRO3OV OFwAn1TeaNP466OWErmqEFr9H9p2Wam5 =2NfD -----END PGP SIGNATURE----- From helmut.kuper at ewetel.de Thu Jan 15 02:58:46 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 15 Jan 2009 11:58:46 +0100 Subject: [Freeswitch-users] SQLExecute catches not all errors Message-ID: <496F16E6.6080004@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I moved the voicemail database from sqlite to mysql via odbc. FS started up and connected successfully to the empty database. Normally voicemail adds the neccessary database tables automaticly during startup. In this case I forgot to add permissions for creating, dropping and altering tables to the database user in mysql. So no SM table was createt in database. Unfortunately FS can not detect this. FS thinks everything is ok. After adding the permissions one table was created but the voicemail_prefs wasn't. This was, because I extented the create statement for this table - and the statement was wrong, so mysql couldn't execute it. This case wasn't detected by FS as well and FS resp. voicemail modul thought everything is fine Is there a way to detect those errors, because bug hunting can be last quite long without error messages especially when you have no way to access mysql.log to see the sql statements from FS. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklvFuYACgkQ4tZeNddg3dwT5ACfd/ArMsLfsLrnc5peY1qxaDWu kbsAn00gvNJjXwtFYIX41lbbgGWF+m1P =GO2m -----END PGP SIGNATURE----- From Prometheus001 at gmx.net Thu Jan 15 03:06:43 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 15 Jan 2009 12:06:43 +0100 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <496F1449.6000407@ewetel.de> References: <496E2493.2000207@gmx.net> <87f2f3b90901141307m3d4e0e60hce337ad36b121369@mail.gmail.com> <496EF1C2.8020607@gmx.net> <496F1449.6000407@ewetel.de> Message-ID: <496F18C3.3030807@gmx.net> Helmut, can you give me a hint, how you worked around this? Best regards Peter Helmut Kuper schrieb: > Hi Michael, > > it must not be the case here, but I had the same error, when incomming > calles used a wrong numbering plan resp not the one, FS expected. > > Just a hint. > > regards > Helmut > > > Am 15.01.2009 09:20, schrieb Peter P GMX: > > 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 at lists.freeswitch.org/msg05792.html > > 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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From imthiyazg at gmail.com Thu Jan 15 03:12:07 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Thu, 15 Jan 2009 16:42:07 +0530 Subject: [Freeswitch-users] Sending SMS to SIPtoGSM gateway Message-ID: <8595daf70901150312t595e2156ha712167fe7876372@mail.gmail.com> Hi I have a IP to GSM gateway which supports SIP. How I can send SMS to the GSM phone using FreeSwitch + SIP GSM GW? Thanks Imthiyaz From scott.ellis at novatex.com.au Thu Jan 15 03:14:03 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Thu, 15 Jan 2009 22:14:03 +1100 Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent - a solution of sorts. In-Reply-To: <496F0D38.5030904@novatex.com.au> References: <496EBFDE.2070305@drlake.com.au> <496EDCB6.4020802@novatex.com.au> <496F0D38.5030904@novatex.com.au> Message-ID: <496F1A7B.1060400@novatex.com.au> So I decided to hack the code to see if I could just get it to do what I wanted - assuming some kind of error in the options setting. First I changed the state change code to just skip straight to IDLE if (!event->channel->ring_count && (event->channel->state == ZAP_CHANNEL_STATE_DOWN && !zap_test_flag(event->channel, ZAP_CHANNEL_INTHREAD))) { // if (zap_test_flag(analog_data, ZAP_ANALOG_CALLERID)) { // zap_set_state_locked(event->channel, ZAP_CHANNEL_STATE_GET_CALLERID); // } else { zap_set_state_locked(event->channel, ZAP_CHANNEL_STATE_IDLE); // } event->channel->ring_count = 1; zap_mutex_unlock(event->channel->mutex); locked = 0; zap_thread_create_detached(zap_analog_channel_run, event->channel); } else { event->channel->ring_count++; } So we skip the GET_CALLERID state altogether. This generated an illegal state change message cannot go from DOWN to IDLE So then changed the code to if (!event->channel->ring_count && (event->channel->state == ZAP_CHANNEL_STATE_DOWN && !zap_test_flag(event->channel, ZAP_CHANNEL_INTHREAD))) { // if (zap_test_flag(analog_data, ZAP_ANALOG_CALLERID)) { zap_set_state_locked(event->channel, ZAP_CHANNEL_STATE_GET_CALLERID); // } else { zap_set_state_locked(event->channel, ZAP_CHANNEL_STATE_IDLE); // } event->channel->ring_count = 1; zap_mutex_unlock(event->channel->mutex); locked = 0; zap_thread_create_detached(zap_analog_channel_run, event->channel); } else { event->channel->ring_count++; } Allowing the state change to GET_CALLERID, then immediately to IDLE. This works perfectly - the call is answered straight away. At the moment I don't know enough about linux debugging to step through the parameter code to see why setting get caller ID to false in openzap.conf.xml does not get passed through, but even if it does the current code will still run into the illegal state change error. 2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:744 process_event() EVENT [RING_START][1:1] STATE [DOWN] 2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:760 process_event() Changing state on 1:1 from DOWN to GET_CALLERID 2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:762 process_event() Changing state on 1:1 from GET_CALLERID to IDLE 2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() ANALOG CHANNEL thread starting. 2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 1:1 for IDLE 2009-01-15 21:59:18 [DEBUG] mod_openzap.c:1165 on_fxo_signal() got FXO sig 1:1 [START] 2009-01-15 21:59:18 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU 20ms 2009-01-15 21:59:18 [DEBUG] mod_openzap.c:1137 zap_channel_from_event() Connect inbound channel OpenZAP/1:1/1 2009-01-15 21:59:18 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/1 [8e2a55c8-e2f3-11dd-adfd-6d934f226ffd] Will go and put this into JIRA in the next couple of days. Scott Scott Ellis wrote: > After poking around in the code, it looks like if I set name="enable-callerid" value="false"/> in openzap.conf.xml, it should > skip the GET_CALLERID state, and I should get the call answered straight > away. > > mod_openzap.c > > } else if (!strcasecmp(var, "enable-callerid")) { > enable_callerid = val; > > > if (zap_configure_span("analog", span, on_analog_signal, > "tonemap", tonegroup, > "digit_timeout", &to, > "max_dialstr", &max, > "hotline", hotline, > "enable_callerid", enable_callerid, > TAG_END) != ZAP_SUCCESS) { > zap_log(ZAP_LOG_ERROR, "Error starting OpenZAP span > %d\n", span_id); > continue; > } > > ozmod_analog.c > > else if (!strcasecmp(var, "enable_callerid")) { > if (!(val = va_arg(ap, char *))) { > break; > } > if (zap_true(val)) { > flags |= ZAP_ANALOG_CALLERID; > } else { > flags &= ~ZAP_ANALOG_CALLERID; > } > > and > > case ZAP_OOB_RING_START: > { > if (event->channel->type != ZAP_CHAN_TYPE_FXO) { > zap_log(ZAP_LOG_ERROR, "Cannot get a RING_START event on > a non-fxo channel, please check your config.\n"); > zap_set_state_locked(event->channel, > ZAP_CHANNEL_STATE_DOWN); > goto end; > } > if (!event->channel->ring_count && (event->channel->state == > ZAP_CHANNEL_STATE_DOWN && !zap_test_flag(event->channel, > ZAP_CHANNEL_INTHREAD))) { > if (zap_test_flag(analog_data, ZAP_ANALOG_CALLERID)) { > zap_set_state_locked(event->channel, > ZAP_CHANNEL_STATE_GET_CALLERID); > } else { > zap_set_state_locked(event->channel, > ZAP_CHANNEL_STATE_IDLE); > } > event->channel->ring_count = 1; > zap_mutex_unlock(event->channel->mutex); > locked = 0; > zap_thread_create_detached(zap_analog_channel_run, > event->channel); > } else { > event->channel->ring_count++; > } > } > break; > > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [DOWN] > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:760 process_event() Changing > state on 1:1 from DOWN to GET_CALLERID > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() > ANALOG CHANNEL thread starting. > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 1:1 for GET_CALLERID > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:45 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:48 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:49 [DEBUG] ozmod_analog.c:292 zap_analog_channel_run() > Changing state on 1:1 from GET_CALLERID to IDLE > 2009-01-15 20:19:49 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 1:1 for IDLE > 2009-01-15 20:19:49 [DEBUG] mod_openzap.c:1165 on_fxo_signal() got FXO > sig 1:1 [START] > > The code all looks right, but I am not getting what I think should > happen. Anyone with any ideas? > > Scott > > Scott Ellis wrote: > >> 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 >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jonas.gauffin at gmail.com Thu Jan 15 03:20:37 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Thu, 15 Jan 2009 12:20:37 +0100 Subject: [Freeswitch-users] spidermonkey problems Message-ID: Hello I got problems with hanging spidermonkey sessions and need some advice on how to debug them. I've made a javascript queue application that uses mod_spidermonkey_socket. It works fine for a while, but after some calls I noticed that calls didnt get transferred to agents. The reason was that earlier calls had not been terminated properly. freeswitch at test1> hupall 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 switch_core_session_hupall() Giving up with 8 sessions remaining API CALL [hupall()] output: +OK hangup all channels with cause MANAGER_REQUEST freeswitch at test1> show calls API CALL [show(calls)] output: 0 total. As you can see, 8 sessions are alive, but none of them are listed as calls. What kind of logs should I turn on to see what is happening with those sessions? Thanks, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/e45052bf/attachment.html From helmut.kuper at ewetel.de Thu Jan 15 04:30:43 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 15 Jan 2009 13:30:43 +0100 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <496F18C3.3030807@gmx.net> References: <496E2493.2000207@gmx.net> <87f2f3b90901141307m3d4e0e60hce337ad36b121369@mail.gmail.com> <496EF1C2.8020607@gmx.net> <496F1449.6000407@ewetel.de> <496F18C3.3030807@gmx.net> Message-ID: <496F2C73.9030400@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Peter, it was simply a change in our TDM Voice Switch. It used a different numbering plan and we changed it to "national" to get it work with FS and openzap in Q921/Q931 mode. What I still search is a way to configure the numberplan in FS. To make it clear: In my case it didn't work from the second FS starts up. So this differs from your problem. To get an idea what's going on on the TDM link I used a TDM D-Channel monitoring device and traced the d-channel messages exchanged between FS and TDM. That should make it easier to see what's wrong when the problems occur. But you can also increase FS debug level to debug and trace the Q921 and Q931 messages in FS console via fs_cli during runtime. You have to set this in openzap.conf.xml: Unfortunately FS doesn't decode the whole Q931 messages, but it shows a hex representation of the message, so you can manually decode it with this documents: Q.931: http://www.itu.int/rec/T-REC-Q.931-199805-I/en Q.921: http://www.itu.int/rec/T-REC-Q.921-199709-I/en I think for numberingplan issues you only have to track the Q.931 messages. The last idea I have to get some light into your problem and to avoid manually decoding, try to convert FS's q931 hexdump into wiresharks pcap format. Wireshark should be able to decode it :) http://wiki.wireshark.org/Q.931 Maybe it's a good idea to implement a wireshark export for those messages in FS. This will make debugging easy and cheap. Hope it helps a bit. regards helmut Am 15.01.2009 12:06, schrieb Peter P GMX: > Helmut, > > can you give me a hint, how you worked around this? > > Best regards > Peter > > Helmut Kuper schrieb: >> Hi Michael, >> >> it must not be the case here, but I had the same error, when incomming >> calles used a wrong numbering plan resp not the one, FS expected. >> >> Just a hint. >> >> regards >> Helmut >> >> >> Am 15.01.2009 09:20, schrieb Peter P GMX: >>> 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 at lists.freeswitch.org/msg05792.html >>> 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 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklvLHMACgkQ4tZeNddg3dxF0ACgpMqGf8hu1iSKbOG7nG2o1HZN qdEAoIpTY3Bgwv9wzhV7lq7IKtvDxO5/ =lDVf -----END PGP SIGNATURE----- From tomasborrella at gmail.com Thu Jan 15 04:35:31 2009 From: tomasborrella at gmail.com (=?ISO-8859-1?Q?Tom=E1s?=) Date: Thu, 15 Jan 2009 13:35:31 +0100 Subject: [Freeswitch-users] No caller/called ID received (Wildcard X101P) In-Reply-To: <496E07F9.8040400@radel.com> References: <46336fde0901140354q5a9ba05cx505ead321de165cc@mail.gmail.com> <191c3a030901140605g54b3b669r4091e4ebb2723005@mail.gmail.com> <82719D59-5275-4FEC-B655-9A4CFB724D69@jerris.com> <46336fde0901140638l5f960600ra0b26e05f52b85e8@mail.gmail.com> <496E07F9.8040400@radel.com> Message-ID: <46336fde0901150435w3dcbba10t94122ef4fd2d39a2@mail.gmail.com> Thank you very much for your help, I've realized I was specting to receive my house phone number having a POTS line and that's not possible. So, I've put my house number in openzap.conf: [span zt] name => OpenZAP number => 919999999 fxo-channel => 1 And I've added an extension on the default dialplan: So I was hopping the IVR answer the call when it is received but instead of that nothing happens, this is the log of one incoming call: 2009-01-15 21:16:56 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/919999999[74cb661e-e341-11dd-acde-9740a65ca868] 2009-01-15 21:16:56 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP->919999999 in context default 2009-01-15 21:16:56 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:1/914021339 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-15 21:16:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 13 (OpenZAP/1:1/919999999) Ended 2009-01-15 21:16:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/919999999 [CS_HANGUP] 2009-01-15 21:17:03 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/919999999 [78fe36b2-e341-11dd-acde-9740a65ca868] 2009-01-15 21:17:03 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP->919999999 in context default 2009-01-15 21:17:03 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:1/914021339 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-15 21:17:03 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 14 (OpenZAP/1:1/919999999 ) Ended 2009-01-15 21:17:03 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/919999999 [CS_HANGUP] 2009-01-15 21:17:09 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/919999999 [7c60cfea-e341-11dd-acde-9740a65ca868] 2009-01-15 21:17:09 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP->919999999 in context default 2009-01-15 21:17:09 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:1/919999999 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-15 21:17:09 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 15 (OpenZAP/1:1/919999999 ) Ended 2009-01-15 21:17:09 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/919999999 [CS_HANGUP] Someone knows what's happening? Thank you. On Wed, Jan 14, 2009 at 4:42 PM, Jon Radel wrote: > Tom?s wrote: > > Hi, > > > > Anthony, I think that's my problem, when I receive a call from the PSTN, > > FS receive number 1 instead of my house number and I don't know why. > > If you use SIP trunking or something like an ISDN-PRI line, the number > the call is to is delivered as part of the signaling, which is only way > to make use of many phone numbers on a single physical circuit or > connection. When you put a POTS line into an FXO port, there is no such > information provided, as there is only one number on the line. (Leaving > aside various schemes found in some countries such as using different > ring patterns to indicate different numbers having been called.) > > So, as Anthony keeps pointing out, if you want FS to know the number of > the line plugged into the FXO port, you have to configure it yourself. > > --Jon Radel > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/06dc45a3/attachment-0001.html From krice at suspicious.org Thu Jan 15 05:44:32 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 15 Jan 2009 07:44:32 -0600 Subject: [Freeswitch-users] Announcing the FreeSWITCH Technology Preview VMWare Appliance. Message-ID: FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac. Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed and operational. See /etc/motd on the running image for all the good information. We'll be unvailing a wiki page for this shortly. For now you can get the head start by downloading this at http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip Have fun guys! Ken krice at freeswitch.org krice at rmktek.com From krice at freeswitch.org Wed Jan 14 20:05:21 2009 From: krice at freeswitch.org (Ken Rice) Date: Wed, 14 Jan 2009 22:05:21 -0600 Subject: [Freeswitch-users] Announcing the FreeSWITCH Technology Preview VMWare Appliance. Message-ID: Hey guys, I'm not trying to start 1 a day releases, Things just happened to fall that way... FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac. Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed and operational. See /etc/motd on the running image for all the good information. We'll be unvailing a wiki page for this shortly. For now you can get the head start by downloading this at http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip Have fun guys! Ken From intralanman at freeswitch.org Thu Jan 15 06:46:40 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 15 Jan 2009 14:46:40 +0000 Subject: [Freeswitch-users] mod_sofia: NAT-ping & RPID bounties In-Reply-To: <496E0D3C.9050408@post.cz> References: <496E0D3C.9050408@post.cz> Message-ID: <496F4C50.3000101@freeswitch.org> kokoska rokoska wrote: > Hi all, > > I have just post two bounties Where did you post these bounties? We've started moving bounties away from the wiki and adding them to jira instead. ( so that progress can be followed more closely ) -Ray -------------- next part -------------- A non-text attachment was scrubbed... Name: intralanman.vcf Type: text/x-vcard Size: 209 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/25143f9e/attachment.vcf From brian at freeswitch.org Thu Jan 15 07:08:20 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 09:08:20 -0600 Subject: [Freeswitch-users] Changes in PlayAndGetDigits In-Reply-To: <27c25bc40901142343l34a3e99ftecf0df971e8e32f6@mail.gmail.com> References: <27c25bc40901142343l34a3e99ftecf0df971e8e32f6@mail.gmail.com> Message-ID: <993E5AD4-FFDF-45CF-8046-9736C81C00CC@freeswitch.org> I added another arg to the list. I'll have to revisit this today to make sure I did this right for your case. /b On Jan 15, 2009, at 1:43 AM, Juan Backson wrote: > 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 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Jan 15 07:21:09 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 09:21:09 -0600 Subject: [Freeswitch-users] spidermonkey problems In-Reply-To: References: Message-ID: <543FB0CF-4EF5-44DD-9F98-0F11B05B9074@freeswitch.org> http://wiki.freeswitch.org/wiki/Report_Issue_Checklist Please open a jira and include your script and a test case. /b On Jan 15, 2009, at 5:20 AM, Jonas Gauffin wrote: > Hello > > I got problems with hanging spidermonkey sessions and need some > advice on how to debug them. > > I've made a javascript queue application that uses > mod_spidermonkey_socket. It works fine for a while, > but after some calls I noticed that calls didnt get transferred to > agents. The reason was that earlier > calls had not been terminated properly. > > freeswitch at test1> hupall > 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 > switch_core_session_hupall() Giving up with 8 sessions remaining > API CALL [hupall()] output: > +OK hangup all channels with cause MANAGER_REQUEST > > > freeswitch at test1> show calls > API CALL [show(calls)] output: > > 0 total. > > > As you can see, 8 sessions are alive, but none of them are listed as > calls. What kind of logs should I turn on to see what is happening > with those sessions? > > Thanks, > Jonas > _________ From brian at freeswitch.org Thu Jan 15 07:22:39 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 09:22:39 -0600 Subject: [Freeswitch-users] Sending SMS to SIPtoGSM gateway In-Reply-To: <8595daf70901150312t595e2156ha712167fe7876372@mail.gmail.com> References: <8595daf70901150312t595e2156ha712167fe7876372@mail.gmail.com> Message-ID: <58E49C41-A2FB-43A4-8EEE-6EDAE795197A@freeswitch.org> Refer to your manual on the SIP GSM GW and then inform us what exactly it needs for it to send an SMS then we can give you some sort of educated guess. Without this we have NO idea what gateway you're taking about... or what exactly it expects to send an SMS. /b On Jan 15, 2009, at 5:12 AM, Imthiyaz Ahmed wrote: > Hi > > I have a IP to GSM gateway which supports SIP. How I can send SMS to > the GSM phone using FreeSwitch + SIP GSM GW? > > Thanks > Imthiyaz From brian at freeswitch.org Thu Jan 15 07:23:30 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 09:23:30 -0600 Subject: [Freeswitch-users] Country specific tones - how to contribute? In-Reply-To: <496EE2BD.2050102@novatex.com.au> References: <496EE2BD.2050102@novatex.com.au> Message-ID: <0743B78F-24C9-4D28-9486-4AA02A881257@freeswitch.org> You can submit patches to http://jira.freeswitch.org thanks, /b On Jan 15, 2009, at 1:16 AM, Scott Ellis 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? > > I tried 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 From brian at freeswitch.org Thu Jan 15 07:39:46 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 09:39:46 -0600 Subject: [Freeswitch-users] Changes in PlayAndGetDigits In-Reply-To: <27c25bc40901142343l34a3e99ftecf0df971e8e32f6@mail.gmail.com> References: <27c25bc40901142343l34a3e99ftecf0df971e8e32f6@mail.gmail.com> Message-ID: <595F975D-6434-418F-ACC1-4A55F5379256@freeswitch.org> Update and try now... I think we fixed this to not break API compatibility. /b On Jan 15, 2009, at 1:43 AM, Juan Backson wrote: > 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 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kokoska.rokoska at post.cz Thu Jan 15 07:56:44 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Thu, 15 Jan 2009 16:56:44 +0100 Subject: [Freeswitch-users] mod_sofia: NAT-ping & RPID bounties In-Reply-To: <496F4C50.3000101@freeswitch.org> References: <496E0D3C.9050408@post.cz> <496F4C50.3000101@freeswitch.org> Message-ID: <496F5CBC.4000901@post.cz> Raymond Chandler napsal(a): > kokoska rokoska wrote: >> Hi all, >> >> I have just post two bounties > Where did you post these bounties? I have posted them to the Boutny wiki page (at the bottom of the page): http://wiki.freeswitch.org/wiki/Bounty > We've started moving bounties away > from the wiki and adding them to jira instead. ( so that progress can be > followed more closely ) > OK. Should I move my bounties or you'll be so kind and move them all (including mine)? :) Best regards, kokoska.rokoska From intralanman at freeswitch.org Thu Jan 15 08:13:54 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 15 Jan 2009 16:13:54 +0000 Subject: [Freeswitch-users] mod_sofia: NAT-ping & RPID bounties In-Reply-To: <496F5CBC.4000901@post.cz> References: <496E0D3C.9050408@post.cz> <496F4C50.3000101@freeswitch.org> <496F5CBC.4000901@post.cz> Message-ID: <496F60C2.4080107@freeswitch.org> kokoska rokoska wrote: > > Raymond Chandler napsal(a): > >> kokoska rokoska wrote: >> >>> Hi all, >>> >>> I have just post two bounties >>> >> Where did you post these bounties? >> > > I have posted them to the Boutny wiki page (at the bottom of the page): > http://wiki.freeswitch.org/wiki/Bounty > > > >> We've started moving bounties away >> from the wiki and adding them to jira instead. ( so that progress can be >> followed more closely ) >> >> > > OK. Should I move my bounties or you'll be so kind and move them all > (including mine)? :) > anyone still interested in a bounty posted, should move it to jira and remove it from the wiki.... most of the ones on the wiki have been done already, iirc http://jira.freeswitch.org/browse/BOUNTY -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/fab556df/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: intralanman.vcf Type: text/x-vcard Size: 209 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/fab556df/attachment.vcf From msc at freeswitch.org Thu Jan 15 08:58:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 15 Jan 2009 08:58:43 -0800 Subject: [Freeswitch-users] No caller/called ID received (Wildcard X101P) In-Reply-To: <46336fde0901150435w3dcbba10t94122ef4fd2d39a2@mail.gmail.com> References: <46336fde0901140354q5a9ba05cx505ead321de165cc@mail.gmail.com> <191c3a030901140605g54b3b669r4091e4ebb2723005@mail.gmail.com> <82719D59-5275-4FEC-B655-9A4CFB724D69@jerris.com> <46336fde0901140638l5f960600ra0b26e05f52b85e8@mail.gmail.com> <496E07F9.8040400@radel.com> <46336fde0901150435w3dcbba10t94122ef4fd2d39a2@mail.gmail.com> Message-ID: <87f2f3b90901150858w5eb1820fn7b88bf15232c87dd@mail.gmail.com> Could you repeat this test with debug loglevel turned on? (Press F8 or type "console loglevel 7"). Please put the results in pastebin.freeswitch.org. -MC On Thu, Jan 15, 2009 at 4:35 AM, Tom?s wrote: > Thank you very much for your help, I've realized I was specting to receive > my house phone number having a POTS line and that's not possible. > > So, I've put my house number in openzap.conf: > > [span zt] > name => OpenZAP > number => 919999999 > fxo-channel => 1 > > And I've added an extension on the default dialplan: > > > > > > > > > > So I was hopping the IVR answer the call when it is received but instead of > that nothing happens, this is the log of one incoming call: > > 2009-01-15 21:16:56 [NOTICE] switch_channel.c:565 switch_channel_set_name() > New Channel OpenZAP/1:1/919999999[74cb661e-e341-11dd-acde-9740a65ca868] > 2009-01-15 21:16:56 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing > OpenZAP->919999999 in context default > 2009-01-15 21:16:56 [NOTICE] switch_core_state_machine.c:168 > switch_core_standard_on_execute() Hangup OpenZAP/1:1/914021339 [CS_EXECUTE] > [NORMAL_CLEARING] > 2009-01-15 21:16:56 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 13 (OpenZAP/1:1/919999999) Ended > 2009-01-15 21:16:56 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel OpenZAP/1:1/919999999 > [CS_HANGUP] > 2009-01-15 21:17:03 [NOTICE] switch_channel.c:565 switch_channel_set_name() > New Channel OpenZAP/1:1/919999999 [78fe36b2-e341-11dd-acde-9740a65ca868] > 2009-01-15 21:17:03 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing > OpenZAP->919999999 in context default > 2009-01-15 21:17:03 [NOTICE] switch_core_state_machine.c:168 > switch_core_standard_on_execute() Hangup OpenZAP/1:1/914021339 [CS_EXECUTE] > [NORMAL_CLEARING] > 2009-01-15 21:17:03 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 14 (OpenZAP/1:1/919999999 ) Ended > 2009-01-15 21:17:03 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel OpenZAP/1:1/919999999 > [CS_HANGUP] > 2009-01-15 21:17:09 [NOTICE] switch_channel.c:565 switch_channel_set_name() > New Channel OpenZAP/1:1/919999999 [7c60cfea-e341-11dd-acde-9740a65ca868] > 2009-01-15 21:17:09 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing > OpenZAP->919999999 in context default > 2009-01-15 21:17:09 [NOTICE] switch_core_state_machine.c:168 > switch_core_standard_on_execute() Hangup OpenZAP/1:1/919999999 [CS_EXECUTE] > [NORMAL_CLEARING] > 2009-01-15 21:17:09 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 15 (OpenZAP/1:1/919999999 ) Ended > 2009-01-15 21:17:09 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel OpenZAP/1:1/919999999 > [CS_HANGUP] > > Someone knows what's happening? > > Thank you. > > On Wed, Jan 14, 2009 at 4:42 PM, Jon Radel wrote: >> >> Tom?s wrote: >> > Hi, >> > >> > Anthony, I think that's my problem, when I receive a call from the PSTN, >> > FS receive number 1 instead of my house number and I don't know why. >> >> If you use SIP trunking or something like an ISDN-PRI line, the number >> the call is to is delivered as part of the signaling, which is only way >> to make use of many phone numbers on a single physical circuit or >> connection. When you put a POTS line into an FXO port, there is no such >> information provided, as there is only one number on the line. (Leaving >> aside various schemes found in some countries such as using different >> ring patterns to indicate different numbers having been called.) >> >> So, as Anthony keeps pointing out, if you want FS to know the number of >> the line plugged into the FXO port, you have to configure it yourself. >> >> --Jon Radel >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From R.Kloosterman at mtel.nl Thu Jan 15 09:01:17 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Thu, 15 Jan 2009 18:01:17 +0100 Subject: [Freeswitch-users] VMWare voice quality In-Reply-To: References: Message-ID: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? Thanks, Remko -----Oorspronkelijk bericht----- Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Ken Rice Verzonden: donderdag 15 januari 2009 5:05 Aan: freeswitch-users at lists.freeswitch.org; freeswitch-dev at lists.freeswitch.org Onderwerp: [Freeswitch-users] Announcing the FreeSWITCH Technology PreviewVMWare Appliance. Hey guys, I'm not trying to start 1 a day releases, Things just happened to fall that way... FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac. Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed and operational. See /etc/motd on the running image for all the good information. We'll be unvailing a wiki page for this shortly. For now you can get the head start by downloading this at http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip Have fun guys! Ken _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From R.Kloosterman at mtel.nl Thu Jan 15 09:08:24 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Thu, 15 Jan 2009 18:08:24 +0100 Subject: [Freeswitch-users] Vmware voice quality Message-ID: <11372C8B9E603F4FACDE6AB18256DEC601479A7F@srvmtel.office.mtel.nl> Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? Thanks, Remko -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/e3d5c78c/attachment.html From krice at suspicious.org Thu Jan 15 09:15:28 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 15 Jan 2009 11:15:28 -0600 Subject: [Freeswitch-users] VMWare voice quality In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> Message-ID: On 1/15/09 11:01 AM, "Remko Kloosterman" wrote: > Hello Ken, hello all, > > I just read about the FreeSWITCH VMware applicance. I'm curious about > your experiences with the audio quality on VMWare, so here's a new > thread. > > I've installed freeswitch on VMware Server for Windows. The IVR audio > always plays choppy, while the server itself has no performance issues. > The same poor voice quality also goes for Asterisk or Yate, even on a > very fast VMware ESX system. > > Did you experience the same and/or do you have pointers on how to > troubleshoot and fix this? There is a high resolution timer you need to enable on vmware... I'm not familiar enuff with all the versions of vmware to advise there that switch is, but they have a couple of articles on it in their knowledge base From msc at freeswitch.org Thu Jan 15 09:32:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 15 Jan 2009 09:32:19 -0800 Subject: [Freeswitch-users] VMWare voice quality In-Reply-To: References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> Message-ID: <87f2f3b90901150932r2d7bf58cxd4d63b30e464e48f@mail.gmail.com> If anyone figures this out please post it to this thread. I am working on a wiki page for the VMWare appliance and I would like to be able to inform people on how to handle this situation. Also, IIUC, those running VMWare Fusion on Macs are not experiencing this, correct? What about those using a hypervisor like ESXi? Any known issues? Thanks, MC On Thu, Jan 15, 2009 at 9:15 AM, Ken Rice wrote: > On 1/15/09 11:01 AM, "Remko Kloosterman" wrote: > >> Hello Ken, hello all, >> >> I just read about the FreeSWITCH VMware applicance. I'm curious about >> your experiences with the audio quality on VMWare, so here's a new >> thread. >> >> I've installed freeswitch on VMware Server for Windows. The IVR audio >> always plays choppy, while the server itself has no performance issues. >> The same poor voice quality also goes for Asterisk or Yate, even on a >> very fast VMware ESX system. >> >> Did you experience the same and/or do you have pointers on how to >> troubleshoot and fix this? > > > There is a high resolution timer you need to enable on vmware... I'm not > familiar enuff with all the versions of vmware to advise there that switch > is, but they have a couple of articles on it in their knowledge base > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Thu Jan 15 09:36:19 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 15 Jan 2009 18:36:19 +0100 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <496F2C73.9030400@ewetel.de> References: <496E2493.2000207@gmx.net> <87f2f3b90901141307m3d4e0e60hce337ad36b121369@mail.gmail.com> <496EF1C2.8020607@gmx.net> <496F1449.6000407@ewetel.de> <496F18C3.3030807@gmx.net> <496F2C73.9030400@ewetel.de> Message-ID: <496F7413.704@gmx.net> Thanks Helmut, I cross-checked with our provider. They use national numbering plan for our lines. So this didn't solve our problem. I also ensured that the local language is DE and ZAP timing is dedicated to span 1. I changed the configs to debug mode for OpenZAP, so I hopefully will get some more info on the next failure. Best regards Peter Helmut Kuper schrieb: > Hi Peter, > > it was simply a change in our TDM Voice Switch. It used a different > numbering plan and we changed it to "national" to get it work with FS > and openzap in Q921/Q931 mode. > > What I still search is a way to configure the numberplan in FS. > > To make it clear: In my case it didn't work from the second FS starts > up. So this differs from your problem. > > > To get an idea what's going on on the TDM link I used a TDM D-Channel > monitoring device and traced the d-channel messages exchanged between FS > and TDM. That should make it easier to see what's wrong when the > problems occur. > But you can also increase FS debug level to debug and trace the Q921 > and Q931 messages in FS console via fs_cli during runtime. You have to > set this in openzap.conf.xml: > > > > > Unfortunately FS doesn't decode the whole Q931 messages, but it shows a > hex representation of the message, so you can manually decode it with > this documents: > > Q.931: http://www.itu.int/rec/T-REC-Q.931-199805-I/en > Q.921: http://www.itu.int/rec/T-REC-Q.921-199709-I/en > > > I think for numberingplan issues you only have to track the Q.931 > messages. > > > The last idea I have to get some light into your problem and to avoid > manually decoding, try to convert FS's q931 hexdump into wiresharks pcap > format. Wireshark should be able to decode it :) > http://wiki.wireshark.org/Q.931 > > Maybe it's a good idea to implement a wireshark export for those > messages in FS. This will make debugging easy and cheap. > > > > Hope it helps a bit. > > > regards > helmut > > Am 15.01.2009 12:06, schrieb Peter P GMX: > > Helmut, > > > can you give me a hint, how you worked around this? > > > Best regards > > Peter > > > Helmut Kuper schrieb: > >> Hi Michael, > >> > >> it must not be the case here, but I had the same error, when incomming > >> calles used a wrong numbering plan resp not the one, FS expected. > >> > >> Just a hint. > >> > >> regards > >> Helmut > >> > >> > >> Am 15.01.2009 09:20, schrieb Peter P GMX: > >>> 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 at lists.freeswitch.org/msg05792.html > >>> 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 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From d at d-man.org Thu Jan 15 09:46:32 2009 From: d at d-man.org (Darren Schreiber) Date: Thu, 15 Jan 2009 09:46:32 -0800 Subject: [Freeswitch-users] VMWare voice quality In-Reply-To: <87f2f3b90901150932r2d7bf58cxd4d63b30e464e48f@mail.gmail.com> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <87f2f3b90901150932r2d7bf58cxd4d63b30e464e48f@mail.gmail.com> Message-ID: <7349AABD5D3049FEB102B92BB6FDC77D@test> I have been running FreeSWITCH on a VM ever since I got involved in the project. It's been almost a year now. I didn't do anything special - it works fine. I get audio problems if I go over 10 or 15 simultaneous calls. This is on the following setup: VMWare Server 1.0.6 and VMWare Server 2.0 (2.0 sucks, btw ) Dell Precision 360 (Desktop) Pentium 4 2.66Ghz 2.5GB RAM (512MB allocated to FS) Fedora Core 8, 2.6.23.1-42.fc8 stock kernel (a bit old) 7.2K 80GB hard drive Yes, fancy machine I have, huh? This is my normal day-to-day workstation as well as my VMWare Server. It works fine, I got occassional missed heartbeat alerts and timer sync notices, but they're rare. - Darren -----Original Message----- From: Michael Collins [mailto:msc at freeswitch.org] Sent: Thursday, January 15, 2009 9:32 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] VMWare voice quality If anyone figures this out please post it to this thread. I am working on a wiki page for the VMWare appliance and I would like to be able to inform people on how to handle this situation. Also, IIUC, those running VMWare Fusion on Macs are not experiencing this, correct? What about those using a hypervisor like ESXi? Any known issues? Thanks, MC On Thu, Jan 15, 2009 at 9:15 AM, Ken Rice wrote: > On 1/15/09 11:01 AM, "Remko Kloosterman" wrote: > >> Hello Ken, hello all, >> >> I just read about the FreeSWITCH VMware applicance. I'm curious about >> your experiences with the audio quality on VMWare, so here's a new >> thread. >> >> I've installed freeswitch on VMware Server for Windows. The IVR audio >> always plays choppy, while the server itself has no performance issues. >> The same poor voice quality also goes for Asterisk or Yate, even on a >> very fast VMware ESX system. >> >> Did you experience the same and/or do you have pointers on how to >> troubleshoot and fix this? > > > There is a high resolution timer you need to enable on vmware... I'm > not familiar enuff with all the versions of vmware to advise there > that switch is, but they have a couple of articles on it in their > knowledge base > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ajlong at worldlink.net Thu Jan 15 10:04:44 2009 From: ajlong at worldlink.net (Adam Long) Date: Thu, 15 Jan 2009 13:04:44 -0500 Subject: [Freeswitch-users] Using mod_managed Linux/Mono In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670235BBB97F@mse17be1.mse17.exchange.ms> References: <00e401c97699$b4935c80$1dba1580$@net> <6E8D2069C08AA84A83D336E996AE4C670235BBB97F@mse17be1.mse17.exchange.ms> Message-ID: <014801c9773b$cfb03fe0$6f10bfa0$@net> Thanks Michael, that did get me a little further. I renamed mod_managed_lib.dll to FreeSWITCH.Managed.dll and that definitely had an effect. but now when I attempt to > load mod_managed FreeSwitch core dumps now. I have tried mono 2.2 and mono 2.0.1 I am running . CentOS 5.2 x86 32bit [root at sipcore-alpha conf]# uname -a Linux sipcore-alpha 2.6.18-92.el5 #1 SMP Tue Jun 10 18:49:47 EDT 2008 i686 athlon i386 GNU/Linux I attached the full output from the console in the txt doc attached above. I'm wondering if the problem is specific to this flavor of linux. perhaps more specifically the kernel. -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Giagnocavo Sent: Thursday, January 15, 2009 12:34 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Using mod_managed Linux/Mono 2.02 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long Sent: Wednesday, January 14, 2009 3:45 PM To: freeswitch-users at 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 at 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 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/e2bc8cc6/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: freeswitch_loadmanaged.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/e2bc8cc6/attachment-0001.txt From kokoska.rokoska at post.cz Thu Jan 15 10:29:08 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Thu, 15 Jan 2009 19:29:08 +0100 Subject: [Freeswitch-users] mod_sofia: NAT-ping & RPID bounties In-Reply-To: <496F60C2.4080107@freeswitch.org> References: <496E0D3C.9050408@post.cz> <496F4C50.3000101@freeswitch.org> <496F5CBC.4000901@post.cz> <496F60C2.4080107@freeswitch.org> Message-ID: <496F8074.3050809@post.cz> Raymond Chandler napsal(a): > kokoska rokoska wrote: >> >> Raymond Chandler napsal(a): >> >>> kokoska rokoska wrote: >>> >>>> Hi all, >>>> >>>> I have just post two bounties >>>> >>> Where did you post these bounties? >>> >> >> I have posted them to the Boutny wiki page (at the bottom of the page): >> http://wiki.freeswitch.org/wiki/Bounty >> >> >> >>> We've started moving bounties away >>> from the wiki and adding them to jira instead. ( so that progress can be >>> followed more closely ) >>> >>> >> >> OK. Should I move my bounties or you'll be so kind and move them all >> (including mine)? :) >> > anyone still interested in a bounty posted, should move it to jira and > remove it from the wiki.... most of the ones on the wiki have been done > already, iirc > > http://jira.freeswitch.org/browse/BOUNTY > OK, I do it ASAP :-) Best regards, kokoska.rokoska From klaus.teller at gmx.net Thu Jan 15 10:55:02 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Thu, 15 Jan 2009 19:55:02 +0100 Subject: [Freeswitch-users] waitForAnswer on the Socket Interface Message-ID: <20090115185502.242820@gmx.net> Hi, Can somebody tell me how to achieve the same behavuior as session.waitForAnswer via the socket interface? That is, when i call a device, i want to block until the call is completely answered (not just early media). Thanks, Klaus. -- Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a From brian at freeswitch.org Thu Jan 15 11:03:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 13:03:56 -0600 Subject: [Freeswitch-users] waitForAnswer on the Socket Interface In-Reply-To: <20090115185502.242820@gmx.net> References: <20090115185502.242820@gmx.net> Message-ID: Then originate the call with {ignore_early_media=true}sofia/blah/blah, It will not return till its actually answered. /b On Jan 15, 2009, at 12:55 PM, Klaus Teller wrote: > Hi, > > Can somebody tell me how to achieve the same behavuior as > session.waitForAnswer via the socket interface? > > That is, when i call a device, i want to block until the call is > completely answered (not just early media). > > Thanks, > Klaus. > -- From msc at freeswitch.org Thu Jan 15 11:06:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 15 Jan 2009 11:06:57 -0800 Subject: [Freeswitch-users] waitForAnswer on the Socket Interface In-Reply-To: <20090115185502.242820@gmx.net> References: <20090115185502.242820@gmx.net> Message-ID: <87f2f3b90901151106s3a56b5d3raf4c8aeb7defe924@mail.gmail.com> Klaus, What is your dialstring? If you ignore_early_media=true then I believe it will have the same net effect, but it would be good to know exactly what you're hoping to accomplish. -MC On Thu, Jan 15, 2009 at 10:55 AM, Klaus Teller wrote: > Hi, > > Can somebody tell me how to achieve the same behavuior as session.waitForAnswer via the socket interface? > > That is, when i call a device, i want to block until the call is completely answered (not just early media). > > Thanks, > Klaus. > -- > Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL > f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From testeador01 at gmail.com Thu Jan 15 11:21:54 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 15 Jan 2009 14:21:54 -0500 Subject: [Freeswitch-users] Problem connecting with gtalk Message-ID: Hello, I'm running fs 1.0.2 on CentOS 5.2 I've been trying to setup my fs to talk with googletalk following the instructions in http://wiki.freeswitch.org/wiki/Mod_dingaling#Sample_Configuration I got the error of TLS not supported so i: INSTALLED: yum install gnutls-devel gnutls REMOVED: rm -f /usr/src/freeswitch-1.0.2/libs/iksemel/.complete rm -f /usr/src/freeswitch-1.0.2/libs/libdingaling/.complete RE-INSTALLED cd /usr/src/freeswitch-1.0.2/ make sure make installall Even after this i keep getting the same error on the console. This is the error: 2009-01-15 14:06:02 [INFO] libdingaling.c:1304 on_log() RECV: ------------------------------------------------------------------------------- 2009-01-15 14:06:02 [DEBUG] libdingaling.c:1175 on_stream() TLS NOT SUPPORTED IN THIS BUILD! 2009-01-15 14:06:02 [INFO] libdingaling.c:1304 on_log() RECV: ------------------------------------------------------------------------------- X-GOOGLE-TOKEN and now when i *shutdown* my fs, the core gets dumped when trying to stop mod_dingaling: 2009-01-15 14:14:21 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_dingaling Segmentation fault Thank you for any help or suggestions you can give me. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/b16ec467/attachment.html From miconda at gmail.com Thu Jan 15 11:24:57 2009 From: miconda at gmail.com (Daniel-Constantin Mierla) Date: Thu, 15 Jan 2009 21:24:57 +0200 Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <191c3a030901130600m46d6348dlb9ab1d859d47b321@mail.gmail.com> References: <21386948.post@talk.nabble.com> <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> <21409759.post@talk.nabble.com> <191c3a030901120822i31e6d40dgdc88dae43d49b05b@mail.gmail.com> <21432304.post@talk.nabble.com> <191c3a030901130600m46d6348dlb9ab1d859d47b321@mail.gmail.com> Message-ID: <496F8D89.70403@gmail.com> On 01/13/2009 04:00 PM, Anthony Minessale wrote: > So I can supply you with 250 thousand lines of C code that make your > application possible. > but you are not willing to show me the silly js code that may be the > cause of your crash? > What security purposes are you kidding? I just need to salute this! I get same silly reasons day by day, everyone wants their issues fixed in no time without proper (any) feedback. Maybe we should collect and build a top of such reasons... Cheers, Daniel > just rename any sensitive data to something else or stop using js > because without seeing the script code that's all I can tell you as > the solution to your problem. -- Daniel-Constantin Mierla http://www.asipto.com From brian at freeswitch.org Thu Jan 15 11:26:26 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 13:26:26 -0600 Subject: [Freeswitch-users] Problem connecting with gtalk In-Reply-To: References: Message-ID: <1B25553A-B877-4215-A003-D02F8EFD4723@freeswitch.org> install gnutls and dev packages and reconfigure/recompile /b On Jan 15, 2009, at 1:21 PM, Milena wrote: > Hello, > > I'm running fs 1.0.2 on CentOS 5.2 > > I've been trying to setup my fs to talk with googletalk following > the instructions in http://wiki.freeswitch.org/wiki/Mod_dingaling#Sample_Configuration > > I got the error of TLS not supported so i: > INSTALLED: > yum install gnutls-devel gnutls > REMOVED: > rm -f /usr/src/freeswitch-1.0.2/libs/iksemel/.complete > rm -f /usr/src/freeswitch-1.0.2/libs/libdingaling/.complete > RE-INSTALLED > cd /usr/src/freeswitch-1.0.2/ > make sure > make installall > > Even after this i keep getting the same error on the console. > This is the error: > 2009-01-15 14:06:02 [INFO] libdingaling.c:1304 on_log() RECV: > > ------------------------------------------------------------------------------- > xmlns:stream="http://etherx.jabber.org/streams" > xmlns="jabber:client"> > > > 2009-01-15 14:06:02 [DEBUG] libdingaling.c:1175 on_stream() TLS NOT > SUPPORTED IN THIS BUILD! > > 2009-01-15 14:06:02 [INFO] libdingaling.c:1304 on_log() RECV: > ------------------------------------------------------------------------------- > > > > > > > > X-GOOGLE-TOKEN > > > > and now when i *shutdown* my fs, the core gets dumped when trying to > stop mod_dingaling: > 2009-01-15 14:14:21 [CONSOLE] switch_loadable_module.c:1231 > do_shutdown() Stopping: mod_dingaling > Segmentation fault > > Thank you for any help or suggestions you can give me. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/286ed63b/attachment.html From klaus.teller at gmx.net Thu Jan 15 11:27:54 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Thu, 15 Jan 2009 20:27:54 +0100 Subject: [Freeswitch-users] waitForAnswer on the Socket Interface In-Reply-To: <87f2f3b90901151106s3a56b5d3raf4c8aeb7defe924@mail.gmail.com> References: <20090115185502.242820@gmx.net> <87f2f3b90901151106s3a56b5d3raf4c8aeb7defe924@mail.gmail.com> Message-ID: <20090115192754.293190@gmx.net> Thanks folks! ignore_early_media=true solves my problem. The dialstring was just sofia/gateway/blah/blah. Klaus. -------- Original-Nachricht -------- > Datum: Thu, 15 Jan 2009 11:06:57 -0800 > Von: "Michael Collins" > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] waitForAnswer on the Socket Interface > Klaus, > > What is your dialstring? If you ignore_early_media=true then I believe > it will have the same net effect, but it would be good to know exactly > what you're hoping to accomplish. > > -MC > > On Thu, Jan 15, 2009 at 10:55 AM, Klaus Teller > wrote: > > Hi, > > > > Can somebody tell me how to achieve the same behavuior as > session.waitForAnswer via the socket interface? > > > > That is, when i call a device, i want to block until the call is > completely answered (not just early media). > > > > Thanks, > > Klaus. > > -- > > Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL > > f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From matthew at matthew.at Thu Jan 15 11:39:02 2009 From: matthew at matthew.at (Matthew Kaufman) Date: Thu, 15 Jan 2009 11:39:02 -0800 Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <496F8D89.70403@gmail.com> References: <21386948.post@talk.nabble.com> <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> <21409759.post@talk.nabble.com> <191c3a030901120822i31e6d40dgdc88dae43d49b05b@mail.gmail.com> <21432304.post@talk.nabble.com> <191c3a030901130600m46d6348dlb9ab1d859d47b321@mail.gmail.com> <496F8D89.70403@gmail.com> Message-ID: <496F90D6.1060006@matthew.at> Daniel-Constantin Mierla wrote: > On 01/13/2009 04:00 PM, Anthony Minessale wrote: > >> So I can supply you with 250 thousand lines of C code that make your >> application possible. >> but you are not willing to show me the silly js code that may be the >> cause of your crash? >> What security purposes are you kidding? >> > I just need to salute this! I get same silly reasons day by day, > everyone wants their issues fixed in no time without proper (any) > feedback. Maybe we should collect and build a top of such reasons... > Just set the bug to UTR and move on. No reason to berate the person who won't supply the script required to reproduce the problem... it just won't get fixed without both a way to reproduce and a developer who cares to dig into finding it. Could you imagine a large software company saying anything other than "you have not supplied enough information for us to reproduce this bug"? Between the time wasted writing a longer response, and the image it creates for clueless users/customers of the developers and the support process, it just isn't worth it. Matthew Kaufman From testeador01 at gmail.com Thu Jan 15 11:39:52 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 15 Jan 2009 14:39:52 -0500 Subject: [Freeswitch-users] Problem connecting with gtalk Message-ID: Hello, Isn't that what I did? if not, what is the right way to "install gnutls and dev packages and reconfigure/recompile" thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/c2690ef9/attachment.html From brian at freeswitch.org Thu Jan 15 12:02:42 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 14:02:42 -0600 Subject: [Freeswitch-users] Problem connecting with gtalk In-Reply-To: References: Message-ID: <67BB8746-B572-4FFA-B57A-D7DB2CE59F0D@freeswitch.org> Well the right way is depends on your distro. Once you have it installed I would ./bootstrap.sh and ./configure again to be safe. /b On Jan 15, 2009, at 1:39 PM, Milena wrote: > Hello, > > Isn't that what I did? > if not, what is the right way to "install gnutls and dev packages > and reconfigure/recompile" > > thank you > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From testeador01 at gmail.com Thu Jan 15 12:04:17 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 15 Jan 2009 15:04:17 -0500 Subject: [Freeswitch-users] Problem connecting with gtalk In-Reply-To: References: Message-ID: Oops, it took me a little while to realize what you meant and why make alone wouldn't work, thank you very much sir, it all works fine now. 2009/1/15 Milena > Hello, > > Isn't that what I did? > if not, what is the right way to "install gnutls and dev packages and > reconfigure/recompile" > > thank you > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/c52d79a2/attachment.html From tharrisone at gmail.com Thu Jan 15 12:03:29 2009 From: tharrisone at gmail.com (Terrance Harris) Date: Thu, 15 Jan 2009 14:03:29 -0600 Subject: [Freeswitch-users] Freeswitch and CELT: Message-ID: Hello, I have recently found out about FS and how great it is. We are trying to use FS as a voip client for radio shows. We have been using Trixbox and Skype but Skype isn't getting it done. I have heard about how great the celt codec is but I don't have enough 'skill' to compile both FS and celt in MSVC++. Is there a binary out there that would make my day or a guide? Thanks much! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/e75d0c3a/attachment.html From damin at nacs.net Thu Jan 15 12:10:27 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Thu, 15 Jan 2009 15:10:27 -0500 Subject: [Freeswitch-users] VMWare voice quality In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> Message-ID: <005d01c9774d$4f2016e0$ed6044a0$@net> You'll never fix this. Voice is a latency specific application unless you figure out how to manipulate time. Any virtualization platform is going to provide less timing granularity than raw hardware. > Hello Ken, hello all, > > I just read about the FreeSWITCH VMware applicance. I'm curious about > your experiences with the audio quality on VMWare, so here's a new > thread. > > I've installed freeswitch on VMware Server for Windows. The IVR audio > always plays choppy, while the server itself has no performance issues. > The same poor voice quality also goes for Asterisk or Yate, even on a > very fast VMware ESX system. > > Did you experience the same and/or do you have pointers on how to > troubleshoot and fix this? From anthony.minessale at gmail.com Thu Jan 15 12:10:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 15 Jan 2009 14:10:38 -0600 Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <496F90D6.1060006@matthew.at> References: <21386948.post@talk.nabble.com> <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> <21409759.post@talk.nabble.com> <191c3a030901120822i31e6d40dgdc88dae43d49b05b@mail.gmail.com> <21432304.post@talk.nabble.com> <191c3a030901130600m46d6348dlb9ab1d859d47b321@mail.gmail.com> <496F8D89.70403@gmail.com> <496F90D6.1060006@matthew.at> Message-ID: <191c3a030901151210n72793a76s4af17ce1be701bc4@mail.gmail.com> Matthew, I am not berating him, I am trying to convince him to give me the script that causes his crash. It seems ridiculous to me that he should be worried about what I will do with his js code when I am clearly only interested in finding out what causes his issue. And there is irony for him to think that I can write FS itself in C then need his js code for anything i want to accomplish. He can easily remove any sensitive information from the script before supplying it. Why exactly are you so abrasive in our community. We just spent like 3 days trying to help you with an issue didn't we? And it was not even a problem in FS itself. Also, why are you comparing us to a company we are not a company we are an open source project and everything is free, what exactly are you expecting? We all spend most of our day helping people here including you on multiple occasions and you seem to repeatedly criticize us for who knows why. I am not mad about your comment I just don't get it. On Thu, Jan 15, 2009 at 1:39 PM, Matthew Kaufman wrote: > Daniel-Constantin Mierla wrote: > > On 01/13/2009 04:00 PM, Anthony Minessale wrote: > > > >> So I can supply you with 250 thousand lines of C code that make your > >> application possible. > >> but you are not willing to show me the silly js code that may be the > >> cause of your crash? > >> What security purposes are you kidding? > >> > > I just need to salute this! I get same silly reasons day by day, > > everyone wants their issues fixed in no time without proper (any) > > feedback. Maybe we should collect and build a top of such reasons... > > > Just set the bug to UTR and move on. No reason to berate the person who > won't supply the script required to reproduce the problem... it just > won't get fixed without both a way to reproduce and a developer who > cares to dig into finding it. > > Could you imagine a large software company saying anything other than > "you have not supplied enough information for us to reproduce this bug"? > Between the time wasted writing a longer response, and the image it > creates for clueless users/customers of the developers and the support > process, it just isn't worth it. > > Matthew Kaufman > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/b334f3cb/attachment.html From damin at nacs.net Thu Jan 15 12:12:34 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Thu, 15 Jan 2009 15:12:34 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> Message-ID: <006d01c9774d$9a94cd00$cfbe6700$@net> That won't eliminate the problem. Just reduce the possibility of it happening. Trust me... I've got a large ESX infrastructure, and there is no way that a software based Voice platform is going to provide skip free audio in a virtualized environment. > -----Original Message----- > From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch- > dev-bounces at lists.freeswitch.org] On Behalf Of Ken Rice > Sent: Thursday, January 15, 2009 12:15 PM > To: freeswitch-users at lists.freeswitch.org; Remko Kloosterman; > freeswitch-dev at lists.freeswitch.org > Subject: Re: [Freeswitch-dev] [Freeswitch-users] VMWare voice quality > > On 1/15/09 11:01 AM, "Remko Kloosterman" wrote: > > > Hello Ken, hello all, > > > > I just read about the FreeSWITCH VMware applicance. I'm curious about > > your experiences with the audio quality on VMWare, so here's a new > > thread. > > > > I've installed freeswitch on VMware Server for Windows. The IVR audio > > always plays choppy, while the server itself has no performance > issues. > > The same poor voice quality also goes for Asterisk or Yate, even on a > > very fast VMware ESX system. > > > > Did you experience the same and/or do you have pointers on how to > > troubleshoot and fix this? > > > There is a high resolution timer you need to enable on vmware... I'm > not > familiar enuff with all the versions of vmware to advise there that > switch > is, but they have a couple of articles on it in their knowledge base > > > > _______________________________________________ > 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 > > -- > This message has been scanned for viruses and > dangerous content by N2Net Mailshield, and is > believed to be clean. From Prometheus001 at gmx.net Thu Jan 15 12:12:41 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 15 Jan 2009 21:12:41 +0100 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <496F7413.704@gmx.net> References: <496E2493.2000207@gmx.net> <87f2f3b90901141307m3d4e0e60hce337ad36b121369@mail.gmail.com> <496EF1C2.8020607@gmx.net> <496F1449.6000407@ewetel.de> <496F18C3.3030807@gmx.net> <496F2C73.9030400@ewetel.de> <496F7413.704@gmx.net> Message-ID: <496F98B9.7040403@gmx.net> I did some more tests. When I sequentially setup calls (only one simultaneous call at a time), it works for hundreds of calls. As soon as I setup 2 calls in parallel ist fails aber a number of calls. Please find another debug ouput (now with Q.921 debug also). The log starts with the latest hangup of a successfull call. After this one I receive a "2009-01-15 20:26:46 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received Release with no matching channel 0" and later "2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse error [-3012] [Q931E_INVALID_CRV]" Is there anyone to fix it? May I donate some money for fixing that? Best regards Peter Debug: 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame ----------------- Q.921 Packet [Outgoing] --------------- SAPI: 0, TEI: 0, C/R: Command (0) Type: S Frame, SV: RR (Receive Ready) P/F: 0, N(R): 81 [V(A): 80, V(R): 81, V(S): 80] Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] ---------------------------------------------- 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:813 state_advance() 2:3 STATE [TERMINATING] 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:1121 state_advance() Terminating: Direction Inbound 2009-01-15 20:26:44 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() got clear channel sig [STOP] 2009-01-15 20:26:44 [NOTICE] mod_openzap.c:1437 on_clear_channel_signal() Hangup OpenZAP/2:3/21658519 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-15 20:26:44 [DEBUG] switch_channel.c:1513 switch_channel_perform_hangup() Send signal OpenZAP/2:3/21658519 [KILL] 2009-01-15 20:26:44 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:3/21658519 [BREAK] 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Receiving message from Layer4 (size: 184, type: 77) 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Sending message to Q.921 (size: 184) 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Creating Q.931 Message Header: ProtDisc 8 (0x8), CRV 126 (0x7e), CRVflag: 1 (0x1), MesType: 77 (0x4d) 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 -------------------------------------------------------------------------------- [08 02 80 7e 4d] 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Got frame from Q.931, type: 4, tei: 0, size: 9 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Enqueueing I frame for TEI 0 [0] 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame ----------------- Q.921 Packet [Outgoing] --------------- SAPI: 0, TEI: 0, C/R: Command (0) Type: I Frame P/F: 0, N(S): 80, N(R): 81 [V(A): 80, V(R): 81, V(S): 80] Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] ---------------------------------------------- 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T200 (timeout: 1000 msecs) started for TEI 0 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T203 stopped for TEI 0 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Q931Rx43 return code: 1 2009-01-15 20:26:44 [DEBUG] mod_event_socket.c:1922 listener_run() Session complete, waiting for children 2009-01-15 20:26:44 [DEBUG] mod_event_socket.c:1946 listener_run() Connection Closed 2009-01-15 20:26:44 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (OpenZAP/2:3/21658519) State EXECUTE going to sleep 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:3/21658519) Running State Change CS_HANGUP 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:3/21658519) State HANGUP 2009-01-15 20:26:44 [DEBUG] mod_openzap.c:472 channel_on_hangup() OpenZAP/2:3/21658519 CHANNEL HANGUP 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/2:3/21658519 Standard HANGUP, cause: NORMAL_CLEARING 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:3/21658519) State HANGUP going to sleep 2009-01-15 20:26:44 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 251 (OpenZAP/2:3/21658519) Locked, Waiting on external entities 2009-01-15 20:26:44 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 251 (OpenZAP/2:3/21658519) Ended 2009-01-15 20:26:44 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel OpenZAP/2:3/21658519 [CS_HANGUP] 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() New packet received (4 bytes) ----------------- Q.921 Packet [Incoming] --------------- SAPI: 0, TEI: 0, C/R: Response (0) Type: S Frame, SV: RR (Receive Ready) P/F: 0, N(R): 81 [V(A): 80, V(R): 81, V(S): 81] Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] ---------------------------------------------- 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T200 stopped for TEI 0 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T203 (timeout: 10000 msecs) restarted for TEI 0 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() New packet received (9 bytes) ----------------- Q.921 Packet [Incoming] --------------- SAPI: 0, TEI: 0, C/R: Command (1) Type: I Frame P/F: 0, N(S): 81, N(R): 81 [V(A): 81, V(R): 81, V(S): 81] Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] ---------------------------------------------- 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 -------------------------------------------------------------------------------- [08 02 00 7e 5a] 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Received message from Q.921 (ind 4, tei 0, size 9) MesType: 90, CRVFlag 0 (Originator), CRV 126 (Dialect: 0) 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Sending message to Layer4 (size: 103) 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[5a] Size:[103] CRV: 126 (0x7e, CTX: Originator) 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan e0020270 (2:3) source isdn_data->channels_remote_crv[0x7e] 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() Changing state on 2:3 from TERMINATING to DOWN 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame ----------------- Q.921 Packet [Outgoing] --------------- SAPI: 0, TEI: 0, C/R: Response (1) Type: S Frame, SV: RR (Receive Ready) P/F: 0, N(R): 82 [V(A): 81, V(R): 82, V(S): 81] Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] ---------------------------------------------- 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T200 stopped for TEI 0 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T203 (timeout: 10000 msecs) restarted for TEI 0 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame ----------------- Q.921 Packet [Outgoing] --------------- SAPI: 0, TEI: 0, C/R: Command (0) Type: S Frame, SV: RR (Receive Ready) P/F: 0, N(R): 82 [V(A): 81, V(R): 82, V(S): 81] Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] ---------------------------------------------- 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:813 state_advance() 2:3 STATE [DOWN] 2009-01-15 20:26:44 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 2:3 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() New packet received (16 bytes) ----------------- Q.921 Packet [Incoming] --------------- SAPI: 0, TEI: 0, C/R: Command (1) Type: I Frame P/F: 0, N(S): 82, N(R): 81 [V(A): 81, V(R): 82, V(S): 81] Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] ---------------------------------------------- 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 12 -------------------------------------------------------------------------------- [08 02 00 7d 4d 08 05 82 e6 33 30 33] 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Received message from Q.921 (ind 4, tei 0, size 16) MesType: 77, CRVFlag 0 (Originator), CRV 125 (Dialect: 0) 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Sending message to Layer4 (size: 110) 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[4d] Size:[110] CRV: 125 (0x7d, CTX: Originator) 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0x7d] 2009-01-15 20:26:46 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received Release with no matching channel 0 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Sending message to Q.921 (size: 110) 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Creating Q.931 Message Header: ProtDisc 8 (0x8), CRV 125 (0x7d), CRVflag: 1 (0x1), MesType: 90 (0x5a) 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse error [-3012] [Q931E_INVALID_CRV] 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() Sending frame ----------------- Q.921 Packet [Outgoing] --------------- SAPI: 0, TEI: 0, C/R: Response (1) Type: S Frame, SV: RR (Receive Ready) P/F: 0, N(R): 83 [V(A): 81, V(R): 83, V(S): 81] Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] ---------------------------------------------- 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() T200 stopped for TEI 0 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() T203 (timeout: 10000 msecs) restarted for TEI 0 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() Sending frame ----------------- Q.921 Packet [Outgoing] --------------- SAPI: 0, TEI: 0, C/R: Command (0) Type: S Frame, SV: RR (Receive Ready) P/F: 0, N(R): 83 [V(A): 81, V(R): 83, V(S): 81] Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] ---------------------------------------------- Peter P GMX schrieb: > Thanks Helmut, > > I cross-checked with our provider. They use national numbering plan for > our lines. So this didn't solve our problem. > I also ensured that the local language is DE and ZAP timing is dedicated > to span 1. > > I changed the configs to debug mode for OpenZAP, so I hopefully will get > some more info on the next failure. > > Best regards > Peter > > Helmut Kuper schrieb: > >> Hi Peter, >> >> it was simply a change in our TDM Voice Switch. It used a different >> numbering plan and we changed it to "national" to get it work with FS >> and openzap in Q921/Q931 mode. >> >> What I still search is a way to configure the numberplan in FS. >> >> To make it clear: In my case it didn't work from the second FS starts >> up. So this differs from your problem. >> >> >> To get an idea what's going on on the TDM link I used a TDM D-Channel >> monitoring device and traced the d-channel messages exchanged between FS >> and TDM. That should make it easier to see what's wrong when the >> problems occur. >> But you can also increase FS debug level to debug and trace the Q921 >> and Q931 messages in FS console via fs_cli during runtime. You have to >> set this in openzap.conf.xml: >> >> >> >> >> Unfortunately FS doesn't decode the whole Q931 messages, but it shows a >> hex representation of the message, so you can manually decode it with >> this documents: >> >> Q.931: http://www.itu.int/rec/T-REC-Q.931-199805-I/en >> Q.921: http://www.itu.int/rec/T-REC-Q.921-199709-I/en >> >> >> I think for numberingplan issues you only have to track the Q.931 >> messages. >> >> >> The last idea I have to get some light into your problem and to avoid >> manually decoding, try to convert FS's q931 hexdump into wiresharks pcap >> format. Wireshark should be able to decode it :) >> http://wiki.wireshark.org/Q.931 >> >> Maybe it's a good idea to implement a wireshark export for those >> messages in FS. This will make debugging easy and cheap. >> >> >> >> Hope it helps a bit. >> >> >> regards >> helmut >> >> Am 15.01.2009 12:06, schrieb Peter P GMX: >> >>> Helmut, >>> >>> can you give me a hint, how you worked around this? >>> >>> Best regards >>> Peter >>> >>> Helmut Kuper schrieb: >>> >>>> Hi Michael, >>>> >>>> it must not be the case here, but I had the same error, when incomming >>>> calles used a wrong numbering plan resp not the one, FS expected. >>>> >>>> Just a hint. >>>> >>>> regards >>>> Helmut >>>> >>>> >>>> Am 15.01.2009 09:20, schrieb Peter P GMX: >>>> >>>>> 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 at lists.freeswitch.org/msg05792.html >> >>>>> 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 >>>>>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From chavpaskov at shaw.ca Thu Jan 15 12:23:23 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Thu, 15 Jan 2009 12:23:23 -0800 Subject: [Freeswitch-users] VMWare voice quality In-Reply-To: <87f2f3b90901150932r2d7bf58cxd4d63b30e464e48f@mail.gmail.com> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <87f2f3b90901150932r2d7bf58cxd4d63b30e464e48f@mail.gmail.com> Message-ID: <496F9B3B.20509@shaw.ca> Michael Collins wrote: > If anyone figures this out please post it to this thread. I am working > on a wiki page for the VMWare appliance and I would like to be able to > inform people on how to handle this situation. > > Also, IIUC, those running VMWare Fusion on Macs are not experiencing > this, correct? What about those using a hypervisor like ESXi? Any > known issues? > > Thanks, > MC > > On Thu, Jan 15, 2009 at 9:15 AM, Ken Rice wrote: > >> On 1/15/09 11:01 AM, "Remko Kloosterman" wrote: >> >> >>> Hello Ken, hello all, >>> >>> I just read about the FreeSWITCH VMware applicance. I'm curious about >>> your experiences with the audio quality on VMWare, so here's a new >>> thread. >>> >>> I've installed freeswitch on VMware Server for Windows. The IVR audio >>> always plays choppy, while the server itself has no performance issues. >>> The same poor voice quality also goes for Asterisk or Yate, even on a >>> very fast VMware ESX system. >>> >>> Did you experience the same and/or do you have pointers on how to >>> troubleshoot and fix this? >>> >> There is a high resolution timer you need to enable on vmware... I'm not >> familiar enuff with all the versions of vmware to advise there that switch >> is, but they have a couple of articles on it in their knowledge base >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Hi All, I'm using freeswitch in production environment running on ESXi . I have no issues with voice /probably because simply i leave the media to flow between endpoints/ . Performance is amazing and i'd recommend this setup to everybody. it is important though when you set your VM on ESXi to set in advance the number of CPUs. Changing # of CPUs later might affect your performance. My recommendation is NOT to use VMWARE server on top of other OS. ESXi as hipervisor is linux in its core that provides you with enough access to the HW and nothing more so the overhead is as minimal as possible /while this is not the case fro VMware server - it needs underlaying OS and so on/. I hope this info helps. If anybody is interested i'd be glad to share me experience on his matter. Best Regards Chav From mike at jerris.com Thu Jan 15 12:30:26 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 15 Jan 2009 15:30:26 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: <006d01c9774d$9a94cd00$cfbe6700$@net> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <006d01c9774d$9a94cd00$cfbe6700$@net> Message-ID: <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com> To the contrary, we have had quite good results in virtualized environments and you don't really need timing that is that accurate to make it work. We work quite well on amazon EC2 for example. There are 2 issues I know about with vmware, 1 is you need to set a setting on the host to extend somewhat sane clocks being available, the second is I have seen issues with the bridged network adapter actually doubling up all packets causing very strange issues, I suggest not using bridged networking if you experience this. Mike On Jan 15, 2009, at 3:12 PM, Gregory Boehnlein wrote: > That won't eliminate the problem. Just reduce the possibility of it > happening. > > Trust me... I've got a large ESX infrastructure, and there is no way > that a > software based Voice platform is going to provide skip free audio in a > virtualized environment. > >> -----Original Message----- >> From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch- >> dev-bounces at lists.freeswitch.org] On Behalf Of Ken Rice >> Sent: Thursday, January 15, 2009 12:15 PM >> To: freeswitch-users at lists.freeswitch.org; Remko Kloosterman; >> freeswitch-dev at lists.freeswitch.org >> Subject: Re: [Freeswitch-dev] [Freeswitch-users] VMWare voice quality >> >> On 1/15/09 11:01 AM, "Remko Kloosterman" >> wrote: >> >>> Hello Ken, hello all, >>> >>> I just read about the FreeSWITCH VMware applicance. I'm curious >>> about >>> your experiences with the audio quality on VMWare, so here's a new >>> thread. >>> >>> I've installed freeswitch on VMware Server for Windows. The IVR >>> audio >>> always plays choppy, while the server itself has no performance >> issues. >>> The same poor voice quality also goes for Asterisk or Yate, even >>> on a >>> very fast VMware ESX system. >>> >>> Did you experience the same and/or do you have pointers on how to >>> troubleshoot and fix this? >> >> >> There is a high resolution timer you need to enable on vmware... I'm >> not >> familiar enuff with all the versions of vmware to advise there that >> switch >> is, but they have a couple of articles on it in their knowledge base >> >> From damin at nacs.net Thu Jan 15 12:37:15 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Thu, 15 Jan 2009 15:37:15 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <006d01c9774d$9a94cd00$cfbe6700$@net> <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com> Message-ID: <009301c97751$0d04e250$270ea6f0$@net> > To the contrary, we have had quite good results in virtualized > environments and you don't really need timing that is that accurate to > make it work. If you don't handle RTP, I'm sure it is amazing. However, if you have to do voicemail, stream audio from the server or do any kind of actual time/latency/jitter sensitive processing, I don't care how much you tune your hypervisor, it's never going to scale. > We work quite well on amazon EC2 for example. There > are 2 issues I know about with vmware, 1 is you need to set a setting > on the host to extend somewhat sane clocks being available, the second > is I have seen issues with the bridged network adapter actually > doubling up all packets causing very strange issues, I suggest not > using bridged networking if you experience this. I've not seen this behavior on Vmware ESX 3.5u2. Maybe an issue on Vmware Server or Workstation? From Prometheus001 at gmx.net Thu Jan 15 12:38:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 15 Jan 2009 21:38:25 +0100 Subject: [Freeswitch-users] VMWare voice quality In-Reply-To: <005d01c9774d$4f2016e0$ed6044a0$@net> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <005d01c9774d$4f2016e0$ed6044a0$@net> Message-ID: <496F9EC1.3020406@gmx.net> Hello all, let me also give some experience from the VirtualBox side (Community Version). Host machine ========== AMD X2 64 3800 with 8GB of RAM OS is a generic Debian 4.0R5 with Kernel 2.6.18-6-amd64 No special parameters in the Kernel. Started with VirtualBox 1.5 and now on 2.0.x Client machine (freeswitch) =========== Ubuntu 8.041 Generic Kernel 2.6.24-18-generic #1 SMP, 1 CPU Experience: =========== A single call produces about 20% CPU load. So this is not usefull for any production environment. I did not discover any dropouts in a normal call between internal and external UAs/gateways since 6 months. So for testing purposes its fine. Voice between User Agents is always fine. Seldomly I hear choppy voice when announcements are played. After some minutes these problems go away. Resume =========== For testing/development purposes, FS on VirtualBox is fine. For any productive environment it's not really usable in our environment. Comparison with Asterisk ================= Asterisk never worked in this environment: Choppe voice between UAs and when playing sound. 100% CPU load on a single call. ========== Best regards Peter Gregory Boehnlein schrieb: > You'll never fix this. Voice is a latency specific application unless you > figure out how to manipulate time. Any virtualization platform is going to > provide less timing granularity than raw hardware. > > >> Hello Ken, hello all, >> >> I just read about the FreeSWITCH VMware applicance. I'm curious about >> your experiences with the audio quality on VMWare, so here's a new >> thread. >> >> I've installed freeswitch on VMware Server for Windows. The IVR audio >> always plays choppy, while the server itself has no performance issues. >> The same poor voice quality also goes for Asterisk or Yate, even on a >> very fast VMware ESX system. >> >> Did you experience the same and/or do you have pointers on how to >> troubleshoot and fix this? >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Jan 15 12:39:46 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 14:39:46 -0600 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: <009301c97751$0d04e250$270ea6f0$@net> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <006d01c9774d$9a94cd00$cfbe6700$@net> <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com> <009301c97751$0d04e250$270ea6f0$@net> Message-ID: We have people running FreeSWITCH in vmware and xen with media and considerable load and it doesn't have a problem. We also work very well inside OpenVZ. /b On Jan 15, 2009, at 2:37 PM, Gregory Boehnlein wrote: > If you don't handle RTP, I'm sure it is amazing. However, if you > have to do > voicemail, stream audio from the server or do any kind of actual > time/latency/jitter sensitive processing, I don't care how much you > tune > your hypervisor, it's never going to scale. From damin at nacs.net Thu Jan 15 12:59:13 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Thu, 15 Jan 2009 15:59:13 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <006d01c9774d$9a94cd00$cfbe6700$@net> <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com> <009301c97751$0d04e250$270ea6f0$@net> Message-ID: <00b701c97754$1f390ac0$5dab2040$@net> > We have people running FreeSWITCH in vmware and xen with media and > considerable load and it doesn't have a problem. We also work very > well inside OpenVZ. I'd be very interested in seeing that, and knowing how it was done. From krice at suspicious.org Thu Jan 15 13:01:45 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 15 Jan 2009 15:01:45 -0600 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: <006d01c9774d$9a94cd00$cfbe6700$@net> Message-ID: Ok if can summarize a little of the intention of releasing this VMWare image. Its really there so you guys can get it and check it out. I personally don't believe in running such services on a virtual machine (too many nightmare stories from the 'day job' from such things) However, for testing and developing applications that ride on top of FreeSWITCH, this is a quick way to get up and running. Remember Voice application especially where you are interacting with the media streams will be affected by latency and jitter much more readily then store and forward things like IRC, Web, eMail and instant messaging. K On 1/15/09 2:12 PM, "Gregory Boehnlein" wrote: > That won't eliminate the problem. Just reduce the possibility of it > happening. > > Trust me... I've got a large ESX infrastructure, and there is no way that a > software based Voice platform is going to provide skip free audio in a > virtualized environment. From brian at freeswitch.org Thu Jan 15 13:02:05 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 15:02:05 -0600 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: <00b701c97754$1f390ac0$5dab2040$@net> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <006d01c9774d$9a94cd00$cfbe6700$@net> <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com> <009301c97751$0d04e250$270ea6f0$@net> <00b701c97754$1f390ac0$5dab2040$@net> Message-ID: On that note the OpenVZ instances could live migrate from box to box without dropping calls and usually had a small acceptable blip in audio. /b On Jan 15, 2009, at 2:59 PM, Gregory Boehnlein wrote: >> We have people running FreeSWITCH in vmware and xen with media and >> considerable load and it doesn't have a problem. We also work very >> well inside OpenVZ. > > I'd be very interested in seeing that, and knowing how it was don From msc at freeswitch.org Thu Jan 15 13:12:13 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 15 Jan 2009 13:12:13 -0800 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <006d01c9774d$9a94cd00$cfbe6700$@net> <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com> <009301c97751$0d04e250$270ea6f0$@net> <00b701c97754$1f390ac0$5dab2040$@net> Message-ID: <185FFF37-E1F8-4CD2-A807-ABFFF52913D5@freeswitch.org> On Jan 15, 2009, at 1:02 PM, Brian West wrote: > On that note the OpenVZ instances could live migrate from box to box > without dropping calls and usually had a small acceptable blip in > audio. > I'd say a small blip is quite acceptable compared to the alternative! -MC > /b > > On Jan 15, 2009, at 2:59 PM, Gregory Boehnlein wrote: > >>> We have people running FreeSWITCH in vmware and xen with media and >>> considerable load and it doesn't have a problem. We also work very >>> well inside OpenVZ. >> >> I'd be very interested in seeing that, and knowing how it was don > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From damin at nacs.net Thu Jan 15 13:18:18 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Thu, 15 Jan 2009 16:18:18 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <006d01c9774d$9a94cd00$cfbe6700$@net> <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com> <009301c97751$0d04e250$270ea6f0$@net> <00b701c97754$1f390ac0$5dab2040$@net> Message-ID: <00d301c97756$c94ba340$5be2e9c0$@net> > On that note the OpenVZ instances could live migrate from box to box > without dropping calls and usually had a small acceptable blip in > audio. OpenVZ is not a hypervisor. It essentially runs all of it's applications natively on the CPU. I would expect that it would work under OpenVZ or other container based (chrooted / jailed setups) well. From klaus.teller at gmx.net Thu Jan 15 13:20:10 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Thu, 15 Jan 2009 22:20:10 +0100 Subject: [Freeswitch-users] No Sound Heared Message-ID: <20090115212010.112080@gmx.net> Hi, Need your help on this. I have the following Javascript statement: session.execute("bridge","sofia/gateway/sip.gafachi.com/someNumber") in a file called gafachiDialout.js Then, i have the following extension in default.xml: When i call this extension (6337), it rings as it should. But then there is NO sound going in either direction. Any idea what i'm doing wrong here? Thanks, Klaus. -- Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a From willbelair at yahoo.com Thu Jan 15 13:42:53 2009 From: willbelair at yahoo.com (Will Smith) Date: Thu, 15 Jan 2009 13:42:53 -0800 (PST) Subject: [Freeswitch-users] No Audio when dial out via gateway Message-ID: <980976.48877.qm@web53608.mail.re2.yahoo.com> Hi, I have successfully installed and configured the FS thanks to the community help. Greatly appreciate all. Now I have some basic error: ? I can dial out from extension 1000 (all default ext) to any number not in the same network. I got the other number rung, and answered, but cannot hear anything from both ends. Strange thing is I can broadcast an audio into the conversation, and both ends can hear the audio, but just cannot talk. ? I just hope that it would be something easy to be pointed out by experienced users. ? Thank you, ? Will -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/e1f76c12/attachment.html From brian at freeswitch.org Thu Jan 15 13:46:38 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 15:46:38 -0600 Subject: [Freeswitch-users] No Audio when dial out via gateway In-Reply-To: <980976.48877.qm@web53608.mail.re2.yahoo.com> References: <980976.48877.qm@web53608.mail.re2.yahoo.com> Message-ID: <35238A38-1070-4D15-A5FD-C09F67B8B4C8@freeswitch.org> did you check our firewall? and various nat settings? /b On Jan 15, 2009, at 3:42 PM, Will Smith wrote: > Hi, > I have successfully installed and configured the FS thanks to the > community help. Greatly appreciate all. > Now I have some basic error: > > I can dial out from extension 1000 (all default ext) to any number > not in the same network. I got the other number rung, and answered, > but cannot hear anything from both ends. Strange thing is I can > broadcast an audio into the conversation, and both ends can hear the > audio, but just cannot talk. > > I just hope that it would be something easy to be pointed out by > experienced users. > > Thank you, > > Will -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/d1472d38/attachment.html From willbelair at yahoo.com Thu Jan 15 13:54:39 2009 From: willbelair at yahoo.com (Will Smith) Date: Thu, 15 Jan 2009 13:54:39 -0800 (PST) Subject: [Freeswitch-users] No Audio when dial out via gateway In-Reply-To: <35238A38-1070-4D15-A5FD-C09F67B8B4C8@freeswitch.org> Message-ID: <110496.46372.qm@web53603.mail.re2.yahoo.com> Thank you so much for responding. Yes, I checked those, everything looks fine, and infact, if the audio stream is blocked by firewall or nat setting, how can I inject the audio file and hear it played on both ends. But as you suggest, I will doublecheck those values. Thanks again --- On Thu, 1/15/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] No Audio when dial out via gateway To: freeswitch-users at lists.freeswitch.org Date: Thursday, January 15, 2009, 1:46 PM did you check our firewall? and various nat settings? /b On Jan 15, 2009, at 3:42 PM, Will Smith wrote: Hi, I have successfully installed and configured the FS thanks to the community help. Greatly appreciate all. Now I have some basic error: ? I can dial out from extension 1000 (all default ext) to any number not in the same network. I got the other number rung, and answered, but cannot hear anything from both ends. Strange thing is I can broadcast an audio into the conversation, and both ends can hear the audio, but just cannot talk. ? I just hope that it would be something easy to be pointed out by experienced users. ? Thank you, ? Will _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/e2090e1f/attachment.html From astmac at stillnewt.org Thu Jan 15 13:54:54 2009 From: astmac at stillnewt.org (Martin Joseph) Date: Thu, 15 Jan 2009 13:54:54 -0800 Subject: [Freeswitch-users] Starting FS on OSX Message-ID: Hello again FreeSwitchers, I have built the 1.02 on 10.4.11(OSX) and had no problems with that. I have never been able to build from the SVN, but that is another story. Now that I have migrated to 1.02 I was wondering if I can get some help on a long standing issue I have with starting FS at boot. I am hoping to use Launchd which is the standard on OSX 10.4 and I attempted to cobble together a script, but haven't had great results. I did search for wiki entries on this, but haven't found any help with it. Ideas? Thanks, Marty From R.Kloosterman at mtel.nl Thu Jan 15 14:02:53 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Thu, 15 Jan 2009 23:02:53 +0100 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: <009301c97751$0d04e250$270ea6f0$@net> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <006d01c9774d$9a94cd00$cfbe6700$@net> <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com> <009301c97751$0d04e250$270ea6f0$@net> Message-ID: <11372C8B9E603F4FACDE6AB18256DEC695A926@srvmtel.office.mtel.nl> Lot's of experience and suggestions here. Thanks. I believe it should be theoretically possible to have blip-free RTP streaming through the appliance. Most windows ethernet drivers allow for QoS packet scheduling. If the VMware network bridge driver honors this and syncs the buffers at 20ms frames (or whatever frame size applies) you should be able to schale up a bit and maintain low jitter. Anyone knows how the VMware network bridge exactly works? -----Oorspronkelijk bericht----- Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Gregory Boehnlein Verzonden: donderdag 15 januari 2009 21:37 Aan: freeswitch-users at lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality > To the contrary, we have had quite good results in virtualized > environments and you don't really need timing that is that accurate to > make it work. If you don't handle RTP, I'm sure it is amazing. However, if you have to do voicemail, stream audio from the server or do any kind of actual time/latency/jitter sensitive processing, I don't care how much you tune your hypervisor, it's never going to scale. > We work quite well on amazon EC2 for example. There are 2 issues I > know about with vmware, 1 is you need to set a setting on the host to > extend somewhat sane clocks being available, the second is I have seen > issues with the bridged network adapter actually doubling up all > packets causing very strange issues, I suggest not using bridged > networking if you experience this. I've not seen this behavior on Vmware ESX 3.5u2. Maybe an issue on Vmware Server or Workstation? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From willbelair at yahoo.com Thu Jan 15 14:06:48 2009 From: willbelair at yahoo.com (Will Smith) Date: Thu, 15 Jan 2009 14:06:48 -0800 (PST) Subject: [Freeswitch-users] No Audio when dial out via gateway In-Reply-To: <35238A38-1070-4D15-A5FD-C09F67B8B4C8@freeswitch.org> Message-ID: <211945.81928.qm@web53606.mail.re2.yahoo.com> I found this: When I call the outside number, first, cannot hear or be heard, then when I put the line on hold, the other party can hear the MOH, and when I switch it back, now we can talk. Something goes wrong here ? --- On Thu, 1/15/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] No Audio when dial out via gateway To: freeswitch-users at lists.freeswitch.org Date: Thursday, January 15, 2009, 1:46 PM did you check our firewall? and various nat settings? /b On Jan 15, 2009, at 3:42 PM, Will Smith wrote: Hi, I have successfully installed and configured the FS thanks to the community help. Greatly appreciate all. Now I have some basic error: ? I can dial out from extension 1000 (all default ext) to any number not in the same network. I got the other number rung, and answered, but cannot hear anything from both ends. Strange thing is I can broadcast an audio into the conversation, and both ends can hear the audio, but just cannot talk. ? I just hope that it would be something easy to be pointed out by experienced users. ? Thank you, ? Will _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/17d20dfb/attachment.html From anthony.minessale at gmail.com Thu Jan 15 14:31:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 15 Jan 2009 16:31:56 -0600 Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent - a solution of sorts. In-Reply-To: <496F1A7B.1060400@novatex.com.au> References: <496EBFDE.2070305@drlake.com.au> <496EDCB6.4020802@novatex.com.au> <496F0D38.5030904@novatex.com.au> <496F1A7B.1060400@novatex.com.au> Message-ID: <191c3a030901151431q139b4563pac9ba24c7fa5891f@mail.gmail.com> open a jira and attach a svn diff and we'll have a look thanks On Thu, Jan 15, 2009 at 5:14 AM, Scott Ellis wrote: > So I decided to hack the code to see if I could just get it to do what I > wanted - assuming some kind of error in the options setting. > > First I changed the state change code to just skip straight to IDLE > > if (!event->channel->ring_count && (event->channel->state == > ZAP_CHANNEL_STATE_DOWN && !zap_test_flag(event->channel, > ZAP_CHANNEL_INTHREAD))) { > // if (zap_test_flag(analog_data, > ZAP_ANALOG_CALLERID)) { > // zap_set_state_locked(event->channel, > ZAP_CHANNEL_STATE_GET_CALLERID); > // } else { > zap_set_state_locked(event->channel, > ZAP_CHANNEL_STATE_IDLE); > // } > event->channel->ring_count = 1; > zap_mutex_unlock(event->channel->mutex); > locked = 0; > > zap_thread_create_detached(zap_analog_channel_run, event->channel); > } else { > event->channel->ring_count++; > } > > So we skip the GET_CALLERID state altogether. > > This generated an illegal state change message cannot go from DOWN to IDLE > > So then changed the code to > > if (!event->channel->ring_count && (event->channel->state == > ZAP_CHANNEL_STATE_DOWN && !zap_test_flag(event->channel, > ZAP_CHANNEL_INTHREAD))) { > // if (zap_test_flag(analog_data, > ZAP_ANALOG_CALLERID)) { > zap_set_state_locked(event->channel, > ZAP_CHANNEL_STATE_GET_CALLERID); > // } else { > zap_set_state_locked(event->channel, > ZAP_CHANNEL_STATE_IDLE); > // } > event->channel->ring_count = 1; > zap_mutex_unlock(event->channel->mutex); > locked = 0; > > zap_thread_create_detached(zap_analog_channel_run, event->channel); > } else { > event->channel->ring_count++; > } > > Allowing the state change to GET_CALLERID, then immediately to IDLE. > > This works perfectly - the call is answered straight away. At the moment > I don't know enough about linux debugging to step through the parameter > code to see why setting get caller ID to false in openzap.conf.xml does > not get passed through, but even if it does the current code will still > run into the illegal state change error. > > 2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [DOWN] > 2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:760 process_event() Changing > state on 1:1 from DOWN to GET_CALLERID > 2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:762 process_event() Changing > state on 1:1 from GET_CALLERID to IDLE > 2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() > ANALOG CHANNEL thread starting. > 2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 1:1 for IDLE > 2009-01-15 21:59:18 [DEBUG] mod_openzap.c:1165 on_fxo_signal() got FXO > sig 1:1 [START] > 2009-01-15 21:59:18 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU > 20ms > 2009-01-15 21:59:18 [DEBUG] mod_openzap.c:1137 zap_channel_from_event() > Connect inbound channel OpenZAP/1:1/1 > 2009-01-15 21:59:18 [NOTICE] switch_channel.c:565 > switch_channel_set_name() New Channel OpenZAP/1:1/1 > [8e2a55c8-e2f3-11dd-adfd-6d934f226ffd] > > Will go and put this into JIRA in the next couple of days. > > Scott > > Scott Ellis wrote: > > After poking around in the code, it looks like if I set > name="enable-callerid" value="false"/> in openzap.conf.xml, it should > > skip the GET_CALLERID state, and I should get the call answered straight > > away. > > > > mod_openzap.c > > > > } else if (!strcasecmp(var, "enable-callerid")) { > > enable_callerid = val; > > > > > > if (zap_configure_span("analog", span, on_analog_signal, > > "tonemap", tonegroup, > > "digit_timeout", &to, > > "max_dialstr", &max, > > "hotline", hotline, > > "enable_callerid", enable_callerid, > > TAG_END) != ZAP_SUCCESS) { > > zap_log(ZAP_LOG_ERROR, "Error starting OpenZAP span > > %d\n", span_id); > > continue; > > } > > > > ozmod_analog.c > > > > else if (!strcasecmp(var, "enable_callerid")) { > > if (!(val = va_arg(ap, char *))) { > > break; > > } > > if (zap_true(val)) { > > flags |= ZAP_ANALOG_CALLERID; > > } else { > > flags &= ~ZAP_ANALOG_CALLERID; > > } > > > > and > > > > case ZAP_OOB_RING_START: > > { > > if (event->channel->type != ZAP_CHAN_TYPE_FXO) { > > zap_log(ZAP_LOG_ERROR, "Cannot get a RING_START event on > > a non-fxo channel, please check your config.\n"); > > zap_set_state_locked(event->channel, > > ZAP_CHANNEL_STATE_DOWN); > > goto end; > > } > > if (!event->channel->ring_count && (event->channel->state == > > ZAP_CHANNEL_STATE_DOWN && !zap_test_flag(event->channel, > > ZAP_CHANNEL_INTHREAD))) { > > if (zap_test_flag(analog_data, ZAP_ANALOG_CALLERID)) { > > zap_set_state_locked(event->channel, > > ZAP_CHANNEL_STATE_GET_CALLERID); > > } else { > > zap_set_state_locked(event->channel, > > ZAP_CHANNEL_STATE_IDLE); > > } > > event->channel->ring_count = 1; > > zap_mutex_unlock(event->channel->mutex); > > locked = 0; > > zap_thread_create_detached(zap_analog_channel_run, > > event->channel); > > } else { > > event->channel->ring_count++; > > } > > } > > break; > > > > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT > > [RING_START][1:1] STATE [DOWN] > > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:760 process_event() Changing > > state on 1:1 from DOWN to GET_CALLERID > > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() > > ANALOG CHANNEL thread starting. > > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > > Executing state handler on 1:1 for GET_CALLERID > > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT > > [RING_START][1:1] STATE [GET_CALLERID] > > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT > > [RING_START][1:1] STATE [GET_CALLERID] > > 2009-01-15 20:19:45 [DEBUG] ozmod_analog.c:744 process_event() EVENT > > [RING_START][1:1] STATE [GET_CALLERID] > > 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT > > [RING_START][1:1] STATE [GET_CALLERID] > > 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT > > [RING_START][1:1] STATE [GET_CALLERID] > > 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT > > [RING_START][1:1] STATE [GET_CALLERID] > > 2009-01-15 20:19:48 [DEBUG] ozmod_analog.c:744 process_event() EVENT > > [RING_START][1:1] STATE [GET_CALLERID] > > 2009-01-15 20:19:49 [DEBUG] ozmod_analog.c:292 zap_analog_channel_run() > > Changing state on 1:1 from GET_CALLERID to IDLE > > 2009-01-15 20:19:49 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > > Executing state handler on 1:1 for IDLE > > 2009-01-15 20:19:49 [DEBUG] mod_openzap.c:1165 on_fxo_signal() got FXO > > sig 1:1 [START] > > > > The code all looks right, but I am not getting what I think should > > happen. Anyone with any ideas? > > > > Scott > > > > Scott Ellis wrote: > > > >> 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 > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/f708b487/attachment-0001.html From krice at suspicious.org Thu Jan 15 15:05:57 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 15 Jan 2009 17:05:57 -0600 Subject: [Freeswitch-users] Starting FS on OSX In-Reply-To: Message-ID: Have you looked at creating a system level startup item in /Library/StartupItems ? Also, to build from source you need the latest DevTools Kit from apple installed. (I don't know if the latest will work w/ 10.4) Ken On 1/15/09 3:54 PM, "Martin Joseph" wrote: > Hello again FreeSwitchers, > > I have built the 1.02 on 10.4.11(OSX) and had no problems with that. > > I have never been able to build from the SVN, but that is another story. > > Now that I have migrated to 1.02 I was wondering if I can get some > help on a long standing issue I have with starting FS at boot. > > I am hoping to use Launchd which is the standard on OSX 10.4 and I > attempted to cobble together a script, but haven't had great results. > > I did search for wiki entries on this, but haven't found any help with > it. > > Ideas? > Thanks, > Marty > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Jan 15 15:10:55 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 15 Jan 2009 18:10:55 -0500 Subject: [Freeswitch-users] Starting FS on OSX In-Reply-To: References: Message-ID: <9B1AAAEE-DF37-442E-ACF5-F2E3078A0991@jerris.com> Your build issue is with your autotools install, I have seen issues if you have ever installed any of the autotools from macports or fink. If you want to build from svn you can run bootstrap on another box (a linux box perhaps) and then tar up that dir and move it to your mac. We pre-bootstrap the release tarballs which is why that is building fine for you. MIke On Jan 15, 2009, at 4:54 PM, Martin Joseph wrote: > Hello again FreeSwitchers, > > I have built the 1.02 on 10.4.11(OSX) and had no problems with that. > > I have never been able to build from the SVN, but that is another > story. > > Now that I have migrated to 1.02 I was wondering if I can get some > help on a long standing issue I have with starting FS at boot. > > I am hoping to use Launchd which is the standard on OSX 10.4 and I > attempted to cobble together a script, but haven't had great results. > > I did search for wiki entries on this, but haven't found any help with > it. > > Ideas? > Thanks, > Marty From mgg at giagnocavo.net Thu Jan 15 15:15:11 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 15 Jan 2009 18:15:11 -0500 Subject: [Freeswitch-users] Freeswitch crashed !!! In-Reply-To: <496F90D6.1060006@matthew.at> References: <21386948.post@talk.nabble.com> <5821FE37-FB04-49CE-AD48-31FDD0489876@freeswitch.org> <21409759.post@talk.nabble.com> <191c3a030901120822i31e6d40dgdc88dae43d49b05b@mail.gmail.com> <21432304.post@talk.nabble.com> <191c3a030901130600m46d6348dlb9ab1d859d47b321@mail.gmail.com> <496F8D89.70403@gmail.com> <496F90D6.1060006@matthew.at> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670235F10C0D@mse17be1.mse17.exchange.ms> >Could you imagine a large software company saying anything other than >"you have not supplied enough information for us to reproduce this bug"? >Between the time wasted writing a longer response, and the image it >creates for clueless users/customers of the developers and the support >process, it just isn't worth it. Well, when FS is a large company and it's a thing on paid support... From ajlong at worldlink.net Thu Jan 15 15:15:34 2009 From: ajlong at worldlink.net (Adam Long) Date: Thu, 15 Jan 2009 18:15:34 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: References: Message-ID: <01f601c97767$2b4dddf0$81e999d0$@net> Hi Tim, I'm having exact same problem, try renaming mod_managed_lib.dll to FreeSWITCH.Managed.dll and then load. Michael confirmed this is supposed to be the case and is building a patch for the Makefile. However, when I do this on my Cent OS 5.2 it now loads successfully but immediately I get a core dump. I'm curious if you will have the same problem or not. Regards, -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim B Sent: Wednesday, January 14, 2009 8:13 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 Got mod_managed compiled and installed. Now it isn't loading. See below... 1) Donwloaded fresh from SVN 2) Compiled... and installed.. OK [root at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig [root at phone2 mod_managed]# make [root at phone2 mod_managed]# make install 3) Added to modules.conf.xml : 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 LiveT: Keep your life in sync. See how it works. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/320613fc/attachment.html From timb0311 at hotmail.com Thu Jan 15 15:17:41 2009 From: timb0311 at hotmail.com (Tim B) Date: Thu, 15 Jan 2009 18:17:41 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: References: Message-ID: 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 at lists.freeswitch.org> Subject: Freeswitch-users Digest, Vol 31, Issue 77> To: freeswitch-users at lists.freeswitch.org> Date: Thu, 15 Jan 2009 00:20:53 -0800> > Send Freeswitch-users mailing list submissions to> freeswitch-users at 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 at lists.freeswitch.org> > You can reach the person managing the list at> freeswitch-users-owner at 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 > Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2> To: > Message-ID: > 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 at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig> [root at phone2 mod_managed]# make> [root at phone2 mod_managed]# make install> > 3) Added to modules.conf.xml :> > > 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_012009> -------------- next part --------------> An HTML attachment was scrubbed...> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/6a5facdc/attachment-0001.html > > ------------------------------> > Message: 2> Date: Thu, 15 Jan 2009 00:34:20 -0500> From: Michael Giagnocavo > Subject: Re: [Freeswitch-users] Using mod_managed Linux/Mono 2.02> To: "freeswitch-users at lists.freeswitch.org"> > Message-ID:> <6E8D2069C08AA84A83D336E996AE4C670235BBB97F at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long> Sent: Wednesday, January 14, 2009 3:45 PM> To: freeswitch-users at 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 at 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 > 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/20090115/73ac27e4/attachment-0001.html > > ------------------------------> > Message: 3> Date: Thu, 15 Jan 2009 17:50:30 +1100> From: Scott Ellis > Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk -> Freeswitch equivalent?> To: "freeswitch-users at lists.freeswitch.org"> > Message-ID: <496EDCB6.4020802 at 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 > Subject: [Freeswitch-users] Country specific tones - how to> contribute?> To: "freeswitch-users at lists.freeswitch.org"> > Message-ID: <496EE2BD.2050102 at 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 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 > Subject: Re: [Freeswitch-users] Country specific tones - how to> contribute?> To: freeswitch-users at lists.freeswitch.org> Message-ID: <20090115072405.GA15789 at jdc.jasonjgw.net>> Content-Type: text/plain; charset=us-ascii> > Scott Ellis 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 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" > Subject: [Freeswitch-users] Changes in PlayAndGetDigits> To: freeswitch-users at lists.freeswitch.org> Message-ID:> <27c25bc40901142343l34a3e99ftecf0df971e8e32f6 at 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 > Subject: Re: [Freeswitch-users] OpenZAP parse error [-3012]> [Q931E_INVALID_CRV]> To: freeswitch-users at lists.freeswitch.org> Message-ID: <496EF1C2.8020607 at gmx.net>> Content-Type: text/plain; charset=ISO-8859-1> > Hello Michael,> > how much $$ are we talking about? I need this issue to be solved quickly> and it's worth to spend some money.> > I've read the following post:> > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05792.html> and have the same symptom with "after hundreds of calls I start to get b> channels that are stuck in states like "TERMINATING" or "HANGUP""> > Best regards> Peter> > Michael Collins schrieb:> > I believe these are all symptoms of something that Stefan is working> > on: better Q931 timers. It's been on the todo list for some time but> > we've had absolutely NOBODY willing to pony up serious $$ 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 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 at 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 at 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 at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/6f0e5a94/attachment-0001.html From ajlong at worldlink.net Thu Jan 15 15:25:26 2009 From: ajlong at worldlink.net (Adam Long) Date: Thu, 15 Jan 2009 18:25:26 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: References: Message-ID: <021001c97768$8c806a60$a5813f20$@net> OK yah i'm experiencing exact same problem here CentOS 5.2 2.6.18-92.el5 #1 SMP Tue Jun 10 18:49:47 EDT 2008 i686 athlon i386 GNU/Linux I too have no problems at all on Windows. I'm going to try a Suse or Ubuntu prebuilt/packaged with Mono 2 I suspect it may be kernel/mono incompatibility. Did you compile mono from tarbal? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim B Sent: Thursday, January 15, 2009 6:18 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 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 at lists.freeswitch.org > Subject: Freeswitch-users Digest, Vol 31, Issue 77 > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 15 Jan 2009 00:20:53 -0800 > > Send Freeswitch-users mailing list submissions to > freeswitch-users at 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 at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at 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 > Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 > To: > Message-ID: > 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 at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig > [root at phone2 mod_managed]# make > [root at phone2 mod_managed]# make install > > 3) Added to modules.conf.xml : > > > 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_0120 09 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/ 6a5facdc/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Thu, 15 Jan 2009 00:34:20 -0500 > From: Michael Giagnocavo > Subject: Re: [Freeswitch-users] Using mod_managed Linux/Mono 2.02 > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: > <6E8D2069C08AA84A83D336E996AE4C670235BBB97F at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long > Sent: Wednesday, January 14, 2009 3:45 PM > To: freeswitch-users at 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 at 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 > 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/ 73ac27e4/attachment-0001.html > > ------------------------------ > > Message: 3 > Date: Thu, 15 Jan 2009 17:50:30 +1100 > From: Scott Ellis > Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - > Freeswitch equivalent? > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: <496EDCB6.4020802 at 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 > Subject: [Freeswitch-users] Country specific tones - how to > contribute? > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: <496EE2BD.2050102 at 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 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 > Subject: Re: [Freeswitch-users] Country specific tones - how to > contribute? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <20090115072405.GA15789 at jdc.jasonjgw.net> > Content-Type: text/plain; charset=us-ascii > > Scott Ellis 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 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 $${variable-name} syntax. > > There's an ITU document referred to on the wiki with the official definitions > of ringback and other tones for various countries. > > > > > ------------------------------ > > Message: 6 > Date: Thu, 15 Jan 2009 15:43:20 +0800 > From: "Juan Backson" > Subject: [Freeswitch-users] Changes in PlayAndGetDigits > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <27c25bc40901142343l34a3e99ftecf0df971e8e32f6 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi, > > Is there a change in the playAndGetDigits api? In the old release, > 11102, my lua script is working but is not working in the latest > release. > The error I am getting is " Error in playAndGetDigits expected 10..10 > args, got 9 ". > > Thanks, > JB > > > > ------------------------------ > > Message: 7 > Date: Thu, 15 Jan 2009 09:20:18 +0100 > From: Peter P GMX > Subject: Re: [Freeswitch-users] OpenZAP parse error [-3012] > [Q931E_INVALID_CRV] > To: freeswitch-users at lists.freeswitch.org > Message-ID: <496EF1C2.8020607 at gmx.net> > Content-Type: text/plain; charset=ISO-8859-1 > > Hello Michael, > > how much $$ 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 at lists.freeswitch.org/msg05792.h tml > 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 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 at 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 at 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 at 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 LiveT: Keep your life in sync. Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/f37b1f92/attachment-0001.html From mgg at giagnocavo.net Thu Jan 15 15:25:40 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 15 Jan 2009 18:25:40 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C670235F10C20@mse17be1.mse17.exchange.ms> OK I'll get a linux machine up and see what's failing. I believe the last time I tested was with Mono 2.0.0. A lot of changes have gone into FS since then too :) -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim B Sent: Thursday, January 15, 2009 4:18 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 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 at lists.freeswitch.org > Subject: Freeswitch-users Digest, Vol 31, Issue 77 > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 15 Jan 2009 00:20:53 -0800 > > Send Freeswitch-users mailing list submissions to > freeswitch-users at 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 at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at 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 > Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 > To: > Message-ID: > 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 at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig > [root at phone2 mod_managed]# make > [root at phone2 mod_managed]# make install > > 3) Added to modules.conf.xml : > > > 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_012009 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/6a5facdc/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Thu, 15 Jan 2009 00:34:20 -0500 > From: Michael Giagnocavo > Subject: Re: [Freeswitch-users] Using mod_managed Linux/Mono 2.02 > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: > <6E8D2069C08AA84A83D336E996AE4C670235BBB97F at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long > Sent: Wednesday, January 14, 2009 3:45 PM > To: freeswitch-users at 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 at 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 > 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/20090115/73ac27e4/attachment-0001.html > > ------------------------------ > > Message: 3 > Date: Thu, 15 Jan 2009 17:50:30 +1100 > From: Scott Ellis > Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - > Freeswitch equivalent? > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: <496EDCB6.4020802 at 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 > Subject: [Freeswitch-users] Country specific tones - how to > contribute? > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: <496EE2BD.2050102 at 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 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 > Subject: Re: [Freeswitch-users] Country specific tones - how to > contribute? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <20090115072405.GA15789 at jdc.jasonjgw.net> > Content-Type: text/plain; charset=us-ascii > > Scott Ellis 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 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" > Subject: [Freeswitch-users] Changes in PlayAndGetDigits > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <27c25bc40901142343l34a3e99ftecf0df971e8e32f6 at 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 > Subject: Re: [Freeswitch-users] OpenZAP parse error [-3012] > [Q931E_INVALID_CRV] > To: freeswitch-users at lists.freeswitch.org > Message-ID: <496EF1C2.8020607 at gmx.net> > Content-Type: text/plain; charset=ISO-8859-1 > > Hello Michael, > > how much $$ are we talking about? I need this issue to be solved quickly > and it's worth to spend some money. > > I've read the following post: > > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05792.html > and have the same symptom with "after hundreds of calls I start to get b > channels that are stuck in states like "TERMINATING" or "HANGUP"" > > Best regards > Peter > > Michael Collins schrieb: > > I believe these are all symptoms of something that Stefan is working > > on: better Q931 timers. It's been on the todo list for some time but > > we've had absolutely NOBODY willing to pony up serious $$ 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 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 at 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 at 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 at 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(tm): Keep your life in sync. Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/5bb52d12/attachment-0001.html From anthony.minessale at gmail.com Thu Jan 15 15:31:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 15 Jan 2009 17:31:58 -0600 Subject: [Freeswitch-users] Freeswitch and CELT: In-Reply-To: References: Message-ID: <191c3a030901151531i71c2d461i335e48ab7e66dcd0@mail.gmail.com> try http://files.freeswitch.org/freeswitch.msi On Thu, Jan 15, 2009 at 2:03 PM, Terrance Harris wrote: > Hello, > > I have recently found out about FS and how great it is. > We are trying to use FS as a voip client for radio shows. > We have been using Trixbox and Skype but Skype isn't getting it done. > I have heard about how great the celt codec is but I don't have > enough 'skill' to compile both FS and celt in MSVC++. > Is there a binary out there that would make my day or a guide? > > Thanks much! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/02ff884f/attachment.html From mgg at giagnocavo.net Thu Jan 15 15:36:02 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 15 Jan 2009 18:36:02 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: <021001c97768$8c806a60$a5813f20$@net> References: <021001c97768$8c806a60$a5813f20$@net> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670235F10C2A@mse17be1.mse17.exchange.ms> Last time I tried it I actually built from a snapshot, which should be less stable. It was on CentOS 5.2, however. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long Sent: Thursday, January 15, 2009 4:25 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 OK yah i'm experiencing exact same problem here CentOS 5.2 2.6.18-92.el5 #1 SMP Tue Jun 10 18:49:47 EDT 2008 i686 athlon i386 GNU/Linux I too have no problems at all on Windows. I'm going to try a Suse or Ubuntu prebuilt/packaged with Mono 2 I suspect it may be kernel/mono incompatibility. Did you compile mono from tarbal? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim B Sent: Thursday, January 15, 2009 6:18 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 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 at lists.freeswitch.org > Subject: Freeswitch-users Digest, Vol 31, Issue 77 > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 15 Jan 2009 00:20:53 -0800 > > Send Freeswitch-users mailing list submissions to > freeswitch-users at 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 at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at 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 > Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 > To: > Message-ID: > 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 at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig > [root at phone2 mod_managed]# make > [root at phone2 mod_managed]# make install > > 3) Added to modules.conf.xml : > > > 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_012009 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/6a5facdc/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Thu, 15 Jan 2009 00:34:20 -0500 > From: Michael Giagnocavo > Subject: Re: [Freeswitch-users] Using mod_managed Linux/Mono 2.02 > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: > <6E8D2069C08AA84A83D336E996AE4C670235BBB97F at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long > Sent: Wednesday, January 14, 2009 3:45 PM > To: freeswitch-users at 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 at 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 > 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/20090115/73ac27e4/attachment-0001.html > > ------------------------------ > > Message: 3 > Date: Thu, 15 Jan 2009 17:50:30 +1100 > From: Scott Ellis > Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - > Freeswitch equivalent? > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: <496EDCB6.4020802 at 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 > Subject: [Freeswitch-users] Country specific tones - how to > contribute? > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: <496EE2BD.2050102 at 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 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 > Subject: Re: [Freeswitch-users] Country specific tones - how to > contribute? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <20090115072405.GA15789 at jdc.jasonjgw.net> > Content-Type: text/plain; charset=us-ascii > > Scott Ellis 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 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 $${variable-name} syntax. > > There's an ITU document referred to on the wiki with the official definitions > of ringback and other tones for various countries. > > > > > ------------------------------ > > Message: 6 > Date: Thu, 15 Jan 2009 15:43:20 +0800 > From: "Juan Backson" > Subject: [Freeswitch-users] Changes in PlayAndGetDigits > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <27c25bc40901142343l34a3e99ftecf0df971e8e32f6 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi, > > Is there a change in the playAndGetDigits api? In the old release, > 11102, my lua script is working but is not working in the latest > release. > The error I am getting is " Error in playAndGetDigits expected 10..10 > args, got 9 ". > > Thanks, > JB > > > > ------------------------------ > > Message: 7 > Date: Thu, 15 Jan 2009 09:20:18 +0100 > From: Peter P GMX > Subject: Re: [Freeswitch-users] OpenZAP parse error [-3012] > [Q931E_INVALID_CRV] > To: freeswitch-users at lists.freeswitch.org > Message-ID: <496EF1C2.8020607 at gmx.net> > Content-Type: text/plain; charset=ISO-8859-1 > > Hello Michael, > > how much $$ 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 at lists.freeswitch.org/msg05792.html > 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 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 at 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 at 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 at 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(tm): Keep your life in sync. Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/a2d86ebb/attachment-0001.html From anthony.minessale at gmail.com Thu Jan 15 15:42:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 15 Jan 2009 17:42:46 -0600 Subject: [Freeswitch-users] No Sound Heared In-Reply-To: <20090115212010.112080@gmx.net> References: <20090115212010.112080@gmx.net> Message-ID: <191c3a030901151542o6d30683dted981444b2c3bbe3@mail.gmail.com> is it only a problem in js what if you call the bridge app in the dialplan? On Thu, Jan 15, 2009 at 3:20 PM, Klaus Teller wrote: > Hi, > > Need your help on this. I have the following Javascript statement: > > session.execute("bridge","sofia/gateway/sip.gafachi.com/someNumber") in a > file called gafachiDialout.js > > Then, i have the following extension in default.xml: > > > > > > > > When i call this extension (6337), it rings as it should. But then there is > NO sound going in either direction. Any idea what i'm doing wrong here? > > Thanks, > Klaus. > > > -- > Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL > f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/794f39ea/attachment.html From anthony.minessale at gmail.com Thu Jan 15 15:48:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 15 Jan 2009 17:48:44 -0600 Subject: [Freeswitch-users] spidermonkey problems In-Reply-To: References: Message-ID: <191c3a030901151548w7504e2a0j5650449e20eff557@mail.gmail.com> do you have any loops in your code that might not check for session.ready() in a exit when its not true. The symptoms you posted would be consistent with held readlocks so if you got a gcore (or windows equiv) of the process you might be able to see what threads where doing what to hang on to the read lock. also are you creating sessions in the script then executing app with them, beware of this because the thread of the script is used to execute apps on a session created that way and not the session thread. On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin wrote: > Hello > I got problems with hanging spidermonkey sessions and need some advice on > how to debug them. > > I've made a javascript queue application that uses mod_spidermonkey_socket. > It works fine for a while, > but after some calls I noticed that calls didnt get transferred to agents. > The reason was that earlier > calls had not been terminated properly. > > freeswitch at test1> hupall > 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 > switch_core_session_hupall() Giving up with 8 sessions remaining > API CALL [hupall()] output: > +OK hangup all channels with cause MANAGER_REQUEST > > > freeswitch at test1> show calls > API CALL [show(calls)] output: > > 0 total. > > > As you can see, 8 sessions are alive, but none of them are listed as calls. > What kind of logs should I turn on to see what is happening with those > sessions? > > Thanks, > Jonas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/c2112733/attachment.html From klaus.teller at gmx.net Thu Jan 15 15:49:25 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Fri, 16 Jan 2009 00:49:25 +0100 Subject: [Freeswitch-users] No Sound Heared In-Reply-To: <191c3a030901151542o6d30683dted981444b2c3bbe3@mail.gmail.com> References: <20090115212010.112080@gmx.net> <191c3a030901151542o6d30683dted981444b2c3bbe3@mail.gmail.com> Message-ID: <20090115234925.310540@gmx.net> Hi Anthony, The problem exists also when i call session.execute("bridge","sofia/gateway/sip.gafachi.com/number"); I tend to believe that this is a firewall issue. Would you confirm? Klaus. -------- Original-Nachricht -------- > Datum: Thu, 15 Jan 2009 17:42:46 -0600 > Von: "Anthony Minessale" > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] No Sound Heared > is it only a problem in js > what if you call the bridge app in the dialplan? > > On Thu, Jan 15, 2009 at 3:20 PM, Klaus Teller > wrote: > > > Hi, > > > > Need your help on this. I have the following Javascript statement: > > > > session.execute("bridge","sofia/gateway/sip.gafachi.com/someNumber") in > a > > file called gafachiDialout.js > > > > Then, i have the following extension in default.xml: > > > > > > > > /> > > > > > > > > When i call this extension (6337), it rings as it should. But then there > is > > NO sound going in either direction. Any idea what i'm doing wrong here? > > > > Thanks, > > Klaus. > > > > > > -- > > Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL > > f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 -- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger From tharrisone at gmail.com Thu Jan 15 16:01:55 2009 From: tharrisone at gmail.com (Terrance Harris) Date: Thu, 15 Jan 2009 18:01:55 -0600 Subject: [Freeswitch-users] Freeswitch and CELT: In-Reply-To: <191c3a030901151531i71c2d461i335e48ab7e66dcd0@mail.gmail.com> References: <191c3a030901151531i71c2d461i335e48ab7e66dcd0@mail.gmail.com> Message-ID: Hello, >From what I heard celt isn't included in the most recent windows builds. I would have to build FS and celt from the source to get it enabled. On Thu, Jan 15, 2009 at 5:31 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try http://files.freeswitch.org/freeswitch.msi > > > On Thu, Jan 15, 2009 at 2:03 PM, Terrance Harris wrote: > >> Hello, >> >> I have recently found out about FS and how great it is. >> We are trying to use FS as a voip client for radio shows. >> We have been using Trixbox and Skype but Skype isn't getting it done. >> I have heard about how great the celt codec is but I don't have >> enough 'skill' to compile both FS and celt in MSVC++. >> Is there a binary out there that would make my day or a guide? >> >> Thanks much! >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/a7477a5b/attachment.html From timb0311 at hotmail.com Thu Jan 15 16:06:22 2009 From: timb0311 at hotmail.com (Tim B) Date: Thu, 15 Jan 2009 19:06:22 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: References: Message-ID: Yeah I compile mono ... i tried both 2.0.1 and 2.2 both error on loading the mod_managed. Tim _________________________________________________________________ Windows Live?: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/0cc9a8f2/attachment.html From scott.ellis at novatex.com.au Thu Jan 15 17:36:54 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Fri, 16 Jan 2009 12:36:54 +1100 Subject: [Freeswitch-users] Would like to pickup a call that is on hold on another extension Message-ID: <496FE4B6.2040808@novatex.com.au> I would like to be able to place a call on hold on one extension, walk to another phone and then dial a sequence (like the barge sequence) say 55+extension number and have the call taken off hold and transferred to the extension I am on. Has anyone done this? (Before I try and work it out for myself!) Scott From brian at freeswitch.org Thu Jan 15 18:06:46 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 20:06:46 -0600 Subject: [Freeswitch-users] Would like to pickup a call that is on hold on another extension In-Reply-To: <496FE4B6.2040808@novatex.com.au> References: <496FE4B6.2040808@novatex.com.au> Message-ID: You would use a combination of storing the UUID... in the internal db... see insert in the default dialplan... then a code to get that out of the db... then run intercept on it using the value returned from the db. See default config's Store it something like this: Then use it something like this: /b On Jan 15, 2009, at 7:36 PM, Scott Ellis wrote: > I would like to be able to place a call on hold on one extension, walk > to another phone and then dial a sequence (like the barge sequence) > say > 55+extension number and have the call taken off hold and transferred > to > the extension I am on. > > Has anyone done this? (Before I try and work it out for myself!) > > Scott From jmesquita at gmail.com Thu Jan 15 18:07:49 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 16 Jan 2009 00:07:49 -0200 Subject: [Freeswitch-users] Would like to pickup a call that is on hold on another extension In-Reply-To: <496FE4B6.2040808@novatex.com.au> References: <496FE4B6.2040808@novatex.com.au> Message-ID: Wouldnt that be call parking?? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park I have been told that would be better o use mod_fifo instead... It would be nice if someone would post something on mod_fifo wiki page about how to do fancy call parking with mod_fifo (even tho it might be pretty easy). Mesquita On Jan 15, 2009, at 11:36 PM, Scott Ellis wrote: > I would like to be able to place a call on hold on one extension, walk > to another phone and then dial a sequence (like the barge sequence) > say > 55+extension number and have the call taken off hold and transferred > to > the extension I am on. > > Has anyone done this? (Before I try and work it out for myself!) > > Scott > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jmesquita at gmail.com Thu Jan 15 18:08:53 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 16 Jan 2009 00:08:53 -0200 Subject: [Freeswitch-users] Would like to pickup a call that is on hold on another extension In-Reply-To: <496FE4B6.2040808@novatex.com.au> References: <496FE4B6.2040808@novatex.com.au> Message-ID: <2F62EB89-AF50-4461-A2B6-447BD7145A77@gmail.com> Well, sorry. That would be better, wouldnt it? http://wiki.freeswitch.org/wiki/Mod_fifo#Park_Time_Out_Example Mesquita On Jan 15, 2009, at 11:36 PM, Scott Ellis wrote: > I would like to be able to place a call on hold on one extension, walk > to another phone and then dial a sequence (like the barge sequence) > say > 55+extension number and have the call taken off hold and transferred > to > the extension I am on. > > Has anyone done this? (Before I try and work it out for myself!) > > Scott > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ajlong at worldlink.net Thu Jan 15 18:29:12 2009 From: ajlong at worldlink.net (Adam Long) Date: Thu, 15 Jan 2009 21:29:12 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: References: Message-ID: <025a01c97782$380008f0$a8001ad0$@net> I just tried 2.0.0 now too and same thing. so I don't think it's a mono version issue. I'm going to try a different kernel version next. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim B Sent: Thursday, January 15, 2009 7:06 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 Yeah I compile mono ... i tried both 2.0.1 and 2.2 both error on loading the mod_managed. Tim _____ Windows LiveT: Keep your life in sync. Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090115/5c1c0df2/attachment.html From scott.ellis at novatex.com.au Thu Jan 15 18:59:18 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Fri, 16 Jan 2009 13:59:18 +1100 Subject: [Freeswitch-users] Would like to pickup a call that is on hold on another extension In-Reply-To: References: <496FE4B6.2040808@novatex.com.au> Message-ID: <496FF806.3010805@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/2aa82c8a/attachment.html From scott.ellis at novatex.com.au Thu Jan 15 19:12:37 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Fri, 16 Jan 2009 14:12:37 +1100 Subject: [Freeswitch-users] Would like to pickup a call that is on hold on another extension In-Reply-To: References: <496FE4B6.2040808@novatex.com.au> Message-ID: <496FFB25.9080507@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/cbfbc370/attachment.html From brian at freeswitch.org Thu Jan 15 19:17:10 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jan 2009 21:17:10 -0600 Subject: [Freeswitch-users] Would like to pickup a call that is on hold on another extension In-Reply-To: <496FFB25.9080507@novatex.com.au> References: <496FE4B6.2040808@novatex.com.au> <496FFB25.9080507@novatex.com.au> Message-ID: <246FF622-6BE7-4081-AE4B-A73D5694F6C1@freeswitch.org> The key is the uuid.. In FreeSWITCH the uuid is the only bit you really need to know to do anything with the session. /b On Jan 15, 2009, at 9:12 PM, Scott Ellis wrote: > Thanks Brian, I had started looking at this, and I think I was > heading in the direction you describe - now I can pursue that with a > bit more confidence! > > So even if we do not originate the call, the last dialled extension > would still be valid as it would be set up during the bridging > process? > (I think I need another method to collect the UUID of the leg of the > bridge that initiated the call - or just the UUID that is active for > that extension) > > Scott From scott.ellis at novatex.com.au Thu Jan 15 20:43:47 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Fri, 16 Jan 2009 15:43:47 +1100 Subject: [Freeswitch-users] Would like to pickup a call that is on hold on another extension In-Reply-To: <246FF622-6BE7-4081-AE4B-A73D5694F6C1@freeswitch.org> References: <496FE4B6.2040808@novatex.com.au> <496FFB25.9080507@novatex.com.au> <246FF622-6BE7-4081-AE4B-A73D5694F6C1@freeswitch.org> Message-ID: <49701083.5040903@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/4637a2a2/attachment.html From astmac at stillnewt.org Thu Jan 15 22:29:56 2009 From: astmac at stillnewt.org (Martin Joseph) Date: Thu, 15 Jan 2009 22:29:56 -0800 Subject: [Freeswitch-users] Starting FS on OSX In-Reply-To: <9B1AAAEE-DF37-442E-ACF5-F2E3078A0991@jerris.com> References: <9B1AAAEE-DF37-442E-ACF5-F2E3078A0991@jerris.com> Message-ID: <02FDD555-C096-4F09-91DC-A9B57A44427F@stillnewt.org> On Jan 15, 2009, at 3:10 PM, Michael Jerris wrote: > Your build issue is with your autotools install, I have seen issues if > you have ever installed any of the autotools from macports or fink. I have never used Fink or Macports so that isn't it. In fact the supposed statements made to the effect that FS will build from SVN fine on 10.4 with the latest available apple dev tools is quite wrong in my experience. I setup a virgin 10.4 and updated everything and had many complaints from FS about tool versions. > > If you want to build from svn you can run bootstrap on another box (a > linux box perhaps) and then tar up that dir and move it to your mac. Huh, interesting. > > We pre-bootstrap the release tarballs which is why that is building > fine for you. Right, Thanks for all your efforts and an outstanding platform! Marty From helmut.kuper at ewetel.de Thu Jan 15 23:25:47 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 16 Jan 2009 08:25:47 +0100 Subject: [Freeswitch-users] FS doesn't maitain PRI- D-channel state right Message-ID: <4970367B.5090901@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I found, that FS doesn't maintain D-Channel's state correctly. I have a PRI with disabled layer 2 and 3 on TDM side. When FS starts up I get this on console: 2009-01-16 08:16:10 [DEBUG] ozmod_isdn.c:1441 zap_isdn_run() ISDN thread starting. 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Starting trunk 0xae411008 (sapi: 0, tei: 0, mode: PTP TE) 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending SABME 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending frame - ----------------- Q.921 Packet [Outgoing] --------------- SAPI: 0, TEI: 0, C/R: Command (0) Type: U Frame (SABME) P/F: 1 Q.921 state: "TEI Assigned" (4) [flags: ----] - ---------------------------------------------- 2009-01-16 08:16:10 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'openzap' 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'disable_ec' 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'oz' 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() New packet received (3 bytes) - ----------------- Q.921 Packet [Incoming] --------------- SAPI: 0, TEI: 0, C/R: Response (0) Type: U Frame (DM (Disconnected Mode)) P/F: 1 Q.921 state: "TEI Assigned" (4) [flags: ----] - ---------------------------------------------- So TDM side tells FS that the link is down: but when I check d-channel's state FS tells me that it is UP: freeswitch at ippbx-prod-node0> oz dump 1 16 API CALL [oz(dump 1 16)] output: span_id: 1 chan_id: 16 physical_span_id: 1 physical_chan_id: 16 type: DQ921 state: UP last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE When I do an outbound call FS throws no error or warning that the link isn't up and gives the call a try. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklwNnsACgkQ4tZeNddg3dzjKQCbBDU/SSOyKbD2JGcJFOJZDyBQ nI0An3CFp9HIuTB0cQWT0iJ1Rlx1+yGk =ycwj -----END PGP SIGNATURE----- From kristian.kielhofner at gmail.com Fri Jan 16 00:17:35 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 16 Jan 2009 03:17:35 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC695A926@srvmtel.office.mtel.nl> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <006d01c9774d$9a94cd00$cfbe6700$@net> <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com> <009301c97751$0d04e250$270ea6f0$@net> <11372C8B9E603F4FACDE6AB18256DEC695A926@srvmtel.office.mtel.nl> Message-ID: <2d9149cd0901160017j3b4ae56ck9f3ae8d00860d9e7@mail.gmail.com> Speaking of networking... After timing that's the next "achilles heel" of RTP handing with virtualization. Very, very few of these platforms were designed to handle massive numbers of very small RTP packets. Everything from interrupt handling on the actual ethernet adapter to getting the data into userspace within the virtual instance is worrisome: http://www.xen.org/files/xensummit_4/NetworkIO_Santos.pdf http://forum.openvz.org/index.php?t=msg&goto=11619& Interestingly enough the Xen paper makes it out to be really bad yet the OpenVZ post praises Xen's performance. Without any real testing, who knows? I just know that scaling 50pps per RTP stream (20ms packetization) can be difficult enough on native hardware, let alone [virtualization technology du jour]. On Thu, Jan 15, 2009 at 5:02 PM, Remko Kloosterman wrote: > Lot's of experience and suggestions here. Thanks. > > I believe it should be theoretically possible to have blip-free RTP > streaming through the appliance. Most windows ethernet drivers allow for > QoS packet scheduling. If the VMware network bridge driver honors this > and syncs the buffers at 20ms frames (or whatever frame size applies) > you should be able to schale up a bit and maintain low jitter. > > Anyone knows how the VMware network bridge exactly works? > > > -----Oorspronkelijk bericht----- > Van: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Gregory > Boehnlein > Verzonden: donderdag 15 januari 2009 21:37 > Aan: freeswitch-users at lists.freeswitch.org > Onderwerp: Re: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality > >> To the contrary, we have had quite good results in virtualized >> environments and you don't really need timing that is that accurate to > >> make it work. > > If you don't handle RTP, I'm sure it is amazing. However, if you have to > do voicemail, stream audio from the server or do any kind of actual > time/latency/jitter sensitive processing, I don't care how much you tune > your hypervisor, it's never going to scale. > >> We work quite well on amazon EC2 for example. There are 2 issues I >> know about with vmware, 1 is you need to set a setting on the host to >> extend somewhat sane clocks being available, the second is I have seen > >> issues with the bridged network adapter actually doubling up all >> packets causing very strange issues, I suggest not using bridged >> networking if you experience this. > > I've not seen this behavior on Vmware ESX 3.5u2. Maybe an issue on > Vmware Server or Workstation? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From jonas.gauffin at gmail.com Fri Jan 16 00:38:13 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Fri, 16 Jan 2009 09:38:13 +0100 Subject: [Freeswitch-users] spidermonkey problems In-Reply-To: <191c3a030901151548w7504e2a0j5650449e20eff557@mail.gmail.com> References: <191c3a030901151548w7504e2a0j5650449e20eff557@mail.gmail.com> Message-ID: I've got a loop, but the first thing checked in each iteration is if session.ready() returns false (and in that case exit the loop). I do create sessions in the script: create, try to originate to a destination and then finally bridge together the caller and the new session. I'll try to give you more details during the day. On Fri, Jan 16, 2009 at 12:48 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > do you have any loops in your code that might not check for session.ready() > in a exit when its not true. > > The symptoms you posted would be consistent with held readlocks so if you > got a gcore (or windows equiv) of the process you might be able to see what > threads where doing what to hang on to the read lock. > > also are you creating sessions in the script then executing app with them, > beware of this because the thread of the script is used to execute apps on a > session created that way and not the session thread. > > > > > On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin wrote: > >> Hello >> I got problems with hanging spidermonkey sessions and need some advice on >> how to debug them. >> >> I've made a javascript queue application that uses >> mod_spidermonkey_socket. It works fine for a while, >> but after some calls I noticed that calls didnt get transferred to agents. >> The reason was that earlier >> calls had not been terminated properly. >> >> freeswitch at test1> hupall >> 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 >> switch_core_session_hupall() Giving up with 8 sessions remaining >> API CALL [hupall()] output: >> +OK hangup all channels with cause MANAGER_REQUEST >> >> >> freeswitch at test1> show calls >> API CALL [show(calls)] output: >> >> 0 total. >> >> >> As you can see, 8 sessions are alive, but none of them are listed as >> calls. What kind of logs should I turn on to see what is happening with >> those sessions? >> >> Thanks, >> Jonas >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/1b3c0088/attachment.html From regs at kinetix.gr Fri Jan 16 02:32:56 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Fri, 16 Jan 2009 12:32:56 +0200 Subject: [Freeswitch-users] mod_opal first unsuccessful test In-Reply-To: <496DCC18.9040006@kinetix.gr> References: <496B66F3.9000902@kinetix.gr> <191c3a030901121119o688c722er745106b014367569@mail.gmail.com> <024701c97510$cb9a1890$62ce49b0$@jongbloed@bigpond.com> <496C7EFC.8060401@kinetix.gr> <496C834E.9070608@kinetix.gr> <02d101c975d1$aebd0e50$0c372af0$@jongbloed@bigpond.com> <496DCC18.9040006@kinetix.gr> Message-ID: <49706258.50501@kinetix.gr> Any news regarding this issue? Apostolos Pantsiopoulos wrote: > I am attaching the wireshark capture. Openphone is on xxx.xxx.xxx.202 and > FS is on xxx.xxx.xxx.212 > > Robert Jongbloed wrote: >> >> Can you send me a WireShark capture? >> >> >> >> Robert Jongbloed >> >> OPAL/OpenH323/PTLib Architect and Co-founder. >> >> >> >> *From:* Apostolos Pantsiopoulos [mailto:regs at kinetix.gr] >> *Sent:* Tuesday, 13 January 2009 11:05 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Cc:* Robert Jongbloed >> *Subject:* Re: [Freeswitch-users] mod_opal first unsuccessful test >> >> >> >> I also tried using Ekiga - which is OPAL based - and got the same >> behavior. No audio - although I can see RTP packets. >> >> Apostolos Pantsiopoulos wrote: >> >> Hi, >> >> Yes, openPhone is working with my soundcard. I am using it >> every day for testing purposes. I use the 1.8.1 version. Is there a newer >> version that uses OPAL? I didn't know that. Where can I get it from? >> >> Robert Jongbloed wrote: >> >> Hi guys, >> >> >> >> I was using the OpenPhone that you build with OPAL for my testing. So >> that is identical (I think) to you. >> >> >> >> I have not (yet) do any third party client testing. >> >> >> >> Two ALERTING messages are fine, perfectly legal and OPAL can handle it. >> >> >> >> You say you can see the RTP packets flowing so that implies that the >> mod_opal is actually working, so let's look somewhere else. Have you >> confirmed that OpenPhone is using the sound card correctly? Made a >> call between two machines JUST using OpenPhone for example? >> >> >> >> >> >> Robert Jongbloed >> >> OPAL/OpenH323/PTLib Architect and Co-founder. >> >> >> >> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] >> *Sent:* Tuesday, 13 January 2009 6:20 AM >> *To:* Robert Jongbloed >> *Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test >> >> >> >> Heh, >> what client are you using in your tests that are working? >> >> >> ---------- Forwarded message ---------- >> From: *Apostolos Pantsiopoulos* > > >> Date: Mon, Jan 12, 2009 at 9:51 AM >> Subject: [Freeswitch-users] mod_opal first unsuccessful test >> To: freeswitch-users at lists.freeswitch.org >> >> >> >> Hi, >> >> I successfully compiled mod_opal using the latest svn for both opal >> and ptlib as Brian suggested. >> >> When I try to establish a call using h323 from my openphone client >> I get no audio although I can see RTP packets in both directions when >> I am >> doing a capture. Some details : >> >> I am using the 11094 revision of the FS trunk. >> I am using the PCMU codec. >> I tried dialing the default IVR (5000) and other testing extensions >> (freeswitch conference, echo test etc.) >> I tried using fast start on and off , h245 tunneling on and off, h245 in >> SETUP on and off. >> >> In my captures I have also noticed a strange behavior : FS sends to >> my client 2 "alerting" packets >> for no apparent reason. Could this be a cause of the problem? >> >> Had anyone any success with mod_opal lately? If yes, could you >> please reply quoting your config >> options (both on FS and on your client)? >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> ------------------------------------------------------------------------ >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/04e35870/attachment-0001.html From jonas.gauffin at gmail.com Fri Jan 16 02:54:30 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Fri, 16 Jan 2009 11:54:30 +0100 Subject: [Freeswitch-users] spidermonkey problems In-Reply-To: References: <191c3a030901151548w7504e2a0j5650449e20eff557@mail.gmail.com> Message-ID: I've found the problem. one js thread wait in socket.read (mod_spidermonkey_socket) on data. That caller have hangup, which means that the garbage collector waits on it to close. All new javascript sessions waits in JS_AWAIT_GC_DONE for the garbage collector to be done before proceeding (which means that all new javascript calls don't do anything after being launched). My server will not send anything until an agent gets free or the session hangs up (detects it through the event socket). And the event socket will not send that the session has been hangup until the socket have received anything (and the script can exit). So it's kind of deadlock between my server and the spidermonkey_socket. Is it possible to add an option to socket.read to make it abort if the session have been closed? I know that I wrote mod_spidermonkey_socket from the start, but I can't figure out how to do it. Will new sessions always wait on old ones to be garbage collected properly? For instance, what happens if a script have a lenghty post process after caller have hang up? On Fri, Jan 16, 2009 at 9:38 AM, Jonas Gauffin wrote: > I've got a loop, but the first thing checked in each iteration is if > session.ready() returns false (and in that case exit the loop). > I do create sessions in the script: create, try to originate to a > destination and then finally bridge together the caller and the new session. > > I'll try to give you more details during the day. > > On Fri, Jan 16, 2009 at 12:48 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> do you have any loops in your code that might not check for >> session.ready() in a exit when its not true. >> >> The symptoms you posted would be consistent with held readlocks so if you >> got a gcore (or windows equiv) of the process you might be able to see what >> threads where doing what to hang on to the read lock. >> >> also are you creating sessions in the script then executing app with them, >> beware of this because the thread of the script is used to execute apps on a >> session created that way and not the session thread. >> >> >> >> >> On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin wrote: >> >>> Hello >>> I got problems with hanging spidermonkey sessions and need some advice on >>> how to debug them. >>> >>> I've made a javascript queue application that uses >>> mod_spidermonkey_socket. It works fine for a while, >>> but after some calls I noticed that calls didnt get transferred to >>> agents. The reason was that earlier >>> calls had not been terminated properly. >>> >>> freeswitch at test1> hupall >>> 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 >>> switch_core_session_hupall() Giving up with 8 sessions remaining >>> API CALL [hupall()] output: >>> +OK hangup all channels with cause MANAGER_REQUEST >>> >>> >>> freeswitch at test1> show calls >>> API CALL [show(calls)] output: >>> >>> 0 total. >>> >>> >>> As you can see, 8 sessions are alive, but none of them are listed as >>> calls. What kind of logs should I turn on to see what is happening with >>> those sessions? >>> >>> Thanks, >>> Jonas >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 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 >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/c4d31086/attachment.html From regs at kinetix.gr Fri Jan 16 04:51:40 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Fri, 16 Jan 2009 14:51:40 +0200 Subject: [Freeswitch-users] Outbound call - choose profile Message-ID: <497082DC.8020904@kinetix.gr> When I am using the following method to place a call from the dialplan : sofia/gateway// how do I tell FS which profile to use (as in the sofia// method?) I am asking that because all my calls to my declared use the 5080 port, and I want them to use the 5060 port. Is there a way to configure a to use a specific profile when making outbound calls? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From cstomi.levlist at gmail.com Fri Jan 16 05:17:12 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Fri, 16 Jan 2009 14:17:12 +0100 Subject: [Freeswitch-users] Channel variable 'call_timeout'. In-Reply-To: <191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com> References: <20677406.post@talk.nabble.com> <191c3a030811250600n5ba54fc0qb219b09e19726adf@mail.gmail.com> <20687620.post@talk.nabble.com> <191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com> <87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com> <20689832.post@talk.nabble.com> <87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com> <87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com> <49395825.2010008@gmail.com> <191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com> Message-ID: <497088D8.4090504@gmail.com> Hello, It seems originate_timeout isn't take effect when we got early media. Our carrier is sending the ring tone in early media, so if I try timeout isn't occur after 15s however with: timeout is OK. but we don't get early media, I think it would be nice to not ignore it I found ring_ready and return_ring_ready could you please tell me what do they mean? So I'd like to make the timer expire only on 200 OK. and don't ignore early media. Thank you, Tomi Anthony Minessale ?rta: > call_timeout is only used if you are bridging 2 channels where one or both > of them is still unanswered. > > what you want to use is originate_timeout and forget about call_timeout > > you also have > leg_timeout and leg_progress_timeout both to be set in the {} > that do the timeout from the perspective of the new channel leg instead of > the caller leg. > > > > On Fri, Dec 5, 2008 at 10:34 AM, Tamas Cseke wrote: > > >> Hello, >> >> I have the same problem, >> >> I don't understand the difference between >> >> progress_timeout >> originate_timeout >> call_timeout. >> >> I log timelimit_sec in switch_ivr_originate function and it seems, >> if I set call_timeout then, timelimit_sec will be this value >> if I set originate_timeout then timelimit_sec will be this value. maybe >> this is for backward compat? >> >> originate_timeout as in the wiki: >> "Determines how long FreeSwitch is going to wait for a response from >> the invite message sent to the gateway. " >> >> I quess this would be an 100 reply. >> >> But how could I set a timeout for 200? I mean timeout for an answer. >> which variable would control this? >> I quess it was call_timeout previosly. >> Please explain me this timeout variables >> >> Thanks, >> Tamas >> >> Michael Collins ?rta: >> >>> FYI, it is on the channel variables page but it's in a crazy place under >>> "unknown functionality" which is silly. >>> http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout >>> >>> Anyway, I've got wiki cleaning on my to-do list and I'll start in earnest >>> next month when I have some time... >>> >>> -MC >>> >>> On Tue, Nov 25, 2008 at 1:32 PM, Michael Collins >>> >> wrote: >> >>> >>>>> I used "call-timeout" because that's how I understood it from the Wiki. >>>>> (?) >>>>> >>>>> >>>>> >>>> Yep, that's all that there is on the wiki. Unfortunately the channel >>>> variables page is one of many in need of some attention. I will add >>>> "originate_timeout" right away. The only question remaining is what, if >>>> anything, does call_timeout do? That channel variable is in the source >>>> >> code >> >>>> but I don't know exactly what it does. >>>> >>>> -MC >>>> >>>> >>>> >>>> >>>>> -- >>>>> View this message in context: >>>>> >>>>> >> http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20689832.html >> >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From intralanman at freeswitch.org Fri Jan 16 06:31:55 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 16 Jan 2009 14:31:55 +0000 Subject: [Freeswitch-users] Outbound call - choose profile In-Reply-To: <497082DC.8020904@kinetix.gr> References: <497082DC.8020904@kinetix.gr> Message-ID: <49709A5B.2060806@freeswitch.org> Apostolos Pantsiopoulos wrote: > When I am using the following method to place a call from the dialplan : > > sofia/gateway// > > how do I tell FS which profile to use (as in the > sofia// method?) > > I am asking that because all my calls to my declared use the > 5080 port, > and I want them to use the 5060 port. Is there a way to configure a > to use > a specific profile when making outbound calls? > > the gateway should always use the profile from which it was included. so, for instance, if you include gw1 from internal.xml, then gw1 should always use the internal profile -Ray -------------- next part -------------- A non-text attachment was scrubbed... Name: intralanman.vcf Type: text/x-vcard Size: 209 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/40445198/attachment.vcf From anthony.minessale at gmail.com Fri Jan 16 06:50:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 16 Jan 2009 08:50:07 -0600 Subject: [Freeswitch-users] FS doesn't maitain PRI- D-channel state right In-Reply-To: <4970367B.5090901@ewetel.de> References: <4970367B.5090901@ewetel.de> Message-ID: <191c3a030901160650t5c4209a5kf6521883e7ca3d3e@mail.gmail.com> Please do not report bugs on the mailing list. It's very hard to keep track of them this way. Please file all bugs to jira so we will not lose track of them. http://jira.freeswitch.org On Fri, Jan 16, 2009 at 1:25 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > > I found, that FS doesn't maintain D-Channel's state correctly. > > I have a PRI with disabled layer 2 and 3 on TDM side. When FS starts up > I get this on console: > > > 2009-01-16 08:16:10 [DEBUG] ozmod_isdn.c:1441 zap_isdn_run() ISDN thread > starting. > 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Starting trunk 0xae411008 > (sapi: 0, tei: 0, mode: PTP TE) > 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending SABME > 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending frame > - ----------------- Q.921 Packet [Outgoing] --------------- > SAPI: 0, TEI: 0, C/R: Command (0) > > Type: U Frame (SABME) > P/F: 1 > > Q.921 state: "TEI Assigned" (4) [flags: ----] > - ---------------------------------------------- > > 2009-01-16 08:16:10 [CONSOLE] switch_loadable_module.c:857 > switch_loadable_module_load_file() Successfully Loaded [mod_openzap] > 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:141 > switch_loadable_module_process() Adding Endpoint 'openzap' > 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:239 > switch_loadable_module_process() Adding Application 'disable_ec' > 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:259 > switch_loadable_module_process() Adding API Function 'oz' > 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() New packet received (3 bytes) > - ----------------- Q.921 Packet [Incoming] --------------- > SAPI: 0, TEI: 0, C/R: Response (0) > > Type: U Frame (DM (Disconnected Mode)) > P/F: 1 > > Q.921 state: "TEI Assigned" (4) [flags: ----] > - ---------------------------------------------- > > So TDM side tells FS that the link is down: > > but when I check d-channel's state FS tells me that it is UP: > > freeswitch at ippbx-prod-node0> oz dump 1 16 > API CALL [oz(dump 1 16)] output: > span_id: 1 > chan_id: 16 > physical_span_id: 1 > physical_chan_id: 16 > type: DQ921 > state: UP > last_state: DOWN > cid_date: > cid_name: > cid_num: > ani: > aniII: > dnis: > rdnis: > cause: NONE > > When I do an outbound call FS throws no error or warning that the link > isn't up and gives the call a try. > > > regards > helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAklwNnsACgkQ4tZeNddg3dzjKQCbBDU/SSOyKbD2JGcJFOJZDyBQ > nI0An3CFp9HIuTB0cQWT0iJ1Rlx1+yGk > =ycwj > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/618e5518/attachment-0001.html From regs at kinetix.gr Fri Jan 16 06:53:47 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Fri, 16 Jan 2009 16:53:47 +0200 Subject: [Freeswitch-users] Outbound call - choose profile In-Reply-To: <49709A5B.2060806@freeswitch.org> References: <497082DC.8020904@kinetix.gr> <49709A5B.2060806@freeswitch.org> Message-ID: <49709F7B.4030905@kinetix.gr> Raymond Chandler wrote: > Apostolos Pantsiopoulos wrote: >> When I am using the following method to place a call from the dialplan : >> >> sofia/gateway// >> >> how do I tell FS which profile to use (as in the >> sofia// method?) >> >> I am asking that because all my calls to my declared use >> the 5080 port, >> and I want them to use the 5060 port. Is there a way to configure a >> to use >> a specific profile when making outbound calls? >> >> > the gateway should always use the profile from which it was included. > so, for instance, if you include gw1 from internal.xml, then gw1 > should always use the internal profile > > -Ray > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > I am using the directory to define my gateways. Specifically, because this certain gateway is related to a specific user I am including this gateway in the user's xml file (in the dieractory). Then I use : in my internal profile to let FS "parse" the gateways that I have declared in the directory. So one would expect that the gateways declared in the user's file would belong to the internal profile. Yet, when I am using the sofia/gateway// notation to send a call through this gateway the SIP packets get send from the 5080 port (which is my external profile's port) -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From msc at freeswitch.org Fri Jan 16 07:30:13 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 16 Jan 2009 07:30:13 -0800 Subject: [Freeswitch-users] Channel variable 'call_timeout'. In-Reply-To: <497088D8.4090504@gmail.com> References: <20677406.post@talk.nabble.com> <191c3a030811250600n5ba54fc0qb219b09e19726adf@mail.gmail.com> <20687620.post@talk.nabble.com> <191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com> <87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com> <20689832.post@talk.nabble.com> <87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com> <87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com> <49395825.2010008@gmail.com> <191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com> <497088D8.4090504@gmail.com> Message-ID: <992FC0F5-E3C8-4DBF-B3E2-4E2BF691451E@freeswitch.org> Tamas, The channel variable won't work for you if you can't ignore early media. Your best bet is to use the variable execute_on_answer to transfer an answered call to a new extension. Then you could just sleep for 15sec and then check the value of endpoint_disposition. What is the application that you are working on? Sent from my iPhone On Jan 16, 2009, at 5:17 AM, Tamas Cseke wrote: > Hello, > > It seems originate_timeout isn't take effect when we got early media. > > Our carrier is sending the ring tone in early media, so if I try > data="{originate_timeout=15}sofia/gateway/mygw/whatsoever"/> > timeout isn't occur after 15s > > however with: > > > timeout is OK. but we don't get early media, I think it would be > nice to > not ignore it > > I found ring_ready and return_ring_ready could you please tell me what > do they mean? > So I'd like to make the timer expire only on 200 OK. and don't ignore > early media. > > Thank you, > Tomi > > > Anthony Minessale ?rta: >> call_timeout is only used if you are bridging 2 channels where one >> or both >> of them is still unanswered. >> >> what you want to use is originate_timeout and forget about >> call_timeout >> >> you also have >> leg_timeout and leg_progress_timeout both to be set in the {} >> that do the timeout from the perspective of the new channel leg >> instead of >> the caller leg. >> >> >> >> On Fri, Dec 5, 2008 at 10:34 AM, Tamas Cseke > >wrote: >> >> >>> Hello, >>> >>> I have the same problem, >>> >>> I don't understand the difference between >>> >>> progress_timeout >>> originate_timeout >>> call_timeout. >>> >>> I log timelimit_sec in switch_ivr_originate function and it seems, >>> if I set call_timeout then, timelimit_sec will be this value >>> if I set originate_timeout then timelimit_sec will be this value. >>> maybe >>> this is for backward compat? >>> >>> originate_timeout as in the wiki: >>> "Determines how long FreeSwitch is going to wait for a response from >>> the invite message sent to the gateway. " >>> >>> I quess this would be an 100 reply. >>> >>> But how could I set a timeout for 200? I mean timeout for an answer. >>> which variable would control this? >>> I quess it was call_timeout previosly. >>> Please explain me this timeout variables >>> >>> Thanks, >>> Tamas >>> >>> Michael Collins ?rta: >>> >>>> FYI, it is on the channel variables page but it's in a crazy >>>> place under >>>> "unknown functionality" which is silly. >>>> http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout >>>> >>>> Anyway, I've got wiki cleaning on my to-do list and I'll start in >>>> earnest >>>> next month when I have some time... >>>> >>>> -MC >>>> >>>> On Tue, Nov 25, 2008 at 1:32 PM, Michael Collins >>> > >>>> >>> wrote: >>> >>>> >>>>>> I used "call-timeout" because that's how I understood it from >>>>>> the Wiki. >>>>>> (?) >>>>>> >>>>>> >>>>>> >>>>> Yep, that's all that there is on the wiki. Unfortunately the >>>>> channel >>>>> variables page is one of many in need of some attention. I will >>>>> add >>>>> "originate_timeout" right away. The only question remaining is >>>>> what, if >>>>> anything, does call_timeout do? That channel variable is in the >>>>> source >>>>> >>> code >>> >>>>> but I don't know exactly what it does. >>>>> >>>>> -MC >>>>> >>>>> >>>>> >>>>> >>>>>> -- >>>>>> View this message in context: >>>>>> >>>>>> >>> http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20689832.html >>> >>>>>> Sent from the Freeswitch-users mailing list archive at >>>>>> Nabble.com. >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ivan at myrvold.org Fri Jan 16 08:09:51 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Fri, 16 Jan 2009 17:09:51 +0100 Subject: [Freeswitch-users] Starting FS on OSX In-Reply-To: References: Message-ID: I haven't tried using launchctl for FreeSWITCH. But when I saw your post, I tried it out. I have no problem getting it to work: I make a file "org.freeswitch.freeswitch.plist" and save it to ~/ Library/LaunchAgents with the following content: KeepAlive Label org.freeswitch.freeswitch Program /usr/local/freeswitch/bin/freeswitch RunAtLoad ServiceIPC Then in Terminal.app, I do a "launchctl load ~/Library/LaunchAgents/ org.freeswitch.freeswitch.plist" If you do the same command, but unload instead of load, it should stop freeswitch. Does this work for you? Ivan Den 15. jan.. 2009 kl. 22:54 skrev Martin Joseph: > Hello again FreeSwitchers, > > I have built the 1.02 on 10.4.11(OSX) and had no problems with that. > > I have never been able to build from the SVN, but that is another > story. > > Now that I have migrated to 1.02 I was wondering if I can get some > help on a long standing issue I have with starting FS at boot. > > I am hoping to use Launchd which is the standard on OSX 10.4 and I > attempted to cobble together a script, but haven't had great results. > > I did search for wiki entries on this, but haven't found any help with > it. > > Ideas? > Thanks, > Marty > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shido6 at gmail.com Fri Jan 16 05:45:14 2009 From: shido6 at gmail.com (Shido Xavier) Date: Fri, 16 Jan 2009 08:45:14 -0500 Subject: [Freeswitch-users] Starting FS on OSX In-Reply-To: <02FDD555-C096-4F09-91DC-A9B57A44427F@stillnewt.org> References: <9B1AAAEE-DF37-442E-ACF5-F2E3078A0991@jerris.com> <02FDD555-C096-4F09-91DC-A9B57A44427F@stillnewt.org> Message-ID: <1c4b29080901160545o21caea3dl45e2360edc1d76c7@mail.gmail.com> Please specify Intel or PPC. -Greg M. On Fri, Jan 16, 2009 at 1:29 AM, Martin Joseph wrote: > > On Jan 15, 2009, at 3:10 PM, Michael Jerris wrote: > >> Your build issue is with your autotools install, I have seen issues if >> you have ever installed any of the autotools from macports or fink. > I have never used Fink or Macports so that isn't it. In fact the > supposed statements made to the effect that FS will build from SVN > fine on 10.4 with the latest available apple dev tools is quite wrong > in my experience. I setup a virgin 10.4 and updated everything and had > many complaints from FS about tool versions. >> >> If you want to build from svn you can run bootstrap on another box (a >> linux box perhaps) and then tar up that dir and move it to your mac. > Huh, interesting. >> >> We pre-bootstrap the release tarballs which is why that is building >> fine for you. > Right, Thanks for all your efforts and an outstanding platform! > > Marty > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Fri Jan 16 09:03:03 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 16 Jan 2009 12:03:03 -0500 Subject: [Freeswitch-users] spidermonkey problems In-Reply-To: References: <191c3a030901151548w7504e2a0j5650449e20eff557@mail.gmail.com> Message-ID: <07F43D7C-3CE2-4A55-819B-E8F1095C5C8B@jerris.com> All long running non js code should be wrapped in the suspend/resume gc stuff. For example: cb_state.ret = BOOLEAN_TO_JSVAL(JS_FALSE); cb_state.saveDepth = JS_SuspendRequest(cx); args.input_callback = dtmf_func; args.buf = bp; args.buflen = len; switch_ivr_sleep(jss->session, ms, sync, &args); JS_ResumeRequest(cx, cb_state.saveDepth); I think this is your issue. Can you please file a bug on jira for this issue (even better with a patch) Mike On Jan 16, 2009, at 5:54 AM, Jonas Gauffin wrote: > I've found the problem. one js thread wait in socket.read > (mod_spidermonkey_socket) on data. > That caller have hangup, which means that the garbage collector > waits on it to close. > > All new javascript sessions waits in JS_AWAIT_GC_DONE for the > garbage collector to be done before proceeding (which means that all > new javascript calls don't do anything after being launched). > > My server will not send anything until an agent gets free or the > session hangs up (detects it through the event socket). And the > event socket will not send that the session has been hangup until > the socket have received anything (and the script can exit). So it's > kind of deadlock between my server and the spidermonkey_socket. > > Is it possible to add an option to socket.read to make it abort if > the session have been closed? I know that I wrote > mod_spidermonkey_socket from the start, but I can't figure out how > to do it. > > Will new sessions always wait on old ones to be garbage collected > properly? For instance, what happens if a script have a lenghty post > process after caller have hang up? > > On Fri, Jan 16, 2009 at 9:38 AM, Jonas Gauffin > wrote: > I've got a loop, but the first thing checked in each iteration is if > session.ready() returns false (and in that case exit the loop). > > I do create sessions in the script: create, try to originate to a > destination and then finally bridge together the caller and the new > session. > > I'll try to give you more details during the day. > > On Fri, Jan 16, 2009 at 12:48 AM, Anthony Minessale > wrote: > do you have any loops in your code that might not check for > session.ready() in a exit when its not true. > > The symptoms you posted would be consistent with held readlocks so > if you got a gcore (or windows equiv) of the process you might be > able to see what threads where doing what to hang on to the read lock. > > also are you creating sessions in the script then executing app with > them, beware of this because the thread of the script is used to > execute apps on a session created that way and not the session thread. > > > > > On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin > wrote: > Hello > > I got problems with hanging spidermonkey sessions and need some > advice on how to debug them. > > I've made a javascript queue application that uses > mod_spidermonkey_socket. It works fine for a while, > but after some calls I noticed that calls didnt get transferred to > agents. The reason was that earlier > calls had not been terminated properly. > > freeswitch at test1> hupall > 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 > switch_core_session_hupall() Giving up with 8 sessions remaining > API CALL [hupall()] output: > +OK hangup all channels with cause MANAGER_REQUEST > > > freeswitch at test1> show calls > API CALL [show(calls)] output: > > 0 total. > > > As you can see, 8 sessions are alive, but none of them are listed as > calls. What kind of logs should I turn on to see what is happening > with those sessions? > > Thanks, > Jonas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/2f1c49b9/attachment-0001.html From anthony.minessale at gmail.com Fri Jan 16 09:15:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 16 Jan 2009 11:15:08 -0600 Subject: [Freeswitch-users] Outbound call - choose profile In-Reply-To: <49709F7B.4030905@kinetix.gr> References: <497082DC.8020904@kinetix.gr> <49709A5B.2060806@freeswitch.org> <49709F7B.4030905@kinetix.gr> Message-ID: <191c3a030901160915n250d791m4451815341b8d60a@mail.gmail.com> all gateways are properties of a profile so even in the parse domain way of handling gateways whatever profile parsed the domain will be the owner of all the gateways discovered from the domain. FYI, The internal profile has nothing to do with users or anything else. You are confusing the default examples for absolute fact. On Fri, Jan 16, 2009 at 8:53 AM, Apostolos Pantsiopoulos wrote: > Raymond Chandler wrote: > > Apostolos Pantsiopoulos wrote: > >> When I am using the following method to place a call from the dialplan : > >> > >> sofia/gateway// > >> > >> how do I tell FS which profile to use (as in the > >> sofia// method?) > >> > >> I am asking that because all my calls to my declared use > >> the 5080 port, > >> and I want them to use the 5060 port. Is there a way to configure a > >> to use > >> a specific profile when making outbound calls? > >> > >> > > the gateway should always use the profile from which it was included. > > so, for instance, if you include gw1 from internal.xml, then gw1 > > should always use the internal profile > > > > -Ray > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > I am using the directory to define my gateways. Specifically, because > this certain gateway > is related to a specific user I am including this gateway in the user's > xml file (in the dieractory). > Then I use : > > > > in my internal profile to let FS "parse" the gateways that I have > declared in the directory. > > So one would expect that the gateways declared in the user's file would > belong to the internal > profile. Yet, when I am using the sofia/gateway// > notation to send a call > through this gateway the SIP packets get send from the 5080 port (which > is my external profile's port) > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/a0336b0c/attachment.html From imthiyazg at gmail.com Fri Jan 16 09:19:25 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Fri, 16 Jan 2009 22:49:25 +0530 Subject: [Freeswitch-users] Sending SMS to SIPtoGSM gateway Message-ID: <8595daf70901160919u8f27353s27e9c1931c5acc05@mail.gmail.com> Hi This is what I found in the manual SMS and MMS The GSM and CDMA modules can send and receive SMS text messages and MMS multimedia messages to or from a mobile phone. These are transmitted over the Ethernet port as SIP MESSAGE messages to ensure compatibility with a variety of different SIP PBXs. You determine for each module the destination of received SMS or MMS messages and you can program that SMS and MMS messages are always sent to a person who sent an SMS or MMS to that phone number within a specified time. The GS8 modular gateway can send and receive text messages in 7 bit mode (160 characters per message) or in Unicode (70 characters per message). It can send and receive multimedia messages comprising images and rich text that are limited to 30 KB. The sending or receiving client is responsible for formatting or interpreting the message to OMA specifications. Thanks From anthony.minessale at gmail.com Fri Jan 16 11:00:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 16 Jan 2009 13:00:56 -0600 Subject: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality In-Reply-To: <2d9149cd0901160017j3b4ae56ck9f3ae8d00860d9e7@mail.gmail.com> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <006d01c9774d$9a94cd00$cfbe6700$@net> <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com> <009301c97751$0d04e250$270ea6f0$@net> <11372C8B9E603F4FACDE6AB18256DEC695A926@srvmtel.office.mtel.nl> <2d9149cd0901160017j3b4ae56ck9f3ae8d00860d9e7@mail.gmail.com> Message-ID: <191c3a030901161100l2184d40ya4723d2e6bc1e3a9@mail.gmail.com> Yes it's hard to trust virtualized stuff because you have no idea what they skimp on in terms of realtime access. I won't endorse using FS on a VM as i have not done it very extensively beyond openvz but I can point out a few reasons why it has a fighting chance. FS uses a timer architecture designed to amplify the work of one timer thread into every timer open by FS. This single thread uses the monotonic clock on the system to try and perfrom a 1ms accurate loop. This single loop updates a soft value for current epoch time and microsecond epoch time with the goal of (again) being as close as possible to being accurate to 1 ms. The timer loop also has a global matrix to all of the timing intervals being subscribed to by a timer open by FS. The loop will tick a counter in each unique timing interval (10ms, 20ms, 60ms etc) and fire a conditional broadcast to all of the timers who are blocking for a tick. This is not perfectly accurate but close enough to end up plus or minus 2ms in resulting rtp traces. So as long as the VM will expose the syscall down to the real monotonic clock rather than doing it's own soft timing technique you have a better chance for success. The other issue with VM is with vmware, the bridged networking mode seems to send 2 of every RTP packet to the channel resulting in garbled audio from the obvious timing issue introduced from too many packets. Anyway evaluating FS with a VM is a good way to get acquainted but, with all the money saved choosing FS, many can afford to buy it a nice 8 core box for it to live on and still have money left over to support the project or ClueCon 2009 this august ;) On Fri, Jan 16, 2009 at 2:17 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Speaking of networking... > > After timing that's the next "achilles heel" of RTP handing with > virtualization. > > Very, very few of these platforms were designed to handle massive > numbers of very small RTP packets. Everything from interrupt handling > on the actual ethernet adapter to getting the data into userspace > within the virtual instance is worrisome: > > http://www.xen.org/files/xensummit_4/NetworkIO_Santos.pdf > http://forum.openvz.org/index.php?t=msg&goto=11619& > > Interestingly enough the Xen paper makes it out to be really bad yet > the OpenVZ post praises Xen's performance. Without any real testing, > who knows? I just know that scaling 50pps per RTP stream (20ms > packetization) can be difficult enough on native hardware, let alone > [virtualization technology du jour]. > > On Thu, Jan 15, 2009 at 5:02 PM, Remko Kloosterman > wrote: > > Lot's of experience and suggestions here. Thanks. > > > > I believe it should be theoretically possible to have blip-free RTP > > streaming through the appliance. Most windows ethernet drivers allow for > > QoS packet scheduling. If the VMware network bridge driver honors this > > and syncs the buffers at 20ms frames (or whatever frame size applies) > > you should be able to schale up a bit and maintain low jitter. > > > > Anyone knows how the VMware network bridge exactly works? > > > > > > -----Oorspronkelijk bericht----- > > Van: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Gregory > > Boehnlein > > Verzonden: donderdag 15 januari 2009 21:37 > > Aan: freeswitch-users at lists.freeswitch.org > > Onderwerp: Re: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality > > > >> To the contrary, we have had quite good results in virtualized > >> environments and you don't really need timing that is that accurate to > > > >> make it work. > > > > If you don't handle RTP, I'm sure it is amazing. However, if you have to > > do voicemail, stream audio from the server or do any kind of actual > > time/latency/jitter sensitive processing, I don't care how much you tune > > your hypervisor, it's never going to scale. > > > >> We work quite well on amazon EC2 for example. There are 2 issues I > >> know about with vmware, 1 is you need to set a setting on the host to > >> extend somewhat sane clocks being available, the second is I have seen > > > >> issues with the bridged network adapter actually doubling up all > >> packets causing very strange issues, I suggest not using bridged > >> networking if you experience this. > > > > I've not seen this behavior on Vmware ESX 3.5u2. Maybe an issue on > > Vmware Server or Workstation? > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/b745ed42/attachment.html From astmac at stillnewt.org Fri Jan 16 11:12:56 2009 From: astmac at stillnewt.org (Martin Joseph) Date: Fri, 16 Jan 2009 11:12:56 -0800 Subject: [Freeswitch-users] Starting FS on OSX (10.4.11 PPC) In-Reply-To: References: Message-ID: <6DD3D54D-8B5F-477B-A5B1-F5CA433F6A5F@stillnewt.org> On Jan 16, 2009, at 8:09 AM, Ivan C Myrvold wrote: > I haven't tried using launchctl for FreeSWITCH. But when I saw your > post, I tried it out. I have no problem getting it to work: > > I make a file "org.freeswitch.freeswitch.plist" and save it to ~/ > Library/LaunchAgents with the following content: > > > "> > > > KeepAlive > > Label > org.freeswitch.freeswitch > Program > /usr/local/freeswitch/bin/freeswitch > RunAtLoad > > ServiceIPC > > > > > Then in Terminal.app, I do a "launchctl load ~/Library/LaunchAgents/ > org.freeswitch.freeswitch.plist" > > If you do the same command, but unload instead of load, it should stop > freeswitch. > > Does this work for you? Huh, I have been trying to do something similar, but putting the file in / System/Library/LaunchDaemons. Strangely, I tried your file right now (in LaunchDaemons) and freeswitch is started, but it doesn't respond to my devices trying to register? Weird. If I kill it and start it manually, it immediately responds and my devices register. Ideas? Thanks, Marty > > > Ivan > > Den 15. jan.. 2009 kl. 22:54 skrev Martin Joseph: > >> Hello again FreeSwitchers, >> >> I have built the 1.02 on 10.4.11(OSX) and had no problems with that. >> >> I have never been able to build from the SVN, but that is another >> story. >> >> Now that I have migrated to 1.02 I was wondering if I can get some >> help on a long standing issue I have with starting FS at boot. >> >> I am hoping to use Launchd which is the standard on OSX 10.4 and I >> attempted to cobble together a script, but haven't had great results. >> >> I did search for wiki entries on this, but haven't found any help >> with >> it. >> >> Ideas? >> Thanks, >> Marty >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From astmac at stillnewt.org Fri Jan 16 11:15:35 2009 From: astmac at stillnewt.org (Martin Joseph) Date: Fri, 16 Jan 2009 11:15:35 -0800 Subject: [Freeswitch-users] Starting FS on OSX In-Reply-To: <1c4b29080901160545o21caea3dl45e2360edc1d76c7@mail.gmail.com> References: <9B1AAAEE-DF37-442E-ACF5-F2E3078A0991@jerris.com> <02FDD555-C096-4F09-91DC-A9B57A44427F@stillnewt.org> <1c4b29080901160545o21caea3dl45e2360edc1d76c7@mail.gmail.com> Message-ID: On Jan 16, 2009, at 5:45 AM, Shido Xavier wrote: > Please specify Intel or PPC. Very good point and I had had that thought as well. I am on PPC 10.4.11. Thanks for any help or ideas. Marty > > > -Greg M. > > > On Fri, Jan 16, 2009 at 1:29 AM, Martin Joseph > wrote: >> >> On Jan 15, 2009, at 3:10 PM, Michael Jerris wrote: >> >>> Your build issue is with your autotools install, I have seen >>> issues if >>> you have ever installed any of the autotools from macports or fink. >> I have never used Fink or Macports so that isn't it. In fact the >> supposed statements made to the effect that FS will build from SVN >> fine on 10.4 with the latest available apple dev tools is quite wrong >> in my experience. I setup a virgin 10.4 and updated everything and >> had >> many complaints from FS about tool versions. >>> >>> If you want to build from svn you can run bootstrap on another box >>> (a >>> linux box perhaps) and then tar up that dir and move it to your mac. >> Huh, interesting. >>> >>> We pre-bootstrap the release tarballs which is why that is building >>> fine for you. >> Right, Thanks for all your efforts and an outstanding platform! >> >> Marty >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ivan at myrvold.org Fri Jan 16 12:54:44 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Fri, 16 Jan 2009 21:54:44 +0100 Subject: [Freeswitch-users] Starting FS on OSX (10.4.11 PPC) In-Reply-To: <6DD3D54D-8B5F-477B-A5B1-F5CA433F6A5F@stillnewt.org> References: <6DD3D54D-8B5F-477B-A5B1-F5CA433F6A5F@stillnewt.org> Message-ID: <1763DA6C-2AB7-4E3B-8673-A683C4491322@myrvold.org> I have chown the freeswitch directory to my user imyrvold, therefore I put it in ~/Library/LaunchDaemons. Do you run freeswitch as root, as you put it in /System/library/ LaunchDaemons? That directory should be reserved anyway for Apple's system tools. A better idea would be to put it in /Library/LaunchDaemons if you run it as root (but you should in my opinion run it as a normal user, as I have (almost) always done). Ivan Den 16. jan.. 2009 kl. 20:12 skrev Martin Joseph: > > On Jan 16, 2009, at 8:09 AM, Ivan C Myrvold wrote: > >> I haven't tried using launchctl for FreeSWITCH. But when I saw your >> post, I tried it out. I have no problem getting it to work: >> >> I make a file "org.freeswitch.freeswitch.plist" and save it to ~/ >> Library/LaunchAgents with the following content: >> >> >> > "> >> >> >> KeepAlive >> >> Label >> org.freeswitch.freeswitch >> Program >> /usr/local/freeswitch/bin/freeswitch >> RunAtLoad >> >> ServiceIPC >> >> >> >> >> Then in Terminal.app, I do a "launchctl load ~/Library/LaunchAgents/ >> org.freeswitch.freeswitch.plist" >> >> If you do the same command, but unload instead of load, it should >> stop >> freeswitch. >> >> Does this work for you? > > Huh, > > I have been trying to do something similar, but putting the file in / > System/Library/LaunchDaemons. > > Strangely, I tried your file right now (in LaunchDaemons) and > freeswitch is started, but it doesn't respond to my devices trying to > register? > > Weird. > > If I kill it and start it manually, it immediately responds and my > devices register. > > Ideas? > Thanks, > > Marty >> >> >> Ivan >> >> Den 15. jan.. 2009 kl. 22:54 skrev Martin Joseph: >> >>> Hello again FreeSwitchers, >>> >>> I have built the 1.02 on 10.4.11(OSX) and had no problems with that. >>> >>> I have never been able to build from the SVN, but that is another >>> story. >>> >>> Now that I have migrated to 1.02 I was wondering if I can get some >>> help on a long standing issue I have with starting FS at boot. >>> >>> I am hoping to use Launchd which is the standard on OSX 10.4 and I >>> attempted to cobble together a script, but haven't had great >>> results. >>> >>> I did search for wiki entries on this, but haven't found any help >>> with >>> it. >>> >>> Ideas? >>> Thanks, >>> Marty >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From willbelair at yahoo.com Fri Jan 16 13:09:01 2009 From: willbelair at yahoo.com (Will Smith) Date: Fri, 16 Jan 2009 13:09:01 -0800 (PST) Subject: [Freeswitch-users] Dialing Out Problem via Gateway Message-ID: <927434.33172.qm@web53611.mail.re2.yahoo.com> Hi, I got a strange problem that I don't really understand, and I hope that you could give me some hint how to fix that: ? When I dial out through a gateway that is defined in the sip_profiles/external?, (The xml file is simple as below. )? I cannot talk or hear from the other end. But when I put the line on hold, two ends can hear music, and when open the line again, this time 2 ends can hear and talk. Is there any where that I can fix this problem? Thank you ? ? ??? ??? ??? ??? ??? ??? ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/09fbf97b/attachment.html From brian at freeswitch.org Fri Jan 16 13:13:00 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 16 Jan 2009 15:13:00 -0600 Subject: [Freeswitch-users] Dialing Out Problem via Gateway In-Reply-To: <927434.33172.qm@web53611.mail.re2.yahoo.com> References: <927434.33172.qm@web53611.mail.re2.yahoo.com> Message-ID: <1FBEB30E-9964-4332-AF52-A5F03648BCC3@freeswitch.org> Can you detail your problem a bit more? /b On Jan 16, 2009, at 3:09 PM, Will Smith wrote: > Hi, > I got a strange problem that I don't really understand, and I hope > that you could give me some hint how to fix that: > > When I dial out through a gateway that is defined in the > sip_profiles/external , (The xml file is simple as below. ) I > cannot talk or hear from the other end. But when I put the line on > hold, two ends can hear music, and when open the line again, this > time 2 ends can hear and talk. Is there any where that I can fix > this problem? Thank you > > > > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/c59b2895/attachment.html From msc at freeswitch.org Fri Jan 16 13:30:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 16 Jan 2009 13:30:58 -0800 Subject: [Freeswitch-users] Dialing Out Problem via Gateway In-Reply-To: <1FBEB30E-9964-4332-AF52-A5F03648BCC3@freeswitch.org> References: <927434.33172.qm@web53611.mail.re2.yahoo.com> <1FBEB30E-9964-4332-AF52-A5F03648BCC3@freeswitch.org> Message-ID: <87f2f3b90901161330g284cc7dcp92c211a3ed7d733e@mail.gmail.com> A SIP trace would be extremely helpful. http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch#Enabling_SIP.2FSofia_Tracing -MC On Fri, Jan 16, 2009 at 1:13 PM, Brian West wrote: > Can you detail your problem a bit more? > /b > On Jan 16, 2009, at 3:09 PM, Will Smith wrote: > > Hi, > I got a strange problem that I don't really understand, and I hope that you > could give me some hint how to fix that: > > When I dial out through a gateway that is defined in the > sip_profiles/external , (The xml file is simple as below. ) I cannot talk > or hear from the other end. But when I put the line on hold, two ends can > hear music, and when open the line again, this time 2 ends can hear and > talk. Is there any where that I can fix this problem? Thank you > > > > > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From willbelair at yahoo.com Fri Jan 16 13:30:42 2009 From: willbelair at yahoo.com (Will Smith) Date: Fri, 16 Jan 2009 13:30:42 -0800 (PST) Subject: [Freeswitch-users] Dialing Out Problem via Gateway Message-ID: <399787.48666.qm@web53606.mail.re2.yahoo.com> Thank you Brian, ? The problem is very simple, I or the other party cannot hear each other when I first dial and the other party picks up the phone. We hear the phone ring, the other end picks up the phone says something, but I cannot hear - nothing, even static. Same thing happen on my end, I say something, and the other end do? not hear a thing. When I? put the call on hold,? the other end can hear music on hold. When I take the call back, now we can talk. Something does not go through when the other end picks up the call. This is the extension in the dialplan/default.xml ? if (typeof YAHOO == "undefined") { var YAHOO = {}; } YAHOO.Shortcuts = YAHOO.Shortcuts || {}; YAHOO.Shortcuts.hasSensitiveText = false; YAHOO.Shortcuts.sensitivityType = []; YAHOO.Shortcuts.doUlt = false; YAHOO.Shortcuts.location = "us"; YAHOO.Shortcuts.document_id = 0; YAHOO.Shortcuts.document_type = ""; YAHOO.Shortcuts.document_title = "t"; YAHOO.Shortcuts.document_publish_date = ""; YAHOO.Shortcuts.document_author = "willbelair at yahoo.com"; YAHOO.Shortcuts.document_url = ""; YAHOO.Shortcuts.document_tags = ""; YAHOO.Shortcuts.document_language = "english"; YAHOO.Shortcuts.annotationSet = { "lw_1232141334_0": { "text": "9054516117", "extended": 0, "startchar": 391, "endchar": 400, "start": 391, "end": 400, "extendedFrom": "", "predictedCategory": "", "predictionProbability": "0", "weight": 1, "relScore": 0, "type": ["shortcuts:/us/instance/identifier/fedex_tracking"], "category": ["IDENTIFIER"], "wikiId": "", "relatedWikiIds": [], "relatedEntities": [], "showOnClick": [], "context": "", "metaData": { "verified": "false", "visible": "true" } } }; YAHOO.Shortcuts.headerID = "a9059b1f35336b4363b0c75035b61d07"; ??? ????? ????? ????? ????? ???? ?? --- On Fri, 1/16/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] Dialing Out Problem via Gateway To: freeswitch-users at lists.freeswitch.org Date: Friday, January 16, 2009, 1:13 PM Can you detail your problem a bit more? /b On Jan 16, 2009, at 3:09 PM, Will Smith wrote: Hi, I got a strange problem that I don't really understand, and I hope that you could give me some hint how to fix that: ? When I dial out through a gateway that is defined in the sip_profiles/external?, (The xml file is simple as below. )? I cannot talk or hear from the other end. But when I put the line on hold, two ends can hear music, and when open the line again, this time 2 ends can hear and talk. Is there any where that I can fix this problem? Thank you ? ? ??? ??? ??? ??? ??? ??? ? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/ab324148/attachment-0001.html From jsacksteder at gmail.com Fri Jan 16 13:33:43 2009 From: jsacksteder at gmail.com (jeff sacksteder) Date: Fri, 16 Jan 2009 16:33:43 -0500 Subject: [Freeswitch-users] Vmware voice quality In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC601479A7F@srvmtel.office.mtel.nl> References: <11372C8B9E603F4FACDE6AB18256DEC601479A7F@srvmtel.office.mtel.nl> Message-ID: <51c8a7be0901161333v6b1830d2gb38d1114e5ac05c8@mail.gmail.com> This is a known issue with all virtualization solutions. The realtime clocks inside the Virtual Machine jitter quite a bit which causes havoc with the udp media streams. I have never heard of someone using any VoIP product inside a VM and being happy with the result. To the best of my knowledge, all the current VM products suffer from this - Hyper-V, Xen and VMware, VMWare being the worst. Xen shares access to the system clock with paravirtualized guests directly, so you might find that to be somewhat better than VMWare. More here - http://www.vmware.com/pdf/vmware_timekeeping.pdf The performance of the server is irrelevant. Google for UDP jitter in combination with VMWare. On Thu, Jan 15, 2009 at 12:08 PM, Remko Kloosterman wrote: > Hello Ken, hello all, > > I just read about the FreeSWITCH VMware applicance. I'm curious about your > experiences with the audio quality on VMWare, so here's a new thread. > > I've installed freeswitch on VMware Server for Windows. The IVR audio always > plays choppy, while the server itself has no performance issues. The same > poor voice quality also goes for Asterisk or Yate, even on a very fast > VMware ESX system. > > Did you experience the same and/or do you have pointers on how to > troubleshoot and fix this? > > Thanks, > > Remko > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Fri Jan 16 13:41:54 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 16 Jan 2009 15:41:54 -0600 Subject: [Freeswitch-users] Dialing Out Problem via Gateway In-Reply-To: <399787.48666.qm@web53606.mail.re2.yahoo.com> References: <399787.48666.qm@web53606.mail.re2.yahoo.com> Message-ID: <6F37E37B-23FA-42C9-B47F-EB0C7AB1EE21@freeswitch.org> NAT involved? /b On Jan 16, 2009, at 3:30 PM, Will Smith wrote: > Thank you Brian, > > The problem is very simple, I or the other party cannot hear each > other when I first dial and the other party picks up the phone. We > hear the phone ring, the other end picks up the phone says > something, but I cannot hear - nothing, even static. Same thing > happen on my end, I say something, and the other end do not hear a > thing. When I put the call on hold, the other end can hear music > on hold. When I take the call back, now we can talk. Something does > not go through when the other end picks up the call. > This is the extension in the dialplan/default.xml > > > > data="effective_caller_id_number=12223334444"/> > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/270e2bd9/attachment.html From willbelair at yahoo.com Fri Jan 16 13:57:20 2009 From: willbelair at yahoo.com (Will Smith) Date: Fri, 16 Jan 2009 13:57:20 -0800 (PST) Subject: [Freeswitch-users] Dialing Out Problem via Gateway In-Reply-To: <6F37E37B-23FA-42C9-B47F-EB0C7AB1EE21@freeswitch.org> Message-ID: <351539.66195.qm@web53611.mail.re2.yahoo.com> Well, if NAT involved, why did I get through after I put the call on hold and take the call back. I am getting the SIP trace, hope that will show something. Thank you all --- On Fri, 1/16/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] Dialing Out Problem via Gateway To: freeswitch-users at lists.freeswitch.org Date: Friday, January 16, 2009, 1:41 PM NAT involved? /b On Jan 16, 2009, at 3:30 PM, Will Smith wrote: Thank you Brian, ? The problem is very simple, I or the other party cannot hear each other when I first dial and the other party picks up the phone. We hear the phone ring, the other end picks up the phone says something, but I cannot hear - nothing, even static. Same thing happen on my end, I say something, and the other end do? not hear a thing. When I? put the call on hold,? the other end can hear music on hold. When I take the call back, now we can talk. Something does not go through when the other end picks up the call. This is the extension in the dialplan/default.xml ? ??? ????? ????? ????? ????? ???? ?? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/f1f07451/attachment.html From brian at freeswitch.org Fri Jan 16 14:04:36 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 16 Jan 2009 16:04:36 -0600 Subject: [Freeswitch-users] Dialing Out Problem via Gateway In-Reply-To: <351539.66195.qm@web53611.mail.re2.yahoo.com> References: <351539.66195.qm@web53611.mail.re2.yahoo.com> Message-ID: <25F0E05B-3A1D-40B4-897A-93D100BFF5E9@freeswitch.org> I would suspect the NAT wasn't punching holes or lied. :) /b On Jan 16, 2009, at 3:57 PM, Will Smith wrote: > Well, if NAT involved, why did I get through after I put the call on > hold and take the call back. I am getting the SIP trace, hope that > will show something. > Thank you all -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090116/b932444f/attachment.html From andrew at hijacked.us Fri Jan 16 15:35:13 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 16 Jan 2009 18:35:13 -0500 Subject: [Freeswitch-users] [ANN] Spice SoftPhone, a softphone GUI for FreeSWITCH Message-ID: <20090116233513.GC20354@hijacked.us> I'd like to announce the first beta release of a cross-platform ruby/tk GUI for using FreeSWITCH like a soft-phone (using mod_portaudio). It's not particularly fancy, but I needed a cross platform softphone with good voice quality that was debuggable and didn't have a ton of features to confuse the users. I couldn't find one so we built one. I've got some sparse documentation up at: http://opencsm.org/wiki/index.php/Spice_SoftPhone And you can download it from http://opencsm.org/download . It's under the MPL and I've been cleared to re-licence my other FreeSWITCH related projects under the MPL too. I've tested it on Windows, FreeBSD, Solaris and OSX (it used to work on linux, I assume it still does). Comments/complaints/bugreports welcome. It's definitely still got some rough spots (I don't think it'll run without a controlling terminal, for example), but we're going to be polishing it up and hopefully putting it in production here in the next few weeks to replace a very buggy closed-source phone we've had to endure far too long. Please download it if you're interested, the download count helps us continue working on this kind of stuff :) Andrew - opencsm.org From msc at freeswitch.org Fri Jan 16 15:53:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 16 Jan 2009 15:53:18 -0800 Subject: [Freeswitch-users] [ANN] Spice SoftPhone, a softphone GUI for FreeSWITCH In-Reply-To: <20090116233513.GC20354@hijacked.us> References: <20090116233513.GC20354@hijacked.us> Message-ID: <87f2f3b90901161553x5fb1aa2dt85e652226aa9d246@mail.gmail.com> Andrew, On behalf of the OSS telephony community, and particularly the FreeSWITCH community, many thanks to you! We appreciate all the stuff you've given back. Thanks for showing the true Open Source spirit! -MC On Fri, Jan 16, 2009 at 3:35 PM, Andrew Thompson wrote: > I'd like to announce the first beta release of a cross-platform ruby/tk > GUI for using FreeSWITCH like a soft-phone (using mod_portaudio). It's > not particularly fancy, but I needed a cross platform softphone with > good voice quality that was debuggable and didn't have a ton of > features to confuse the users. I couldn't find one so we built one. > > I've got some sparse documentation up at: > > http://opencsm.org/wiki/index.php/Spice_SoftPhone > > And you can download it from http://opencsm.org/download . It's under > the MPL and I've been cleared to re-licence my other FreeSWITCH related > projects under the MPL too. I've tested it on Windows, FreeBSD, Solaris > and OSX (it used to work on linux, I assume it still does). > > Comments/complaints/bugreports welcome. It's definitely still got some > rough spots (I don't think it'll run without a controlling terminal, for > example), but we're going to be polishing it up and hopefully putting it > in production here in the next few weeks to replace a very buggy > closed-source phone we've had to endure far too long. > > Please download it if you're interested, the download count helps us > continue working on this kind of stuff :) > > Andrew - opencsm.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Jan 16 15:54:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 16 Jan 2009 15:54:41 -0800 Subject: [Freeswitch-users] spidermonkey problems In-Reply-To: <07F43D7C-3CE2-4A55-819B-E8F1095C5C8B@jerris.com> References: <191c3a030901151548w7504e2a0j5650449e20eff557@mail.gmail.com> <07F43D7C-3CE2-4A55-819B-E8F1095C5C8B@jerris.com> Message-ID: <87f2f3b90901161554u47b652e3yeadbaf6d2ef2090@mail.gmail.com> FYI, I opened http://jira.freeswitch.org/browse/MODLANG-97 on this issue. -MC On Fri, Jan 16, 2009 at 9:03 AM, Michael Jerris wrote: > All long running non js code should be wrapped in the suspend/resume gc > stuff. For example: > cb_state.ret = BOOLEAN_TO_JSVAL(JS_FALSE); > cb_state.saveDepth = JS_SuspendRequest(cx); > args.input_callback = dtmf_func; > args.buf = bp; > args.buflen = len; > switch_ivr_sleep(jss->session, ms, sync, &args); > JS_ResumeRequest(cx, cb_state.saveDepth); > I think this is your issue. Can you please file a bug on jira for this > issue (even better with a patch) > Mike > > > On Jan 16, 2009, at 5:54 AM, Jonas Gauffin wrote: > > I've found the problem. one js thread wait in socket.read > (mod_spidermonkey_socket) on data. > That caller have hangup, which means that the garbage collector waits on it > to close. > > All new javascript sessions waits in JS_AWAIT_GC_DONE for the garbage > collector to be done before proceeding (which means that all new javascript > calls don't do anything after being launched). > My server will not send anything until an agent gets free or the session > hangs up (detects it through the event socket). And the event socket will > not send that the session has been hangup until the socket have received > anything (and the script can exit). So it's kind of deadlock between my > server and the spidermonkey_socket. > Is it possible to add an option to socket.read to make it abort if the > session have been closed? I know that I wrote mod_spidermonkey_socket from > the start, but I can't figure out how to do it. > Will new sessions always wait on old ones to be garbage collected properly? > For instance, what happens if a script have a lenghty post process after > caller have hang up? > On Fri, Jan 16, 2009 at 9:38 AM, Jonas Gauffin > wrote: >> >> I've got a loop, but the first thing checked in each iteration is if >> session.ready() returns false (and in that case exit the loop). >> I do create sessions in the script: create, try to originate to a >> destination and then finally bridge together the caller and the new session. >> I'll try to give you more details during the day. >> On Fri, Jan 16, 2009 at 12:48 AM, Anthony Minessale >> wrote: >>> >>> do you have any loops in your code that might not check for >>> session.ready() in a exit when its not true. >>> >>> The symptoms you posted would be consistent with held readlocks so if you >>> got a gcore (or windows equiv) of the process you might be able to see what >>> threads where doing what to hang on to the read lock. >>> >>> also are you creating sessions in the script then executing app with >>> them, beware of this because the thread of the script is used to execute >>> apps on a session created that way and not the session thread. >>> >>> >>> >>> >>> On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin >>> wrote: >>>> >>>> Hello >>>> I got problems with hanging spidermonkey sessions and need some advice >>>> on how to debug them. >>>> I've made a javascript queue application that uses >>>> mod_spidermonkey_socket. It works fine for a while, >>>> but after some calls I noticed that calls didnt get transferred to >>>> agents. The reason was that earlier >>>> calls had not been terminated properly. >>>> freeswitch at test1> hupall >>>> 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 >>>> switch_core_session_hupall() Giving up with 8 sessions remaining >>>> API CALL [hupall()] output: >>>> +OK hangup all channels with cause MANAGER_REQUEST >>>> >>>> freeswitch at test1> show calls >>>> API CALL [show(calls)] output: >>>> 0 total. >>>> >>>> As you can see, 8 sessions are alive, but none of them are listed as >>>> calls. What kind of logs should I turn on to see what is happening with >>>> those sessions? >>>> Thanks, >>>> Jonas >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 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 >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From scott.ellis at novatex.com.au Sat Jan 17 01:51:14 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Sat, 17 Jan 2009 20:51:14 +1100 Subject: [Freeswitch-users] OpenZap detect tones question Message-ID: <4971AA12.10706@novatex.com.au> Quick question, when specifying a "detect-busy" tone in the tones.conf file - is the cadence used? (The US examples to not have cadence) In the tests I have does it does not seem to be. This is a problem in Australia, as we have managed to have our busy tone 425Hz, 375ms on 375ms off, also in our dial tone 400+425+450. So when I go to dial a call, I often get the Zap channel hanging up again as it thinks the line is busy. Scott From regs at kinetix.gr Sat Jan 17 07:29:48 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Sat, 17 Jan 2009 17:29:48 +0200 Subject: [Freeswitch-users] Outbound call - choose profile In-Reply-To: <191c3a030901160915n250d791m4451815341b8d60a@mail.gmail.com> References: <497082DC.8020904@kinetix.gr> <49709A5B.2060806@freeswitch.org> <49709F7B.4030905@kinetix.gr> <191c3a030901160915n250d791m4451815341b8d60a@mail.gmail.com> Message-ID: <4971F96C.6050406@kinetix.gr> Yes, I am aware of that fact : insteaad of the pre-configured internal/external profiles I coulf have myprofile1/myprofile2. But since I am using the pre-configured profiles (without changing the ports) and since the internal profile initiated the parsing I would expect that all the gateways defined in my directory user's xml files would belong to the internal profile (which is preconfigured to use port 5060). But when I am capturing the packets sent to that gateway I can see that the port used from FS is 5080 (which is the preconfigured port of the internal profile). Anthony Minessale wrote: > all gateways are properties of a profile > so even in the parse domain way of handling gateways whatever profile > parsed the domain > will be the owner of all the gateways discovered from the domain. > > FYI, > > The internal profile has nothing to do with users or anything else. > You are confusing the default examples for absolute fact. > > On Fri, Jan 16, 2009 at 8:53 AM, Apostolos Pantsiopoulos > > wrote: > > Raymond Chandler wrote: > > Apostolos Pantsiopoulos wrote: > >> When I am using the following method to place a call from the > dialplan : > >> > >> sofia/gateway// > >> > >> how do I tell FS which profile to use (as in the > >> sofia// method?) > >> > >> I am asking that because all my calls to my declared use > >> the 5080 port, > >> and I want them to use the 5060 port. Is there a way to configure a > >> to use > >> a specific profile when making outbound calls? > >> > >> > > the gateway should always use the profile from which it was > included. > > so, for instance, if you include gw1 from internal.xml, then gw1 > > should always use the internal profile > > > > -Ray > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > I am using the directory to define my gateways. Specifically, because > this certain gateway > is related to a specific user I am including this gateway in the > user's > xml file (in the dieractory). > Then I use : > > > > in my internal profile to let FS "parse" the gateways that I have > declared in the directory. > > So one would expect that the gateways declared in the user's file > would > belong to the internal > profile. Yet, when I am using the sofia/gateway// > notation to send a call > through this gateway the SIP packets get send from the 5080 port > (which > is my external profile's port) > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090117/996bc49d/attachment.html From brian at freeswitch.org Sat Jan 17 07:50:43 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 17 Jan 2009 09:50:43 -0600 Subject: [Freeswitch-users] Outbound call - choose profile In-Reply-To: <4971F96C.6050406@kinetix.gr> References: <497082DC.8020904@kinetix.gr> <49709A5B.2060806@freeswitch.org> <49709F7B.4030905@kinetix.gr> <191c3a030901160915n250d791m4451815341b8d60a@mail.gmail.com> <4971F96C.6050406@kinetix.gr> Message-ID: The external profile is setup to parse the gateways... internal is setup to parse the domains and apply an alias to the internal profile for all domains in the directory. On internal you have: Then on external you have: Notice this tells internal to add an alias for every domain to internal, while external is told to parse for gateways. /b On Jan 17, 2009, at 9:29 AM, Apostolos Pantsiopoulos wrote: > Yes, I am aware of that fact : insteaad of the pre-configured > internal/external profiles I coulf have myprofile1/myprofile2. > But since I am using the pre-configured profiles (without changing > the ports) > and since the internal profile initiated the parsing I would expect > that all the gateways defined in my directory user's xml files would > belong to the internal profile (which is preconfigured to use port > 5060). > But when I am capturing the packets sent to that gateway I can see > that > the port used from FS is 5080 (which is the preconfigured port of > the internal profile). From helmut.kuper at ewetel.de Sat Jan 17 08:16:46 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Sat, 17 Jan 2009 17:16:46 +0100 Subject: [Freeswitch-users] Q931 decoding Message-ID: <4972046E.8020102@ewetel.de> Hello, if anyone is interessted in a Q931 decoder for FS, I found a way to get FS's Q931 debug hexdumps decoded in wireshark. It's quite simple and mainly about encapsulating each Q931 dump into a TPKT paket and that one in a TCP/IP paket and to put all TCP/IP pakets into a pcap file. regards helmut From brian at freeswitch.org Sat Jan 17 08:27:13 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 17 Jan 2009 10:27:13 -0600 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <4972046E.8020102@ewetel.de> References: <4972046E.8020102@ewetel.de> Message-ID: <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> Any automated way of doing this? Open a Jira on it.. it would be neat to be able to save the q931 into a pcap file.... maybe libpcap can help? /b On Jan 17, 2009, at 10:16 AM, Helmut Kuper wrote: > Hello, > > if anyone is interessted in a Q931 decoder for FS, I found a way to > get > FS's Q931 debug hexdumps decoded in wireshark. It's quite simple and > mainly about encapsulating each Q931 dump into a TPKT paket and that > one > in a TCP/IP paket and to put all TCP/IP pakets into a pcap file. > > > regards > helmut From helmut.kuper at ewetel.de Sat Jan 17 08:58:12 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Sat, 17 Jan 2009 17:58:12 +0100 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> Message-ID: <49720E24.3050806@ewetel.de> Hi Brian, currently it's a simple perl script which greps all Q931 hexdumps from FS logfile converting them to a TPKT packet, and writing those to a separate local file (wireshark's text2pcap file format). This 1st step is easy to put right into FS ozmod_isdn debug code. The 2nd step is to convert the new file into a .pcap file with adding TCP/IP dummy packet in front of each TPKT packet. This is done via test2pcap. This .pcap file is ready to be decoded by wireshark. I put both steps into a little shell script and added a 3rd step to get those .pcap file emailed from the Server to my Desktop. regards helmut Brian West schrieb: > Any automated way of doing this? Open a Jira on it.. it would be neat > to be able to save the q931 into a pcap file.... maybe libpcap can help? > > /b From brian at freeswitch.org Sat Jan 17 09:20:41 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 17 Jan 2009 11:20:41 -0600 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <49720E24.3050806@ewetel.de> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> Message-ID: <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> Maybe open a jira with this info? Maybe it can all be done as a one step process in the ozmod_isdn ;) /b On Jan 17, 2009, at 10:58 AM, Helmut Kuper wrote: > Hi Brian, > > currently it's a simple perl script which greps all Q931 hexdumps from > FS logfile converting them to a TPKT packet, and writing those to a > separate local file (wireshark's text2pcap file format). This 1st step > is easy to put right into FS ozmod_isdn debug code. > The 2nd step is to convert the new file into a .pcap file with adding > TCP/IP dummy packet in front of each TPKT packet. This is done via > test2pcap. This .pcap file is ready to be decoded by wireshark. > > I put both steps into a little shell script and added a 3rd step to > get > those .pcap file emailed from the Server to my Desktop. > > regards > helmut From astmac at stillnewt.org Sat Jan 17 10:15:57 2009 From: astmac at stillnewt.org (Martin Joseph) Date: Sat, 17 Jan 2009 10:15:57 -0800 Subject: [Freeswitch-users] Starting FS on OSX (10.4.11 PPC) In-Reply-To: <1763DA6C-2AB7-4E3B-8673-A683C4491322@myrvold.org> References: <6DD3D54D-8B5F-477B-A5B1-F5CA433F6A5F@stillnewt.org> <1763DA6C-2AB7-4E3B-8673-A683C4491322@myrvold.org> Message-ID: <060967E3-D7D9-4E8B-8817-7378FCD353B7@stillnewt.org> On Jan 16, 2009, at 12:54 PM, Ivan C Myrvold wrote: > I have chown the freeswitch directory to my user imyrvold, therefore > I put it in ~/Library/LaunchDaemons. > Do you run freeswitch as root, as you put it in /System/library/ > LaunchDaemons? That directory should be reserved anyway for Apple's > system tools. > A better idea would be to put it in /Library/LaunchDaemons if you run > it as root (but you should in my opinion run it as a normal user, as I > have (almost) always done). Yes, I have been running it as root (blush). Any idea why it might start, but not except registrations? That seems strange to me. Thanks for the help (again), Marty > > > Ivan > > Den 16. jan.. 2009 kl. 20:12 skrev Martin Joseph: > >> >> On Jan 16, 2009, at 8:09 AM, Ivan C Myrvold wrote: >> >>> I haven't tried using launchctl for FreeSWITCH. But when I saw your >>> post, I tried it out. I have no problem getting it to work: >>> >>> I make a file "org.freeswitch.freeswitch.plist" and save it to ~/ >>> Library/LaunchAgents with the following content: >>> >>> >>> >> "> >>> >>> >>> KeepAlive >>> >>> Label >>> org.freeswitch.freeswitch >>> Program >>> /usr/local/freeswitch/bin/freeswitch >>> RunAtLoad >>> >>> ServiceIPC >>> >>> >>> >>> >>> Then in Terminal.app, I do a "launchctl load ~/Library/LaunchAgents/ >>> org.freeswitch.freeswitch.plist" >>> >>> If you do the same command, but unload instead of load, it should >>> stop >>> freeswitch. >>> >>> Does this work for you? >> >> Huh, >> >> I have been trying to do something similar, but putting the file in / >> System/Library/LaunchDaemons. >> >> Strangely, I tried your file right now (in LaunchDaemons) and >> freeswitch is started, but it doesn't respond to my devices trying to >> register? >> >> Weird. >> >> If I kill it and start it manually, it immediately responds and my >> devices register. >> >> Ideas? >> Thanks, >> >> Marty >>> >>> >>> Ivan >>> >>> Den 15. jan.. 2009 kl. 22:54 skrev Martin Joseph: >>> >>>> Hello again FreeSwitchers, >>>> >>>> I have built the 1.02 on 10.4.11(OSX) and had no problems with >>>> that. >>>> >>>> I have never been able to build from the SVN, but that is another >>>> story. >>>> >>>> Now that I have migrated to 1.02 I was wondering if I can get some >>>> help on a long standing issue I have with starting FS at boot. >>>> >>>> I am hoping to use Launchd which is the standard on OSX 10.4 and I >>>> attempted to cobble together a script, but haven't had great >>>> results. >>>> >>>> I did search for wiki entries on this, but haven't found any help >>>> with >>>> it. >>>> >>>> Ideas? >>>> Thanks, >>>> Marty >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andresmartinochoa at gmail.com Sat Jan 17 11:14:14 2009 From: andresmartinochoa at gmail.com (=?ISO-8859-1?Q?Andr=E9s_Mart=EDn_=2D_martyn?=) Date: Sat, 17 Jan 2009 14:14:14 -0500 Subject: [Freeswitch-users] [FS-es] Spanish Freeswitch community Message-ID: <8c1b00b30901171114u64d8942cyb8d05c4efae07e85@mail.gmail.com> Hi. I think that is a good idea expan the freeswitch project arround the wolrd, btw this site is specially to hispan community [1], and the irc channel is #freeswitch-es :D. [1] http://www.freeswitch.es Regards. -- Andr?s Mart?n Ochoa; passport: andresmartin at linuxmail.org; Linux Registered User #436420; Asterisk User Number: 1000; PBX: (57) 1 578 20 30; Ext: 106 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090117/8626fbc1/attachment.html From msc at freeswitch.org Sat Jan 17 12:44:26 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 17 Jan 2009 12:44:26 -0800 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> Message-ID: Guys this is awesome! Helmut, if you need any help with jira just let me know. -MC Sent from my iPhone On Jan 17, 2009, at 9:20 AM, Brian West wrote: > Maybe open a jira with this info? Maybe it can all be done as a one > step process in the ozmod_isdn ;) > > /b > > On Jan 17, 2009, at 10:58 AM, Helmut Kuper wrote: > >> Hi Brian, >> >> currently it's a simple perl script which greps all Q931 hexdumps >> from >> FS logfile converting them to a TPKT packet, and writing those to a >> separate local file (wireshark's text2pcap file format). This 1st >> step >> is easy to put right into FS ozmod_isdn debug code. >> The 2nd step is to convert the new file into a .pcap file with adding >> TCP/IP dummy packet in front of each TPKT packet. This is done via >> test2pcap. This .pcap file is ready to be decoded by wireshark. >> >> I put both steps into a little shell script and added a 3rd step to >> get >> those .pcap file emailed from the Server to my Desktop. >> >> regards >> helmut > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jmesquita at gmail.com Sat Jan 17 16:06:50 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 17 Jan 2009 22:06:50 -0200 Subject: [Freeswitch-users] [ANN] Spice SoftPhone, a softphone GUI for FreeSWITCH In-Reply-To: <20090116233513.GC20354@hijacked.us> References: <20090116233513.GC20354@hijacked.us> Message-ID: <79F15BF3-E8C0-44C5-B4D9-5D6FCFB7DA3A@gmail.com> Andrew, if you are interested, you could check out http://www.pjsip.org These guys have build a great lib that runs multiplatform on top o PA as well and are _REALLY_ small footprint. Perfect for a client, right? Thanks, Mesquita On Jan 16, 2009, at 9:35 PM, Andrew Thompson wrote: > I'd like to announce the first beta release of a cross-platform ruby/ > tk > GUI for using FreeSWITCH like a soft-phone (using mod_portaudio). It's > not particularly fancy, but I needed a cross platform softphone with > good voice quality that was debuggable and didn't have a ton of > features to confuse the users. I couldn't find one so we built one. > > I've got some sparse documentation up at: > > http://opencsm.org/wiki/index.php/Spice_SoftPhone > > And you can download it from http://opencsm.org/download . It's under > the MPL and I've been cleared to re-licence my other FreeSWITCH > related > projects under the MPL too. I've tested it on Windows, FreeBSD, > Solaris > and OSX (it used to work on linux, I assume it still does). > > Comments/complaints/bugreports welcome. It's definitely still got some > rough spots (I don't think it'll run without a controlling terminal, > for > example), but we're going to be polishing it up and hopefully > putting it > in production here in the next few weeks to replace a very buggy > closed-source phone we've had to endure far too long. > > Please download it if you're interested, the download count helps us > continue working on this kind of stuff :) > > Andrew - opencsm.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Sat Jan 17 16:25:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Sat, 17 Jan 2009 16:25:36 -0800 Subject: [Freeswitch-users] Changing internal profile In-Reply-To: <0427A651-04AA-486C-A4C5-F6D3A2C2309E@freeswitch.org> References: <0427A651-04AA-486C-A4C5-F6D3A2C2309E@freeswitch.org> Message-ID: <87f2f3b90901171625x743cbe90pb48fc130c66a62f5@mail.gmail.com> On Sat, Jan 3, 2009 at 10:21 AM, Brian West wrote: > The detected domain is 212.235.180.41 which caused this problem. > > You have two options here. > > Since we detected and set your default domain to 212.235.180.41 on > start up... you're registering to the 192.168.0.1 ip aka the internal > interface. > > The inbound register packet has: > > From: > > We take the part before the @ aka the username, then we take the part > after the @ aka the domain name. > > FreeSWITCH will then look thru your directory looking for domain which > in this case is 192.168.0.1 > which it can't find because we detected your public IP and set it up > as 212.235.180.41. > So the error message is telling you that you do not have a domain > called 192.168.0.1 with a user 1000 in it. > > So what you have to do here is understand that SIP like email works on > the concept of domains. user at host. > > A few things you need to know are this: > > (SOMEONE WIKIFY THIS PLEASE and expand on it. Find me on IRC if you > have questions) I've put this stuff on the Sofia wiki page: http://wiki.freeswitch.org/wiki/Sofia#Using_A_Single_Domain_For_All_Registrations I'm sure it could use some fleshing out and input from those who've actually used it in a real environment. -MC > > sofia profile params: > challenge-realm: > > (default > configuration uses auto_from) > > Choose the realm challenge key. Default is auto_to if not set. > > auto_from - uses the from field as the value for the sip realm. > auto_to - uses the to field as the value for the sip realm. > - you can input any value to use for the sip realm. > > force-register-domain: > > > > This will force the profile to ignore the domain in the to or from > packet and force > it to the value listed here for this param. > > This will store the info into the database with the user@ in sip packet> > > force-register-db-domain: > > > > This will work in conjunction with force-register-domain so that the > forced domain > is stored in the database also. > > > > ATTENTION HERE IS WHAT YOU SHOULD DO: > > So what I recommend for you is to open up internal.xml and uncomment > the force-register-domain > and force-register-db-domain params on the internal profile. Also > make sure internal.xml has > sip-ip and rtp-ip set to 192.168.0.1 and make sure your phones > register to 192.168.0.1. > > /b > > On Jan 3, 2009, at 11:46 AM, kriko wrote: > >> I changed exactly what I wrote in previous mails. >> Everything else is default. >> >> Is there anything else to change? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ivan at myrvold.org Sun Jan 18 04:15:50 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Sun, 18 Jan 2009 13:15:50 +0100 Subject: [Freeswitch-users] Starting FS on OSX (10.4.11 PPC) In-Reply-To: <060967E3-D7D9-4E8B-8817-7378FCD353B7@stillnewt.org> References: <6DD3D54D-8B5F-477B-A5B1-F5CA433F6A5F@stillnewt.org> <1763DA6C-2AB7-4E3B-8673-A683C4491322@myrvold.org> <060967E3-D7D9-4E8B-8817-7378FCD353B7@stillnewt.org> Message-ID: <7508A552-D788-442C-BF6C-CEA304FAD47C@myrvold.org> Den 17. jan.. 2009 kl. 19:15 skrev Martin Joseph: > > On Jan 16, 2009, at 12:54 PM, Ivan C Myrvold wrote: > >> I have chown the freeswitch directory to my user imyrvold, therefore >> I put it in ~/Library/LaunchDaemons. >> Do you run freeswitch as root, as you put it in /System/library/ >> LaunchDaemons? That directory should be reserved anyway for Apple's >> system tools. >> A better idea would be to put it in /Library/LaunchDaemons if you run >> it as root (but you should in my opinion run it as a normal user, >> as I >> have (almost) always done). > > Yes, I have been running it as root (blush). Any idea why it might > start, but not except registrations? That seems strange to me. Do you have any messages in Console that can explain it? Ivan > > > Thanks for the help (again), > Marty > > >> >> >> Ivan >> >> Den 16. jan.. 2009 kl. 20:12 skrev Martin Joseph: >> >>> >>> On Jan 16, 2009, at 8:09 AM, Ivan C Myrvold wrote: >>> >>>> I haven't tried using launchctl for FreeSWITCH. But when I saw your >>>> post, I tried it out. I have no problem getting it to work: >>>> >>>> I make a file "org.freeswitch.freeswitch.plist" and save it to ~/ >>>> Library/LaunchAgents with the following content: >>>> >>>> >>>> >>> "> >>>> >>>> >>>> KeepAlive >>>> >>>> Label >>>> org.freeswitch.freeswitch >>>> Program >>>> /usr/local/freeswitch/bin/freeswitch >>>> RunAtLoad >>>> >>>> ServiceIPC >>>> >>>> >>>> >>>> >>>> Then in Terminal.app, I do a "launchctl load ~/Library/ >>>> LaunchAgents/ >>>> org.freeswitch.freeswitch.plist" >>>> >>>> If you do the same command, but unload instead of load, it should >>>> stop >>>> freeswitch. >>>> >>>> Does this work for you? >>> >>> Huh, >>> >>> I have been trying to do something similar, but putting the file >>> in / >>> System/Library/LaunchDaemons. >>> >>> Strangely, I tried your file right now (in LaunchDaemons) and >>> freeswitch is started, but it doesn't respond to my devices trying >>> to >>> register? >>> >>> Weird. >>> >>> If I kill it and start it manually, it immediately responds and my >>> devices register. >>> >>> Ideas? >>> Thanks, >>> >>> Marty >>>> >>>> >>>> Ivan >>>> >>>> Den 15. jan.. 2009 kl. 22:54 skrev Martin Joseph: >>>> >>>>> Hello again FreeSwitchers, >>>>> >>>>> I have built the 1.02 on 10.4.11(OSX) and had no problems with >>>>> that. >>>>> >>>>> I have never been able to build from the SVN, but that is another >>>>> story. >>>>> >>>>> Now that I have migrated to 1.02 I was wondering if I can get some >>>>> help on a long standing issue I have with starting FS at boot. >>>>> >>>>> I am hoping to use Launchd which is the standard on OSX 10.4 and I >>>>> attempted to cobble together a script, but haven't had great >>>>> results. >>>>> >>>>> I did search for wiki entries on this, but haven't found any help >>>>> with >>>>> it. >>>>> >>>>> Ideas? >>>>> Thanks, >>>>> Marty >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From cyberalby at gmail.com Sat Jan 17 08:02:24 2009 From: cyberalby at gmail.com (Alberto Ceccarelli) Date: Sat, 17 Jan 2009 17:02:24 +0100 Subject: [Freeswitch-users] Dialing Out Problem via Gateway Message-ID: <94c121cc0901170802i18c883cdv104660fc8f132315@mail.gmail.com> Hi, I've the same problem. I've create the log with only the problem (the log is in attach). One gateway (messagenet.it) and two internal account (1002 and 1008). My dialplan: My Sip Profile: My freeswith is runnig on Windows Server 2003 with 2 NIC (1 interna 192.168.0.15 and 1 external 10.0.3.6 (212.001.001.001)) For privacy I've change my IP and my phone number. -- Alby -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090117/1e623650/attachment-0001.html -------------- next part -------------- 2009-01-17 16:09:35 [INFO] switch_core_sqldb.c:487 switch_core_sqldb_start() Opening DB 2009-01-17 16:09:35 [DEBUG] switch_scheduler.c:214 switch_scheduler_add_task() Added task 1 heartbeat (core) to run at 1232204975 2009-01-17 16:09:35 [NOTICE] switch_scheduler.c:166 switch_scheduler_task_thread() Starting task thread 2009-01-17 16:09:35 [CONSOLE] switch_core.c:1270 switch_core_init_and_modload() Bringing up environment. 2009-01-17 16:09:35 [CONSOLE] switch_core.c:1271 switch_core_init_and_modload() Loading Modules. 2009-01-17 16:09:35 [INFO] switch_time.c:656 switch_load_timezones() Timezone loaded 530 definitions 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [CORE_SOFTTIMER_MODULE] 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:221 switch_loadable_module_process() Adding Timer 'soft' 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [CORE_PCM_MODULE] 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PROXY-VID' (PROXY VIDEO PASS-THROUGH) 90000hz 0ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PROXY' (PROXY PASS-THROUGH) 8000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 11025hz 40ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 22050hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 8ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 6ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 4ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 2ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 8ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 6ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 4ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 2ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 8ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 6ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 4ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 2ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 10ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 8ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 6ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 4ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 2ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 10ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 30ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 40ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 50ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 60ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 10ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 30ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW 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switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 50ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 60ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 70ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 80ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 90ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 100ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 110ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 120ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 10ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 30ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 40ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 50ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 60ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 70ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 80ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 90ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 100ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 110ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 120ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 10ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 30ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 40ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 50ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 60ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 70ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 80ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 90ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 100ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 110ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 120ms 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_console] 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'console' 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_logfile] 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_enum] 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:203 switch_loadable_module_process() Adding Dialplan 'enum' 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'enum' 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'enum' 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'enum_auto' 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_cdr_csv] 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_event_socket] 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'socket' 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'event_sink' 2009-01-17 16:09:35 [NOTICE] sofia_reg.c:1872 sofia_reg_add_gateway() Added gateway 'messagenet.it' to profile 'external' 2009-01-17 16:09:35 [NOTICE] sofia.c:2234 config_sofia() Adding Alias [outbound] for profile [external] 2009-01-17 16:09:35 [NOTICE] sofia.c:2234 config_sofia() Adding Alias [nat] for profile [external] 2009-01-17 16:09:35 [NOTICE] sofia.c:2244 config_sofia() Started Profile external [sofia_reg_external] 2009-01-17 16:09:35 [NOTICE] sofia.c:1181 parse_domain_tag() Adding Alias [192.168.0.74] for profile [internal] 2009-01-17 16:09:35 [NOTICE] sofia.c:2234 config_sofia() Adding Alias [default] for profile [internal] 2009-01-17 16:09:35 [NOTICE] sofia.c:2244 config_sofia() Started Profile internal [sofia_reg_internal] su_socket_port_init(00DE3B58, 02491E00) called su_pthread_port_init(00DE3B58, 02491E00) called su_socket_port_init(0137A468, 02491E00) called su_pthread_port_init(0137A468, 02491E00) called soa_create("default", 0137A8D8, 00DD7F90) called soa_set_params(static::01ADEDC8, ...) called soa_set_params(static::01ADEDC8, ...) called nta_agent_create: initialized hash tables nta_agent_create: initialized transports nta_agent_create: initialized random identifiers nta_agent_create: initialized timer su_socket_port_init(01ADEFF8, 02491E00) called su_pthread_port_init(01ADEFF8, 02491E00) called nta_agent_create: initialized resolver tport_create(): 00DEC198 nta: master transport created tport(00DEC198) to */10.0.3.6:5080/sip tport(00DEC198): calling tport_listen for udp tport(00DEC198): new primary tport 00DD8D20 su_socket_port_init(01AE9DD8, 02491E00) called su_pthread_port_init(01AE9DD8, 02491E00) called soa_create("default", 01AE28C0, 01AE5580) called soa_set_params(static::01AE57C0, ...) called soa_set_params(static::01AE57C0, ...) called nta_agent_create: initialized hash tables nta_agent_create: initialized transports nta_agent_create: initialized random identifiers nta_agent_create: initialized timer nta_agent_create: initialized resolver tport_create(): 01AE9368 nta: master transport created tport(01AE9368) to */192.168.0.74:5060/sip tport(01AE9368): calling tport_listen for udp tport(01AE9368): new primary tport 01AE9650 tport(00DD8D20): listening at udp/10.0.3.6:5080/sip tport(00DEC198): calling tport_listen for tcp tport(00DEC198): new primary tport 01AE1540 tport(01AE9650): listening at udp/192.168.0.74:5060/sip tport(01AE9368): calling tport_listen for tcp tport(01AE9368): new primary tport 01AE17F8 tport(01AE17F8): listening at tcp/192.168.0.74:5060/sip nta: bound to (192.168.0.74:5060;transport=*) nta: agent_init_via: SIP/2.0/udp 192.168.0.74 (sip) nta: agent_init_via: SIP/2.0/tcp 192.168.0.74 (sip) nta: Via fields initialized nta: Contact header created nua_register: Adding contact URL '192.168.0.74' to list. tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01AE5FB8 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 01AE5FB8 (0 bytes) from udp/192.168.0.74:5060/sip next=00000000 tport(01AE1540): listening at tcp/10.0.3.6:5080/sip nta: bound to (212.001.001.001:5080;transport=*;maddr=10.0.3.6) nta: agent_init_via: SIP/2.0/udp 212.001.001.001:5080 (sip) nta: agent_init_via: SIP/2.0/tcp 212.001.001.001:5080 (sip) nta: Via fields initialized nta: Contact header created nua_register: Adding contact URL '212.001.001.001' to list. nua(00000000): sent signal r_set_params nua(00000000): recv signal r_set_params soa_set_params(static::01AE57C0, ...) called nua(00000000): event r_set_params 200 OK tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01AF0170 from (udp/10.0.3.6:5080) has 4 bytes, veclen = 1 tport(00DD8D20): bad msg 01AF0170 (0 bytes) from udp/10.0.3.6:5080/sip next=00000000 nua(00000000): sent signal r_set_params nua(00000000): recv signal r_set_params soa_set_params(static::01ADEDC8, ...) called nua(00000000): event r_set_params 200 OK 2009-01-17 16:09:36 [NOTICE] sofia_reg.c:265 sofia_reg_check_gateway() Registering messagenet.it nua(01AE32E0): sent signal r_register nua(01AE32E0): recv signal r_register soa_clone(static::01ADEDC8, 0137A8D8, 01AE32E0) called soa_set_params(static::01AEC380, ...) called soa_set_params(static::01AEC380, ...) called nta_leg_tcreate(01AED9C0) nua(01AE32E0): adding register usage nta: selecting scheme sip nta: for "sip.messagenet.it" query "_sip._udp.sip.messagenet.it" SRV nta: for "sip.messagenet.it" query "sip.messagenet.it" A nta: sip.messagenet.it. IN A 212.97.59.76 tport_tsend(00DD8D20) tpn = udp/212.97.59.76:5060 tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name udp/212.97.59.76:5060 tport_vsend(00DD8D20): 639 bytes of 639 to udp/212.97.59.76:5060 tport_vsend returned 639 send 639 bytes to udp/[212.97.59.76]:5060 at 15:09:36.803764: ------------------------------------------------------------------------ REGISTER sip:sip.messagenet.it;transport=udp SIP/2.0 Via: SIP/2.0/UDP 212.001.001.001:5080;rport;branch=z9hG4bKmD9FZQBy2FKXp Max-Forwards: 70 From: ;tag=mypD78F0vQSjc To: Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984152 REGISTER Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ nta: sent REGISTER (109984152) to udp/212.97.59.76:5060 tport_pend(00DD8D20): pending 01AF0170 for udp/10.0.3.6:5080 (already 0) nta: timer set to 32000 ms nta: timer shortened to 500 ms tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01B5D978 from (udp/10.0.3.6:5080) has 1278 bytes, veclen = 1 recv 564 bytes from udp/[212.97.59.76]:5060 at 15:09:36.803764: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.001.001.001:5080;rport=5080;branch=z9hG4bKmD9FZQBy2FKXp From: ;tag=mypD78F0vQSjc To: Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984152 REGISTER Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5060 "Noisy feedback tells: pid=15517 req_src_ip=212.001.001.001 req_src_port=5080 in_uri=sip:sip.messagenet.it;transport=udp out_uri=sip:sip.messagenet.it;transport=udp via_cnt==1" ------------------------------------------------------------------------ tport(00DD8D20): msg 01B5D978 (564 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received 100 Trying for REGISTER (109984152) nta: 100 Trying is going to a transaction nta_outgoing: RTT is 0 ms tport(00DD8D20): 01AF0170 by 01AED108 with 01B5D978 (preliminary) tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01B5D978 from (udp/10.0.3.6:5080) has 714 bytes, veclen = 1 recv 714 bytes from udp/[212.97.59.76]:5060 at 15:09:36.803764: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 212.001.001.001:5080;rport=5080;branch=z9hG4bKmD9FZQBy2FKXp From: ;tag=mypD78F0vQSjc To: ;tag=797b035a1d6b0049ed33903c5ef0eddf.062a Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984152 REGISTER WWW-Authenticate: Digest realm="sip.messagenet.it", nonce="4971f5dfd4d7b34dd71873b0bc184126b45c1700" Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5060 "Noisy feedback tells: pid=15517 req_src_ip=212.001.001.001 req_src_port=5080 in_uri=sip:sip.messagenet.it;transport=udp out_uri=sip:sip.messagenet.it;transport=udp via_cnt==1" ------------------------------------------------------------------------ tport(00DD8D20): msg 01B5D978 (714 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received 401 Unauthorized for REGISTER (109984152) nta: 401 Unauthorized is going to a transaction tport(00DD8D20): 01AF0170 by 01AED108 with 01B5D978 auth_digest_challenge_get(): got 2 nua(01AE32E0): event r_register 401 Unauthorized nua(01AE32E0): recv signal r_authenticate auth_digest_a1() has A1 = MD5(5343504:sip.messagenet.it:zanebap) = 7d86e12ae1b9db5b55b0653441352198 A2 = MD5(REGISTER:sip:sip.messagenet.it;transport=udp) auth_response: 937ac9aceb3126881d30de0ff6c9bcea = MD5(7d86e12ae1b9db5b55b0653441352198:4971f5dfd4d7b34dd71873b0bc184126b45c1700:c7eb2509efe71a9f29aa95f4d4019686) (qop=NONE) nta: selecting scheme sip nta: for "sip.messagenet.it" query "_sip._udp.sip.messagenet.it" SRV (cached) nta: for "sip.messagenet.it" query "sip.messagenet.it" A (cached) nta: sip.messagenet.it. IN A 212.97.59.76 tport_tsend(00DD8D20) tpn = udp/212.97.59.76:5060 tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name udp/212.97.59.76:5060 tport_vsend(00DD8D20): 861 bytes of 861 to udp/212.97.59.76:5060 tport_vsend returned 861 send 861 bytes to udp/[212.97.59.76]:5060 at 15:09:36.803764: ------------------------------------------------------------------------ REGISTER sip:sip.messagenet.it;transport=udp SIP/2.0 nua(01AE32E0): sent signal r_authenticate Via: SIP/2.0/UDP 212.001.001.001:5080;rport;branch=z9hG4bKNp280jv1Zr9Fj Max-Forwards: 70 From: ;tag=mypD78F0vQSjc To: Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984153 REGISTER Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Authorization: Digest username="5343504", realm="sip.messagenet.it", nonce="4971f5dfd4d7b34dd71873b0bc184126b45c1700", algorithm=MD5, uri="sip:sip.messagenet.it;transport=udp", response="937ac9aceb3126881d30de0ff6c9bcea" Content-Length: 0 ------------------------------------------------------------------------ nta: sent REGISTER (109984153) to udp/212.97.59.76:5060 tport_pend(00DD8D20): pending 01B58858 for udp/10.0.3.6:5080 (already 0) tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01B59988 from (udp/10.0.3.6:5080) has 564 bytes, veclen = 1 recv 564 bytes from udp/[212.97.59.76]:5060 at 15:09:36.819385: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.001.001.001:5080;rport=5080;branch=z9hG4bKNp280jv1Zr9Fj From: ;tag=mypD78F0vQSjc To: Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984153 REGISTER Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5060 "Noisy feedback tells: pid=15519 req_src_ip=212.001.001.001 req_src_port=5080 in_uri=sip:sip.messagenet.it;transport=udp out_uri=sip:sip.messagenet.it;transport=udp via_cnt==1" ------------------------------------------------------------------------ tport(00DD8D20): msg 01B59988 (564 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received 100 Trying for REGISTER (109984153) nta: 100 Trying is going to a transaction nta_outgoing: RTT is 15.621 ms tport(00DD8D20): 01B58858 by 01B5C388 with 01B59988 (preliminary) tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01B59988 from (udp/10.0.3.6:5080) has 715 bytes, veclen = 1 recv 715 bytes from udp/[212.97.59.76]:5060 at 15:09:36.819385: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 212.001.001.001:5080;rport=5080;branch=z9hG4bKNp280jv1Zr9Fj From: ;tag=mypD78F0vQSjc To: ;tag=797b035a1d6b0049ed33903c5ef0eddf.bdba Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984153 REGISTER Date: Sat, 17 Jan 2009 15:09:39 GMT Contact: ;q=0.5;expires=3600 Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5060 "Noisy feedback tells: pid=15519 req_src_ip=212.001.001.001 req_src_port=5080 in_uri=sip:sip.messagenet.it;transport=udp out_uri=sip:sip.messagenet.it;transport=udp via_cnt==1" ------------------------------------------------------------------------ tport(00DD8D20): msg 01B59988 (715 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received 200 OK for REGISTER (109984153) nta: 200 OK is going to a transaction tport(00DD8D20): 01B58858 by 01B5C388 with 01B59988 nua(): refresh register after 2568 seconds (in [900..2700]) nua(): refresh register after 2568 seconds nua(01AE32E0): event r_register 200 OK 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_sofia] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'sofia' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sofia' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sofia_contact' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'sip' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:394 switch_loadable_module_process() Adding Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1] 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_loopback] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'loopback' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_commands] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'group_call' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'in_group' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_flush_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'md5' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'hupall' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'strftime_tz' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'originate' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'tone_detect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_kill' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_park' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'reloadacl' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'reloadxml' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'unload' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'reload' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'load' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_transfer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'pause' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'break' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'show' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'complete' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'alias' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'status' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_session_heartbeat' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_bridge' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_setvar' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_setvar_multi' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_getvar' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_dump' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'global_setvar' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'global_getvar' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_displace' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_record' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_broadcast' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_hold' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_display' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_media' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'fsctl' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'help' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'version' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sched_hangup' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sched_broadcast' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sched_transfer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'create_uuid' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sched_api' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'unsched_api' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'bgapi' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sched_del' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'xml_wrap' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'is_lan_addr' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'cond' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'regex' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'acl' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_chat' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_deflect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'find_user_xml' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'user_exists' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'xml_locate' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'user_data' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'url_encode' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'url_decode' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'module_exists' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'domain_exists' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_send_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'eval' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'system' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'time_test' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_conference] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'conference' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'conference_set_auto_outcall' 2009-01-17 16:09:37 [CONSOLE] sofia_presence.c:621 sofia_presence_event_thread_run() Event Thread Started 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'conference' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'conf' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_dptools] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'error' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'group' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'user' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:203 switch_loadable_module_process() Adding Dialplan 'inline' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'privacy' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'flush_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'hold' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'unhold' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'transfer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'check_acl' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'verbose_events' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'sleep' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'delay_echo' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'strftime' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'phrase' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'eval' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'pre_answer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'answer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'hangup' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'set_name' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'presence' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'log' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'info' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'event' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'export' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'set' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'set_global' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'set_profile_var' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'unset' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'ring_ready' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'remove_bugs' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'break' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'detect_speech' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'ivr' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'redirect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'send_display' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'respond' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'deflect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'queue_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'send_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'sched_hangup' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'sched_broadcast' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'sched_transfer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'execute_extension' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'mkdir' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'soft_hold' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'bind_meta_app' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'unbind_meta_app' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'intercept' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'eavesdrop' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'three_way' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'set_user' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'stop_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'start_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'stop_dtmf_generate' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'start_dtmf_generate' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'stop_tone_detect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'fax_detect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'tone_detect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'echo' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'park' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'park_state' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'gentones' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'playback' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'att_xfer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'read' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'stop_record_session' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'record_session' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'record' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'stop_displace_session' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'displace_session' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'speak' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'clear_speech_cache' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'bridge' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'system' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'say' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'wait_for_silence' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'strepoch' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'chat' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'strftime' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'presence' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'event' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'api' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_expr] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'expr' 2009-01-17 16:09:37 [INFO] mod_fifo.c:1811 load_config() cool_fifo at 192.168.0.74 configured 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_fifo] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'fifo' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'fifo' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'fifo_member' 2009-01-17 16:09:37 [INFO] mod_voicemail.c:765 load_config() Added Profile default 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_voicemail] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'voicemail' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'voicemail' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'voicemail_inject' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'vm_boxcount' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_limit] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'limit' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'limit_hash' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'db' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'hash' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'group' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'limit_hash_usage' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'limit_usage' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'db' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'hash' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'group' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_esf] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'esf_page_group' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_fsv] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'play_fsv' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'record_fsv' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_dialplan_xml] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:203 switch_loadable_module_process() Adding Dialplan 'XML' nta: timer set next to 4485 ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_dialplan_asterisk] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'SIP' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'IAX2' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:203 switch_loadable_module_process() Adding Dialplan 'asterisk' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'Dial' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'Goto' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'AvoidingDeadlock' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_voipcodecs] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'LPC' (LPC-10) 8000hz 90ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_g723_1] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G723' (G.723.1 6.3k) 8000hz 30ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_g729] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 120ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_amr] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AMR' (AMR) 8000hz 20ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_ilbc] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC' (iLBC) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC' (iLBC) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC20ms' (iLBC) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC102' (iLBC) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC' (iLBC) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC' (iLBC) 8000hz 20ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_speex] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'SPEEX' (Speex) 32000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'SPEEX' (Speex) 16000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'SPEEX' (Speex) 8000hz 20ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_h26x] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'H264' (H.264 Video (passthru)) 90000hz 0ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'H263' (H.263 Video (passthru)) 90000hz 0ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'H263-1998' (H.263+ Video (passthru)) 90000hz 0ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'H263-2000' (H.263++ Video (passthru)) 90000hz 0ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'H261' (H.261 Video (passthru)) 90000hz 0ms 2009-01-17 16:09:37 [INFO] mod_sndfile.c:308 setup_formats() LibSndFile Version : libsndfile-1.0.12 Supported Formats ================================================================================ AIFF (Apple/SGI) (extension "aiff") AU (Sun/NeXT) (extension "au") AVR (Audio Visual Research) (extension "avr") CAF (Apple Core Audio File) (extension "caf") HTK (HMM Tool Kit) (extension "htk") IFF (Amiga IFF/SVX8/SV16) (extension "iff") MAT4 (GNU Octave 2.0 / Matlab 4.2) (extension "mat") MAT5 (GNU Octave 2.1 / Matlab 5.0) (extension "mat") PAF (Ensoniq PARIS) (extension "paf") PVF (Portable Voice Format) (extension "pvf") RAW (header-less) (extension "raw") SD2 (Sound Designer II) (extension "sd2") SDS (Midi Sample Dump Standard) (extension "sds") SF (Berkeley/IRCAM/CARL) (extension "sf") VOC (Creative Labs) (extension "voc") W64 (SoundFoundry WAVE 64) (extension "w64") WAV (Microsoft) (extension "wav") WAV (NIST Sphere) (extension "wav") WAVEX (Microsoft) (extension "wav") XI (FastTracker 2) (extension "xi") ================================================================================ 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_sndfile] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'aiff' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'au' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'avr' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'caf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'htk' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'iff' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'mat' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'mat' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'paf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'pvf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'raw' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'sd2' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'sds' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'sf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'voc' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'w64' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'wav' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'wav' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'wav' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'xi' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'r8' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'r16' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'r24' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'r32' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'gsm' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'ul' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'al' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_native_file] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'H263' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'iLBC20ms' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'AMR' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'H263-1998' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'SPEEX' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G729' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G723' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G726-16' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'H261' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'AAL2-G726-16' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'DVI4' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'PCMA' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'PCMU' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'L16' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'iLBC20ms' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'PROXY' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'PROXY-VID' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'AAL2-G726-24' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'AAL2-G726-32' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'H263-2000' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'H264' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G726-32' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'iLBC20ms' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G722' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'AAL2-G726-40' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G726-40' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'GSM' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'LPC' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_local_stream] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'stop_local_stream' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'start_local_stream' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'local_stream' 2009-01-17 16:09:37 [CONSOLE] mod_local_stream.c:142 read_stream_thread() Can't open directory: C:\Program Files\FreeSwitch/sounds/music/16000 2009-01-17 16:09:37 [CONSOLE] mod_local_stream.c:142 read_stream_thread() Can't open directory: C:\Program Files\FreeSwitch/sounds/music/32000 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_tone_stream] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'tone_stream' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'silence_stream' 2009-01-17 16:09:37 [CONSOLE] mod_spidermonkey.c:948 sm_load_file() Successfully Loaded [C:\Program Files\FreeSwitch\mod\mod_spidermonkey_teletone.dll] 2009-01-17 16:09:37 [CONSOLE] mod_spidermonkey.c:948 sm_load_file() Successfully Loaded [C:\Program Files\FreeSwitch\mod\mod_spidermonkey_core_db.dll] 2009-01-17 16:09:37 [CONSOLE] mod_spidermonkey.c:948 sm_load_file() Successfully Loaded [C:\Program Files\FreeSwitch\mod\mod_spidermonkey_socket.dll] 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_spidermonkey] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'javascript' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'jsrun' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'jsapi' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_lua] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'lua' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'luarun' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'lua' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_say_en] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:372 switch_loadable_module_process() Adding Say interface 'en' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:119 switch_loadable_module_runtime() Starting runtime thread for CORE_SOFTTIMER_MODULE 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:119 switch_loadable_module_runtime() Starting runtime thread for mod_event_socket 2009-01-17 16:09:37 [CONSOLE] switch_core.c:865 switch_load_network_lists() Created ip list dl-candidates default (allow) 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 10.0.0.0/8 (deny) to list dl-candidates 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 172.16.0.0/12 (deny) to list dl-candidates 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 192.168.0.0/16 (deny) to list dl-candidates 2009-01-17 16:09:37 [CONSOLE] switch_core.c:865 switch_load_network_lists() Created ip list rfc1918 default (deny) 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 10.0.0.0/8 (allow) to list rfc1918 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 172.16.0.0/12 (allow) to list rfc1918 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 192.168.0.0/16 (allow) to list rfc1918 2009-01-17 16:09:37 [CONSOLE] switch_core.c:865 switch_load_network_lists() Created ip list lan default (allow) 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 192.168.42.0/24 (deny) to list lan 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 192.168.42.42/32 (allow) to list lan 2009-01-17 16:09:37 [CONSOLE] switch_core.c:865 switch_load_network_lists() Created ip list strict default (deny) 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 208.102.123.124/32 (allow) to list strict 2009-01-17 16:09:37 [CONSOLE] switch_core.c:865 switch_load_network_lists() Created ip list domains default (deny) 2009-01-17 16:09:37 [CONSOLE] switch_core.c:1287 switch_core_init_and_modload() FreeSWITCH Version 1.0.trunk (UNKNOWN) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] freeswitch at inetmail2> nta: timer K fired, terminate REGISTER (109984152) outgoing_reclaim_all(00000000, 00000000, 027FFEC0) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 1 ms nta: timer K fired, terminate REGISTER (109984153) outgoing_reclaim_all(00000000, 00000000, 027FFEC0) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01B59988 from (udp/192.168.0.74:5060) has 543 bytes, veclen = 1 recv 543 bytes from udp/[192.168.0.52]:63214 at 15:09:57.844578: ------------------------------------------------------------------------ REGISTER sip:192.168.0.74 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-10626c213e281d27-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Alberto" From: "Alberto";tag=544adf7f Call-ID: ZjNlNGY5ZGE5OGI5MjFkNTc3Y2RjMzhhMDc4MmM1NDc. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 01B59988 (543 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received REGISTER sip:192.168.0.74 SIP/2.0 (CSeq 1) nta: canonizing sip:192.168.0.74 with contact nta: REGISTER (1) going to a default leg soa_clone(static::01AE57C0, 01AE28C0, 01B6FBD8) called soa_set_params(static::01B5EF70, ...) called nua(01B6FBD8): event i_register 100 Trying nua(01B6FBD8): sent signal r_respond nua(01B6FBD8): recv signal r_respond 401 Unauthorized soa_set_params(static::01B5EF70, ...) called nua(01B6FBD8): sent signal r_destroy tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 667 bytes of 667 to udp/192.168.0.52:63214 tport_vsend returned 667 send 667 bytes to udp/[192.168.0.52]:63214 at 15:09:57.844578: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-10626c213e281d27-1---d8754z-;rport=63214 From: "Alberto";tag=544adf7f To: "Alberto" ;tag=mypD78F0vQSjc Call-ID: ZjNlNGY5ZGE5OGI5MjFkNTc3Y2RjMzhhMDc4MmM1NDc. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="192.168.0.74", nonce="8d55768a-5992-1c4d-912e-92961f1f4e5e", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ nta: sent 401 Unauthorized for REGISTER (1) nta: timer set to 32000 ms nua(01B6FBD8): recv signal r_destroy nta_leg_destroy(00000000) soa_destroy(static::01B5EF70) called tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01B5D978 from (udp/192.168.0.74:5060) has 792 bytes, veclen = 1 recv 792 bytes from udp/[192.168.0.52]:63214 at 15:09:58.047644: ------------------------------------------------------------------------ REGISTER sip:192.168.0.74 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-53075b0a800fd422-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Alberto" From: "Alberto";tag=544adf7f Call-ID: ZjNlNGY5ZGE5OGI5MjFkNTc3Y2RjMzhhMDc4MmM1NDc. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="1008",realm="192.168.0.74",nonce="8d55768a-5992-1c4d-912e-92961f1f4e5e",uri="sip:192.168.0.74",response="55d0982a97d14c1b35e41e9617710f4f",cnonce="d4204f159a249cc80d1ea728bac88ad2",nc=00000001,qop=auth,algorithm=MD5 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 01B5D978 (792 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received REGISTER sip:192.168.0.74 SIP/2.0 (CSeq 2) nta: canonizing sip:192.168.0.74 with contact nta: REGISTER (2) going to a default leg soa_clone(static::01AE57C0, 01AE28C0, 01B6FBD8) called soa_set_params(static::01B75A20, ...) called nua(01B6FBD8): event i_register 100 Trying nua(01B6FBD8): sent signal r_respond nua(01B6FBD8): recv signal r_respond 200 OK soa_set_params(static::01B75A20, ...) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 654 bytes of 654 to udp/192.168.0.52:63214 tport_vsend returned 654 nua(01B6FBD8): sent signal r_destroy send 654 bytes to udp/[192.168.0.52]:63214 at 15:09:58.047644: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-53075b0a800fd422-1---d8754z-;rport=63214 From: "Alberto";tag=544adf7f To: "Alberto" ;tag=N7F68303S0F5Q Call-ID: ZjNlNGY5ZGE5OGI5MjFkNTc3Y2RjMzhhMDc4MmM1NDc. CSeq: 2 REGISTER Contact: ;expires=3600 Date: Sat, 17 Jan 2009 15:09:58 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ nta: sent 200 OK for REGISTER (2) nua(01B6FBD8): recv signal r_destroy nta_leg_destroy(00000000) soa_destroy(static::01B75A20) called nua(01B6FBD8): sent signal r_notify nua(01B6FBD8): recv signal r_notify soa_clone(static::01AE57C0, 01AE28C0, 01B6FBD8) called soa_set_params(static::01B75A20, ...) called soa_set_params(static::01B75A20, ...) called nta_leg_tcreate(00DD7010) nua(01B6FBD8): adding notify usage with event message-summary nta: selecting scheme sip tport_tsend(01AE9650) tpn = */192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name */192.168.0.52:63214 tport_vsend(01AE9650): 887 bytes of 887 to udp/192.168.0.52:63214 tport_vsend returned 887 send 887 bytes to udp/[192.168.0.52]:63214 at 15:09:58.141367: ------------------------------------------------------------------------ NOTIFY sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.74;rport;branch=z9hG4bKmD9FZQBy2FKXp Max-Forwards: 70 From: ;tag=pg9yaZH7p95QK To: Call-ID: beeb792e-5f4b-122c-2780-39a48cb53b8d CSeq: 109984163 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;timeout Content-Type: application/simple-message-summary Content-Length: 64 Messages-Waiting: no Message-Account: sip:1008 at 192.168.0.74 ------------------------------------------------------------------------ nta: sent NOTIFY (109984163) to */192.168.0.52:63214 tport_pend(01AE9650): pending 01B72C38 for udp/192.168.0.74:5060 (already 0) nta: timer shortened to 500 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F2FC50 from (udp/192.168.0.74:5060) has 546 bytes, veclen = 1 recv 546 bytes from udp/[192.168.0.52]:63214 at 15:09:58.141367: ------------------------------------------------------------------------ SUBSCRIBE sip:1008 at 192.168.0.74 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-f3490746570af404-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Alberto" From: "Alberto";tag=d11f6f44 Call-ID: ZDY2ZWMwN2M0MTkwZDA1NDJkOGNhZmNmYTg5NDgyNGY. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Event: message-summary Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F2FC50 (546 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received SUBSCRIBE sip:1008 at 192.168.0.74 SIP/2.0 (CSeq 1) nta: canonizing sip:1008 at 192.168.0.74 with contact nta: SUBSCRIBE (1) going to a default leg soa_clone(static::01AE57C0, 01AE28C0, 01B72720) called soa_set_params(static::01B5EF70, ...) called nta_leg_tcreate(01B6F530) nua(01B72720): adding notify usage with event message-summary nua(01B72720): event i_subscribe 100 Trying nua(): refresh notify after 600 seconds (in [600..600]) nua(): refresh notify after 600 seconds nua(01B72720): recv signal r_respond 202 Accepted soa_set_params(static::01B5EF70, ...) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 771 bytes of 771 to udp/192.168.0.52:63214 tport_vsend returned 771 send 771 bytes to udp/[192.168.0.52]:63214 at 15:09:58.141367: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-f3490746570af404-1---d8754z-;rport=63214 From: "Alberto";tag=d11f6f44 To: "Alberto" ;tag=QS2Qct2amjvaF Call-ID: ZDY2ZWMwN2M0MTkwZDA1NDJkOGNhZmNmYTg5NDgyNGY.nua(01B72720): sent signal r_respond CSeq: 1 SUBSCRIBE Contact: "Alberto" Expires: 300 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=300 Content-Length: 0 ------------------------------------------------------------------------ nta: sent 202 Accepted for SUBSCRIBE (1) tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01D71768 from (udp/192.168.0.74:5060) has 350 bytes, veclen = 1 recv 350 bytes from udp/[192.168.0.52]:63214 at 15:09:58.141367: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.74;rport=5060;branch=z9hG4bKmD9FZQBy2FKXp Contact: To: ;tag=321bea76 From: ;tag=pg9yaZH7p95QK Call-ID: beeb792e-5f4b-122c-2780-39a48cb53b8d CSeq: 109984163 NOTIFY User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 01D71768 (350 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received 200 OK for NOTIFY (109984163) nta: 200 OK is going to a transaction nta_outgoing: RTT is 15.621 ms tport(01AE9650): 01B72C38 by 01AED108 with 01D71768 nua(01B6FBD8): event r_notify 200 OK nua(01B6FBD8): removing notify usage with event message-summary nta_leg_destroy(00DD7010) nua(01B6FBD8): sent signal r_destroy nua(01B6FBD8): recv signal r_destroy nta_leg_destroy(00000000) soa_destroy(static::01B75A20) called nta: timer set next to 4484 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F4BC48 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 02F4BC48 (4 bytes) from udp/192.168.0.38:5060/sip next=00000000 nta: timer K fired, terminate NOTIFY (109984163) outgoing_reclaim_all(00000000, 00000000, 028FFEC0) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer set next to 26689 ms tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01D71768 from (udp/10.0.3.6:5080) has 1168 bytes, veclen = 1 recv 1168 bytes from udp/[212.97.59.76]:5060 at 15:10:08.076005: ------------------------------------------------------------------------ INVITE sip:5343504 at 212.001.001.001:5080;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKe61f.f220cd02.0 Via: SIP/2.0/UDP 212.97.59.87:5060;branch=z9hG4bK57267a99;rport=5060 From: "+39020000000" ;tag=as302aa103 To: Contact: Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 102 INVITE User-Agent: alpha Max-Forwards: 69 Date: Sat, 17 Jan 2009 15:10:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 395 v=0 o=root 3859 3859 IN IP4 212.97.59.87 s=session c=IN IP4 212.97.59.91 t=0 0 m=audio 38196 RTP/AVP 18 3 97 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes ------------------------------------------------------------------------ tport(00DD8D20): msg 01D71768 (1168 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received INVITE sip:5343504 at 212.001.001.001:5080;transport=udp SIP/2.0 (CSeq 102) nta: canonizing sip:5343504 at 212.001.001.001:5080 with contact nta: INVITE (102) going to a default leg nta: timer set to 200 ms soa_clone(static::01ADEDC8, 0137A8D8, 02EDF6E0) called soa_set_params(static::01B75A20, ...) called nta_leg_tcreate(02F004E0) soa_init_offer_answer(static::01B75A20) called soa_set_remote_sdp(static::01B75A20, 00000000, 02F47CAD, 395) called nua(02EDF6E0): adding session usage tport_tsend(00DD8D20) tpn = UDP/212.97.59.76:5060 tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name UDP/212.97.59.76:5060 tport_vsend(00DD8D20): 546 bytes of 546 to udp/212.97.59.76:5060 tport_vsend returned 546 send 546 bytes to udp/[212.97.59.76]:5060 at 15:10:08.076005: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKe61f.f220cd02.0 Via: SIP/2.0/UDP 212.97.59.87:5060;branch=z9hG4bK57267a99;rport=5060 Record-Route: Record-Route: From: "+39020000000" ;tag=as302aa103 To: Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (102) nua(02EDF6E0): event i_invite 100 Trying nua(02EDF6E0): call state changed: init -> received, received offer soa_get_remote_sdp(static::01B75A20, [027FFC14], [027FFC10], [00000000]) called nua(02EDF6E0): event i_state 100 Trying 2009-01-17 16:10:08 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/external/+39020000000 at sip.messagenet.it [711a2124-077d-f442-ae20-e63b2273f326] 2009-01-17 16:10:08 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing +39020000000->5343504 in context public 2009-01-17 16:10:08 [NOTICE] switch_ivr.c:1255 switch_ivr_session_transfer() Transfer sofia/external/+39020000000 at sip.messagenet.it to XML[1008 at default] 2009-01-17 16:10:08 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing +39020000000->1008 in context default 2009-01-17 16:10:08 [ERR] switch_regex.c:94 switch_regex_perform() COMPILE ERROR: 1 [nothing to repeat][^+39020000000$] 2009-01-17 16:10:08 [INFO] switch_ivr_async.c:1644 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-01-17 16:10:08 [INFO] switch_ivr_async.c:1644 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::C:Program FilesFreeSwitch/recordings/+39020000000.2009-01-17-16-10-08.wav 2009-01-17 16:10:08 [INFO] switch_ivr_async.c:1644 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2009-01-17 16:10:08 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 [698623ef-686c-9c4d-93c7-3cc6db803902] nua(02EDF468): sent signal r_invite nua(02EDF468): recv signal r_invite soa_clone(static::01AE57C0, 01AE28C0, 02EDF468) called soa_set_params(static::01D7FE78, ...) called soa_set_params(static::01D7FE78, ...) called soa_set_user_sdp(static::01D7FE78, 00000000, 012733C2, -1) called soa_set_capability_sdp(static::01D7FE78, 00000000, 012733C2, -1) called nta_leg_tcreate(02F47280) nua(02EDF468): adding session usage soa_init_offer_answer(static::01D7FE78) called soa_generate_offer(static::01D7FE78, 0) called soa_static_offer_answer_action(01D7FE78, soa_generate_offer): called soa_static(01D7FE78, soa_generate_offer): generating local description soa_static(01D7FE78, soa_generate_offer): upgrade with local description soa_sdp_mode_set(028FDCAC, 00000000, ""): called soa_static(01D7FE78, soa_generate_offer): storing local description soa_get_local_sdp(static::01D7FE78, [00000000], [028FFDC4], [028FFDC0]) called nta: selecting scheme sip tport_tsend(01AE9650) tpn = */192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name */192.168.0.52:63214 tport_vsend(01AE9650): 1261 bytes of 1261 to udp/192.168.0.52:63214 tport_vsend returned 1261 send 1261 bytes to udp/[192.168.0.52]:63214 at 15:10:08.216590: ------------------------------------------------------------------------ INVITE sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.74;rport;branch=z9hG4bKNp280jv1Zr9Fj Max-Forwards: 67 From: "+39020000000" ;tag=r2UgeNKeHUjXa To: Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 109984168 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 333 Remote-Party-ID: "+39020000000" ;screen=yes;privacy=off v=0 o=FreeSWITCH 6699757194367626025 7696645903169343292 IN IP4 192.168.0.74 s=FreeSWITCH c=IN IP4 192.168.0.74 t=0 0 m=audio 29688 RTP/AVP 3 9 0 8 101 13 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ nta: sent INVITE (109984168) to */192.168.0.52:63214 tport_pend(01AE9650): pending 02FC1940 for udp/192.168.0.74:5060 (already 0) nta: timer shortened to 500 ms nua(02EDF468): call state changed: init -> calling, sent offer soa_get_local_sdp(static::01D7FE78, [028FFDC4], [028FFDC0], [00000000]) called nua(02EDF468): event i_state INVITE sent nta: timer not set 2009-01-17 16:10:08 [INFO] mod_sofia.c:1294 sofia_receive_message() Asked to send early media by sofia/external/+39020000000 at sip.messagenet.it 2009-01-17 16:10:08 [INFO] mod_sofia.c:1335 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1232175320 1232175321 IN IP4 212.001.001.001 s=FreeSWITCH c=IN IP4 212.001.001.001 t=0 0 m=audio 29688 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-01-17 16:10:08 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Ring-Ready sofia/external/+39020000000 at sip.messagenet.it! 2009-01-17 16:10:08 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Pre-Answer sofia/external/+39020000000 at sip.messagenet.it! nua(02EDF6E0): sent signal r_respond nua(02EDF6E0): recv signal r_respond 183 Session Progress soa_set_params(static::01B75A20, ...) called soa_set_user_sdp(static::01B75A20, 00000000, 00DECA53, -1) called soa_set_capability_sdp(static::01B75A20, 00000000, 00DECA53, -1) called soa_generate_answer(static::01B75A20) called soa_static_offer_answer_action(01B75A20, soa_generate_answer): called soa_static(01B75A20, soa_generate_answer): generating local description soa_static(01B75A20, soa_generate_answer): upgrade with remote description soa_sdp_mode_set(027FDCF8, 01274870, ""): called soa_static(01B75A20, soa_generate_answer): storing local description soa_activate(static::01B75A20, (nil)) called soa_get_local_sdp(static::01B75A20, [00000000], [027FFE0C], [027FFE08]) called tport_tsend(00DD8D20) tpn = UDP/212.97.59.76:5060 tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name UDP/212.97.59.76:5060 tport_vsend(00DD8D20): 1170 bytes of 1170 to udp/212.97.59.76:5060 tport_vsend returned 1170 send 1170 bytes to udp/[212.97.59.76]:5060 at 15:10:08.310313: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKe61f.f220cd02.0 Via: SIP/2.0/UDP 212.97.59.87:5060;branch=z9hG4bK57267a99;rport=5060 Record-Route: Record-Route: From: "+39020000000" ;tag=as302aa103 To: ;tag=N7F68303S0F5Q Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 102 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 266 v=0 o=FreeSWITCH 1110445663877587962 6324161778166975444 IN IP4 212.001.001.001 s=FreeSWITCH c=IN IP4 212.001.001.001 t=0 0 m=audio 29688 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ nta: sent 183 Session Progress for INVITE (102) nta: timer set to 60000 ms nua(02EDF6E0): call state changed: received -> early, sent answer soa_get_local_sdp(static::01B75A20, [027FFE48], [027FFE44], [00000000]) called soa_get_params(static::01B75A20, ...) called nua(02EDF6E0): event i_state 183 Session Progress tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F4BC48 from (udp/192.168.0.74:5060) has 316 bytes, veclen = 1 recv 316 bytes from udp/[192.168.0.52]:63214 at 15:10:08.325933: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.74;rport=5060;branch=z9hG4bKNp280jv1Zr9Fj To: From: "+39020000000" ;tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 109984168 INVITE Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F4BC48 (316 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received 100 Trying for INVITE (109984168) nta: 100 Trying is going to a transaction nta_outgoing: RTT is 109.343 ms tport(01AE9650): 02FC1940 by 02F30CF8 with 02F4BC48 (preliminary) tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F4BC48 from (udp/192.168.0.74:5060) has 442 bytes, veclen = 1 recv 442 bytes from udp/[192.168.0.52]:63214 at 15:10:08.638343: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.74;rport=5060;branch=z9hG4bKNp280jv1Zr9Fj Contact: To: ;tag=8b08c857 From: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 109984168 INVITE User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F4BC48 (442 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received 180 Ringing for INVITE (109984168) nta: 180 Ringing is going to a transaction tport(01AE9650): 02FC1940 by 02F30CF8 with 02F4BC48 (preliminary) nua(02EDF468): event r_invite 180 Ringing nua(02EDF468): call state changed: calling -> proceeding nua(02EDF468): event i_state 180 Ringing 2009-01-17 16:10:08 [NOTICE] sofia.c:2627 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5! nta: timer set next to 21113 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F51A68 from (udp/192.168.0.74:5060) has 740 bytes, veclen = 1 recv 740 bytes from udp/[192.168.0.52]:63214 at 15:10:12.059233: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.74;rport=5060;branch=z9hG4bKNp280jv1Zr9Fj Contact: To: ;tag=8b08c857 From: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 109984168 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 187 v=0 o=- 6 2 IN IP4 192.168.0.52 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.52 t=0 0 m=audio 28768 RTP/AVP 3 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ tport(01AE9650): msg 02F51A68 (740 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received 200 OK for INVITE (109984168) nta: 200 OK is going to a transaction tport(01AE9650): 02FC1940 by 02F30CF8 with 02F51A68 soa_set_remote_sdp(static::01D7FE78, 00000000, 02FD0601, 187) called soa_process_answer(static::01D7FE78) called soa_static_offer_answer_action(01D7FE78, soa_process_answer): called soa_sdp_mode_set(02FC27A0, 03007728, ""): called soa_static(01D7FE78, soa_process_answer): upgrade codecs with remote description soa_static(01D7FE78, soa_process_answer): storing local description soa_activate(static::01D7FE78, (nil)) called nua(02EDF468): INVITE: processed SDP answer in 200 OK nua(02EDF468): event r_invite 200 OK soa_activate(static::01D7FE78, (nil)) called nta: selecting scheme sip tport_tsend(01AE9650) tpn = */192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name */192.168.0.52:63214 tport_vsend(01AE9650): 431 bytes of 431 to udp/192.168.0.52:63214 tport_vsend returned 431 send 431 bytes to udp/[192.168.0.52]:63214 at 15:10:12.059233: ------------------------------------------------------------------------ ACK sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.74;rport;branch=z9hG4bKpZU12DD5v1Z2D Max-Forwards: 70 From: "+39020000000" ;tag=r2UgeNKeHUjXa To: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 109984168 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ nta: sent ACK (109984168) to */192.168.0.52:63214 nua(02EDF468): call state changed: proceeding -> ready, received answer soa_get_remote_sdp(static::01D7FE78, [028FFC2C], [028FFC28], [00000000]) called soa_get_params(static::01D7FE78, ...) called nua(02EDF468): event i_state 200 OK nua(02EDF468): event i_active 200 Call active nua(02EDF6E0): recv signal r_respond 200 OK soa_set_params(static::01B75A20, ...) called soa_set_user_sdp(static::01B75A20, 00000000, 0127FA5D, -1) called soa_get_local_sdp(static::01B75A20, [00000000], [027FFE0C], [027FFE08]) called tport_tsend(00DD8D20) tpn = UDP/212.97.59.76:5060 2009-01-17 16:10:12 [NOTICE] sofia.c:3065 sofia_handle_sip_i_state() Channel [sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5] has been answered tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name UDP/212.97.59.76:5060 tport_vsend(00DD8D20): 1144 bytes of 1144 to udp/212.97.59.76:5060 tport_vsend returned 1144 send 1144 bytes to udp/[212.97.59.76]:5060 at 15:10:12.074853: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKe61f.f220cd02.0 Via: SIP/2.0/UDP 212.97.59.87:5060;branch=z9hG4bK57267a99;rport=5060 Record-Route: Record-Route: From: "+39020000000" ;tag=as302aa103 To: ;tag=N7F68303S0F5Q Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 102 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 266 v=0 o=FreeSWITCH 1110445663877587962 6324161778166975444 IN IP4 212.001.001.001 s=FreeSWITCH c=IN IP4 212.001.001.001 t=0 0 m=audio 29688 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (102) nta: timer shortened to 500 ms nua(02EDF6E0): call state changed: early -> completed, sent answer soa_get_local_sdp(static::01B75A20, [027FFE48], [027FFE44], [00000000]) called soa_get_params(static::01B75A20, ...) called nua(02EDF6E0): event i_state 200 OK nua(02EDF6E0): sent signal r_respond 2009-01-17 16:10:12 [NOTICE] switch_ivr_originate.c:1597 switch_ivr_originate() Channel [sofia/external/+39020000000 at sip.messagenet.it] has been answered tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 02F87F58 from (udp/10.0.3.6:5080) has 628 bytes, veclen = 1 recv 628 bytes from udp/[212.97.59.76]:5060 at 15:10:12.106094: ------------------------------------------------------------------------ ACK sip:mod_sofia at 212.001.001.001:5080;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKe61f.f220cd02.2 Via: SIP/2.0/UDP 212.97.59.87:5060;branch=z9hG4bK0498006e;rport=5060 From: "+39020000000" ;tag=as302aa103 To: ;tag=N7F68303S0F5Q Contact: Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 102 ACK User-Agent: alpha Max-Forwards: 69 Content-Length: 0 ------------------------------------------------------------------------ tport(00DD8D20): msg 02F87F58 (628 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received ACK sip:mod_sofia at 212.001.001.001:5080;transport=udp SIP/2.0 (CSeq 102) nta: ACK (102) is going to INVITE (102) soa_clear_remote_sdp(static::01B75A20) called nua(02EDF6E0): event i_ack 200 OK nua(02EDF6E0): call state changed: completed -> ready nua(02EDF6E0): event i_state 200 OK nua(02EDF6E0): event i_active 200 Call active tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F87F58 from (udp/192.168.0.74:5060) has 574 bytes, veclen = 1 recv 574 bytes from udp/[192.168.0.38]:51091 at 15:10:12.527848: ------------------------------------------------------------------------ SUBSCRIBE sip:1002 at 192.168.0.74 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.38:51091;branch=z9hG4bK-d8754z-08453f6c6e106020-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Alessandro Ceccarelli" From: "Alessandro Ceccarelli";tag=333d701e Call-ID: NTU2NDA2N2VjMzZlZTY4ODJlYmUwN2Y1MDc3NmFiOGY. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Event: message-summary Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F87F58 (574 bytes) from udp/192.168.0.38:5060/sip next=00000000 nta: received SUBSCRIBE sip:1002 at 192.168.0.74 SIP/2.0 (CSeq 1) nta: canonizing sip:1002 at 192.168.0.74 with contact nta: SUBSCRIBE (1) going to a default leg soa_clone(static::01AE57C0, 01AE28C0, 02F49AA0) called soa_set_params(static::02F88ED0, ...) called nta_leg_tcreate(02F039B0) nua(02F49AA0): adding notify usage with event message-summary nua(02F49AA0): event i_subscribe 100 Trying nua(): refresh notify after 600 seconds (in [600..600]) nua(): refresh notify after 600 seconds nua(02F49AA0): sent signal r_respond nua(02F49AA0): recv signal r_respond 202 Accepted soa_set_params(static::02F88ED0, ...) called tport_tsend(01AE9650) tpn = UDP/192.168.0.38:51091 tport_resolve addrinfo = 192.168.0.38:51091 tport(01AE9650): not found by name UDP/192.168.0.38:51091 tport_vsend(01AE9650): 813 bytes of 813 to udp/192.168.0.38:51091 tport_vsend returned 813 send 813 bytes to udp/[192.168.0.38]:51091 at 15:10:12.527848: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.0.38:51091;branch=z9hG4bK-d8754z-08453f6c6e106020-1---d8754z-;rport=51091 From: "Alessandro Ceccarelli";tag=333d701e To: "Alessandro Ceccarelli" ;tag=SBN9Fg4He48Fp Call-ID: NTU2NDA2N2VjMzZlZTY4ODJlYmUwN2Y1MDc3NmFiOGY. CSeq: 1 SUBSCRIBE Contact: "Alessandro Ceccarelli" Expires: 300 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=300 Content-Length: 0 ------------------------------------------------------------------------ nta: sent 202 Accepted for SUBSCRIBE (1) nta: timer set next to 4453 ms nta: timer I fired, terminate 200 response incoming_reclaim_all(00000000, 00000000, 027FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01D71768 from (udp/192.168.0.74:5060) has 801 bytes, veclen = 1 recv 801 bytes from udp/[192.168.0.52]:63214 at 15:10:19.104078: ------------------------------------------------------------------------ INVITE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4e76a82c441aae66-1---d8754z-;rport Max-Forwards: 70 Contact: To: "+39020000000";tag=r2UgeNKeHUjXa From: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 182 v=0 o=- 6 3 IN IP4 192.168.0.52 s=CounterPath X-Lite 3.0 c=IN IP4 0.0.0.0 t=0 0 m=audio 28768 RTP/AVP 3 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendonly ------------------------------------------------------------------------ tport(01AE9650): msg 01D71768 (801 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received INVITE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 (CSeq 2) nta: canonizing sip:mod_sofia at 192.168.0.74:5060 with contact nta: INVITE (2) going to existing leg nta: timer shortened to 200 ms soa_init_offer_answer(static::01D7FE78) called soa_set_remote_sdp(static::01D7FE78, 00000000, 01279F8B, 182) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 402 bytes of 402 to udp/192.168.0.52:63214 tport_vsend returned 402 send 402 bytes to udp/[192.168.0.52]:63214 at 15:10:19.119699: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4e76a82c441aae66-1---d8754z-;rport=63214 From: ;tag=8b08c857 To: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (2) nua(02EDF468): event i_invite 100 Trying nua(02EDF468): ready call updated: received received offer soa_get_remote_sdp(static::01D7FE78, [028FFC14], [028FFC10], [00000000]) called nua(02EDF468): event i_state 100 Trying nta: timer set next to 10538 ms nua(02EDF468): sent signal r_respond nua(02EDF468): recv signal r_respond 200 OK soa_set_params(static::01D7FE78, ...) called soa_set_user_sdp(static::01D7FE78, 00000000, 01B8170C, -1) called soa_generate_answer(static::01D7FE78) called soa_static_offer_answer_action(01D7FE78, soa_generate_answer): called soa_static(01D7FE78, soa_generate_answer): upgrade with remote description soa_static(01D7FE78, soa_generate_answer): marking rejected media soa_sdp_mode_set(028FDCF8, 01D705C8, ""): called soa_sdp_mode_set(028FDCF8, 01D705C8, ""): called soa_static(01D7FE78, soa_generate_answer): storing local description soa_activate(static::01D7FE78, (nil)) called soa_get_local_sdp(static::01D7FE78, [00000000], [028FFE0C], [028FFE08]) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 984 bytes of 984 to udp/192.168.0.52:63214 tport_vsend returned 984 send 984 bytes to udp/[192.168.0.52]:63214 at 15:10:19.369627: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4e76a82c441aae66-1---d8754z-;rport=63214 From: ;tag=8b08c857 To: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 276 v=0 o=FreeSWITCH 6699757194367626025 7696645903169343293 IN IP4 192.168.0.74 s=FreeSWITCH c=IN IP4 192.168.0.74 t=0 0 m=audio 29688 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=recvonly a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (2) nta: timer shortened to 500 ms nua(02EDF468): ready call updated: completed sent answer soa_get_local_sdp(static::01D7FE78, [028FFE48], [028FFE44], [00000000]) called soa_get_params(static::01D7FE78, ...) called nua(02EDF468): event i_state 200 OK tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F4BC48 from (udp/192.168.0.74:5060) has 497 bytes, veclen = 1 recv 497 bytes from udp/[192.168.0.52]:63214 at 15:10:19.478970: ------------------------------------------------------------------------ ACK sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-0a230c1b27666c30-1---d8754z-;rport Max-Forwards: 70 Contact: To: "+39020000000";tag=r2UgeNKeHUjXa From: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 2 ACK User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F4BC48 (497 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received ACK sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 (CSeq 2) nta: ACK (2) is going to INVITE (2) soa_clear_remote_sdp(static::01D7FE78) called nua(02EDF468): event i_ack 200 OK nua(02EDF468): ready call updated: ready nua(02EDF468): event i_state 200 OK nua(02EDF468): event i_active 200 Call active nta: timer set next to 4594 ms nta: timer I fired, terminate 200 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/5 term, 1/5 free nta: timer set next to 5352 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01D71768 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 01D71768 (4 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: timer J fired, terminate 401 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/4 term, 1/4 free nta: timer set next to 197 ms nta: timer J fired, terminate 200 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free nta: timer set next to 87 ms nta: timer J fired, terminate 202 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 13911 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01D71768 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 01D71768 (4 bytes) from udp/192.168.0.38:5060/sip next=00000000 tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01D71768 from (udp/192.168.0.74:5060) has 802 bytes, veclen = 1 recv 802 bytes from udp/[192.168.0.52]:63214 at 15:10:39.504451: ------------------------------------------------------------------------ INVITE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4b5f9a669369bb1f-1---d8754z-;rport Max-Forwards: 70 Contact: To: "+39020000000";tag=r2UgeNKeHUjXa From: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 3 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 183 v=0 o=- 6 4 IN IP4 192.168.0.52 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.52 t=0 0 m=audio 28768 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ tport(01AE9650): msg 01D71768 (802 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received INVITE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 (CSeq 3) nta: canonizing sip:mod_sofia at 192.168.0.74:5060 with contact nta: INVITE (3) going to existing leg nta: timer shortened to 200 ms soa_init_offer_answer(static::01D7FE78) called soa_set_remote_sdp(static::01D7FE78, 00000000, 01279F8B, 183) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 402 bytes of 402 to udp/192.168.0.52:63214 tport_vsend returned 402 send 402 bytes to udp/[192.168.0.52]:63214 at 15:10:39.504451: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4b5f9a669369bb1f-1---d8754z-;rport=63214 From: ;tag=8b08c857 To: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 3 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (3) nua(02EDF468): event i_invite 100 Trying nua(02EDF468): ready call updated: received received offer soa_get_remote_sdp(static::01D7FE78, [028FFC14], [028FFC10], [00000000]) called nua(02EDF468): event i_state 100 Trying nta: timer set next to 4351 ms nua(02EDF468): sent signal r_respond nua(02EDF468): recv signal r_respond 200 OK soa_set_params(static::01D7FE78, ...) called soa_set_user_sdp(static::01D7FE78, 00000000, 01B8170C, -1) called soa_generate_answer(static::01D7FE78) called soa_static_offer_answer_action(01D7FE78, soa_generate_answer): called soa_static(01D7FE78, soa_generate_answer): upgrade with remote description soa_sdp_mode_set(028FDCF8, 00DE0118, ""): called soa_static(01D7FE78, soa_generate_answer): storing local description soa_activate(static::01D7FE78, (nil)) called soa_get_local_sdp(static::01D7FE78, [00000000], [028FFE0C], [028FFE08]) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 972 bytes of 972 to udp/192.168.0.52:63214 tport_vsend returned 972 send 972 bytes to udp/[192.168.0.52]:63214 at 15:10:39.785620: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4b5f9a669369bb1f-1---d8754z-;rport=63214 From: ;tag=8b08c857 To: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 3 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 264 v=0 o=FreeSWITCH 6699757194367626025 7696645903169343294 IN IP4 192.168.0.74 s=FreeSWITCH c=IN IP4 192.168.0.74 t=0 0 m=audio 29688 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (3) nta: timer shortened to 500 ms nua(02EDF468): ready call updated: completed sent answer soa_get_local_sdp(static::01D7FE78, [028FFE48], [028FFE44], [00000000]) called soa_get_params(static::01D7FE78, ...) called nua(02EDF468): event i_state 200 OK tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F04E20 from (udp/192.168.0.74:5060) has 497 bytes, veclen = 1 recv 497 bytes from udp/[192.168.0.52]:63214 at 15:10:39.894964: ------------------------------------------------------------------------ ACK sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-b131c97e6c16e653-1---d8754z-;rport Max-Forwards: 70 Contact: To: "+39020000000";tag=r2UgeNKeHUjXa From: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 3 ACK User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F04E20 (497 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received ACK sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 (CSeq 3) nta: ACK (3) is going to INVITE (3) soa_clear_remote_sdp(static::01D7FE78) called nua(02EDF468): event i_ack 200 OK nua(02EDF468): ready call updated: ready nua(02EDF468): event i_state 200 OK nua(02EDF468): event i_active 200 Call active nta: timer set next to 3758 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F04E20 from (udp/192.168.0.74:5060) has 537 bytes, veclen = 1 recv 537 bytes from udp/[192.168.0.52]:63214 at 15:10:42.316141: ------------------------------------------------------------------------ BYE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-5f47b655ea7fe925-1---d8754z-;rport Max-Forwards: 70 Contact: To: "+39020000000";tag=r2UgeNKeHUjXa From: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 4 BYE User-Agent: X-Lite release 1100l stamp 47546 Reason: SIP;description="User Hung Up" Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F04E20 (537 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received BYE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 (CSeq 4) nta: canonizing sip:mod_sofia at 192.168.0.74:5060 with contact nta: BYE (4) going to existing leg tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 560 bytes of 560 to udp/192.168.0.52:63214 tport_vsend returned 560 send 560 bytes to udp/[192.168.0.52]:63214 at 15:10:42.316141: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-5f47b655ea7fe925-1---d8754z-;rport=63214 From: ;tag=8b08c857 To: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 4 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ nta: sent 200 OK for BYE (4) nua(02EDF468): event i_bye 200 Session Terminated nua(02EDF468): removing session usage nua(02EDF468): call state changed: ready -> terminated nua(02EDF468): event i_state 200 Session Terminated nua(02EDF468): event i_terminated 200 Session Terminated soa_destroy(static::01D7FE78) called nta_leg_destroy(02F47280) 2009-01-17 16:10:42 [NOTICE] sofia.c:3124 sofia_handle_sip_i_state() Hangup sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] nua(02EDF468): sent signal r_destroy nua(02EDF468): event i_terminated dropped nua(02EDF468): recv signal r_destroy nta_leg_destroy(00000000) 2009-01-17 16:10:42 [NOTICE] switch_ivr_bridge.c:955 switch_ivr_multi_threaded_bridge() Hangup sofia/external/+39020000000 at sip.messagenet.it [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-17 16:10:42 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 2 (sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5) Ended 2009-01-17 16:10:42 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 [CS_HANGUP] nua(02EDF6E0): recv signal r_bye soa_set_params(static::01B75A20, ...) called soa_terminate(static::01B75A20) called soa_init_offer_answer(static::01B75A20) called nta: selecting scheme sip tport_tsend(00DD8D20) tpn = */212.97.59.76:5060 nua(02EDF6E0): sent signal r_bye tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name */212.97.59.76:5060 tport_vsend(00DD8D20): 796 bytes of 796 to udp/212.97.59.76:5060 tport_vsend returned 796 send 796 bytes to udp/[212.97.59.76]:5060 at 15:10:42.331762: ------------------------------------------------------------------------ BYE sip:+39020000000 at 212.97.59.87 SIP/2.0 Via: SIP/2.0/UDP 212.001.001.001:5080;rport;branch=z9hG4bKpZU12DD5v1Z2D Route: Route: Max-Forwards: 70 From: ;tag=N7F68303S0F5Q To: "+39020000000" ;tag=as302aa103 Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 109984185 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ nta: sent BYE (109984185) to */212.97.59.76:5060 tport_pend(00DD8D20): pending 02F4B840 for udp/10.0.3.6:5080 (already 0) nta: timer set to 32000 ms nta: timer shortened to 500 ms 2009-01-17 16:10:42 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 1 (sofia/external/+39020000000 at sip.messagenet.it) Ended 2009-01-17 16:10:42 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/external/+39020000000 at sip.messagenet.it [CS_HANGUP] tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 02F8AAA8 from (udp/10.0.3.6:5080) has 612 bytes, veclen = 1 recv 612 bytes from udp/[212.97.59.76]:5060 at 15:10:42.347382: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 212.001.001.001:5080;rport=5080;branch=z9hG4bKpZU12DD5v1Z2D Record-Route: Record-Route: From: ;tag=N7F68303S0F5Q To: "+39020000000" ;tag=as302aa103 Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 109984185 BYE User-Agent: alpha Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 ------------------------------------------------------------------------ tport(00DD8D20): msg 02F8AAA8 (612 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received 200 OK for BYE (109984185) nta: 200 OK is going to a transaction nta_outgoing: RTT is 31.241 ms tport(00DD8D20): 02F4B840 by 01B70810 with 02F8AAA8 nua(02EDF6E0): event r_bye 200 OK nua(02EDF6E0): call state changed: terminating -> terminated nua(02EDF6E0): event i_state 200 to BYE nua(02EDF6E0): event i_terminated 200 to BYE nua(02EDF6E0): removing session usage soa_destroy(static::01B75A20) called nta_leg_destroy(02F004E0) nua: terminated session 02EDF6E0 nua(02EDF6E0): recv signal r_destroy nta_leg_destroy(00000000) nua(02EDF6E0): sent signal r_destroy nta: timer set next to 4500 ms nta: timer D fired, terminate INVITE (109984168) nta: timer F fired, terminating ACK (109984168) outgoing_reclaim_all(00000000, 00000000, 028FFEC0) nta_outgoing_timer: 0/0 resent, 1/1 tout, 1/1 term, 2/2 free nta: timer set next to 477 ms nta: timer J fired, terminate 202 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free nta: timer set next to 345 ms nta: timer I fired, terminate 200 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 29407 ms nta: timer K fired, terminate BYE (109984185) outgoing_reclaim_all(00000000, 00000000, 027FFEC0) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set 2009-01-17 16:10:57 [CONSOLE] switch_console.c:229 switch_console_process() Bye! 2009-01-17 16:10:57 [CONSOLE] switch_core.c:1459 switch_core_destroy() End existing sessions 2009-01-17 16:10:57 [CONSOLE] switch_core.c:1461 switch_core_destroy() Clean up modules. 2009-01-17 16:10:57 [CONSOLE] switch_core_memory.c:443 switch_core_memory_stop() Stopping memory pool queue. 2009-01-17 16:10:57 [CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim() Returning 3 recycled memory pool(s) 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'lua' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'luarun' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'lua' 2009-01-17 16:10:57 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_lua 2009-01-17 16:10:57 [NOTICE] switch_event.c:1189 switch_event_unbind() Event Binding deleted for mod_lua:MESSAGE_QUERY 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'limit' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'limit_hash' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'db' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'hash' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'group' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'limit_hash_usage' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'limit_usage' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'db' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'hash' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'group' 2009-01-17 16:10:57 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_limit has no shutdown routine 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'fifo' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'fifo' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'fifo_member' 2009-01-17 16:10:57 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_fifo 2009-01-17 16:10:57 [NOTICE] switch_event.c:1189 switch_event_unbind() Event Binding deleted for mod_fifo:PRESENCE_PROBE 2009-01-17 16:10:57 [NOTICE] switch_event.c:374 switch_event_free_subclass_detailed() Subclass reservation deleted for .\mod_fifo.c:fifo::info tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F053A0 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 02F053A0 (4 bytes) from udp/192.168.0.52:5060/sip next=00000000 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'expr' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_expr has no shutdown routine 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'aiff' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'au' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'avr' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'caf' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'htk' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'iff' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'mat' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'mat' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'paf' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'pvf' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'raw' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'sd2' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'sds' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'sf' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'voc' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'w64' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'wav' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'wav' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'wav' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'xi' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'r8' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'r16' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'r24' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'r32' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'gsm' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'ul' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'al' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_sndfile 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:700 switch_loadable_module_unprocess() Deleting Say interface 'en' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_say_en has no shutdown routine 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:434 switch_loadable_module_unprocess() Deleting Endpoint 'loopback' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_loopback 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'conference' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'conference_set_auto_outcall' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'conference' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:677 switch_loadable_module_unprocess() Deleting Chat interface 'conf' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_conference 2009-01-17 16:10:58 [NOTICE] switch_event.c:1189 switch_event_unbind() Event Binding deleted for mod_conference:PRESENCE_PROBE 2009-01-17 16:10:58 [NOTICE] switch_event.c:374 switch_event_free_subclass_detailed() Subclass reservation deleted for .\mod_conference.c:conference::maintenance 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PROXY-VID' (PROXY VIDEO PASS-THROUGH) 90000hz 0ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PROXY' (PROXY PASS-THROUGH) 8000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 11025hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 22050hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 8ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 6ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 4ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 2ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 8ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 6ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 4ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 2ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 8ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 6ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 4ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 2ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 8ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 6ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 4ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 2ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 30ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 50ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 60ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 30ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 50ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 60ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 30ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 50ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 60ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 70ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 80ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 90ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 100ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 110ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 120ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 30ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 50ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 60ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 70ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 80ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 90ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 100ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 110ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 120ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 30ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 50ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 60ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 70ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 80ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 90ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 100ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 110ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 120ms 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: CORE_PCM_MODULE 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'socket' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'event_sink' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_event_socket 2009-01-17 16:10:58 [NOTICE] switch_event.c:1189 switch_event_unbind() Event Binding deleted for mod_event_socket:ALL 2009-01-17 16:10:58 [NOTICE] mod_event_socket.c:2079 mod_event_socket_runtime() Shutting Down 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:95 switch_loadable_module_exec() Thread ended for mod_event_socket 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'AMR' (AMR) 8000hz 20ms 2009-01-17 16:10:59 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_amr has no shutdown routine 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:434 switch_loadable_module_unprocess() Deleting Endpoint 'error' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:434 switch_loadable_module_unprocess() Deleting Endpoint 'group' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:434 switch_loadable_module_unprocess() Deleting Endpoint 'user' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:479 switch_loadable_module_unprocess() Deleting Dialplan 'inline' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'privacy' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'flush_dtmf' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'hold' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'unhold' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'transfer' tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F053A0 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 02F053A0 (4 bytes) from udp/192.168.0.38:5060/sip next=00000000 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'check_acl' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'verbose_events' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'sleep' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'delay_echo' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'strftime' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'phrase' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'eval' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'pre_answer' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'answer' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'hangup' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'set_name' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'presence' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'log' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'info' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'event' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'export' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'set' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'set_global' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'set_profile_var' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'unset' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'ring_ready' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'remove_bugs' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'break' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'detect_speech' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'ivr' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'redirect' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'send_display' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'respond' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'deflect' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'queue_dtmf' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'send_dtmf' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'sched_hangup' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'sched_broadcast' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'sched_transfer' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'execute_extension' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'mkdir' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'soft_hold' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'bind_meta_app' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'unbind_meta_app' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'intercept' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'eavesdrop' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'three_way' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'set_user' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'stop_dtmf' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'start_dtmf' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'stop_dtmf_generate' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'start_dtmf_generate' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'stop_tone_detect' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'fax_detect' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'tone_detect' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'echo' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'park' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'park_state' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'gentones' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'playback' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'att_xfer' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'read' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'stop_record_session' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'record_session' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'record' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'stop_displace_session' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'displace_session' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'speak' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'clear_speech_cache' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'bridge' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'system' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'say' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'wait_for_silence' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'strepoch' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'chat' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'strftime' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'presence' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:677 switch_loadable_module_unprocess() Deleting Chat interface 'event' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:677 switch_loadable_module_unprocess() Deleting Chat interface 'api' 2009-01-17 16:11:03 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_dptools has no shutdown routine 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'G723' (G.723.1 6.3k) 8000hz 30ms 2009-01-17 16:11:03 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_g723_1 has no shutdown routine 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'console' 2009-01-17 16:11:03 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_logfile 2009-01-17 16:11:03 [NOTICE] switch_event.c:1189 switch_event_unbind() Event Binding deleted for mod_logfile:TRAP 2009-01-17 16:11:03 [INFO] mod_logfile.c:391 mod_logfile_shutdown() Closing C:\Program Files\FreeSwitch\log\freeswitch.log 2009-01-17 16:11:03 [DEBUG] switch_loadable_module.c:427 switch_loadable_module_unprocess() Write lock interface 'sofia' to wait for existing references. 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:434 switch_loadable_module_unprocess() Deleting Endpoint 'sofia' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'sofia' 2009-01-17 16:11:03 [DEBUG] switch_loadable_module.c:538 switch_loadable_module_unprocess() Write lock interface 'sofia' to wait for existing references. 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'sofia_contact' 2009-01-17 16:11:03 [DEBUG] switch_loadable_module.c:538 switch_loadable_module_unprocess() Write lock interface 'sofia_contact' to wait for existing references. 2009-01-17 16:11:03 [DEBUG] switch_loadable_module.c:669 switch_loadable_module_unprocess() Write lock interface 'sip' to wait for existing referen -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.log Type: application/octet-stream Size: 148830 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090117/1e623650/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: tport_sip.log Type: application/octet-stream Size: 21119 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090117/1e623650/attachment-0003.obj From cyberalby at gmail.com Sun Jan 18 05:29:56 2009 From: cyberalby at gmail.com (Alberto Ceccarelli) Date: Sun, 18 Jan 2009 14:29:56 +0100 Subject: [Freeswitch-users] Dialing Out Problem via Gateway In-Reply-To: <94c121cc0901170802i18c883cdv104660fc8f132315@mail.gmail.com> References: <94c121cc0901170802i18c883cdv104660fc8f132315@mail.gmail.com> Message-ID: <94c121cc0901180529q2d15da82g8c58c215fd82532@mail.gmail.com> > > Hi, > I've the same problem. > I've create the log with only the problem (the log is in attach). > One gateway (messagenet.it) and two internal account (1002 and 1008). > My dialplan: > > > > > > > > > > My Sip Profile: > > > > > > > > > > > > My freeswitch (I try 1.0.2 version and last SVN version) is runnig on > Windows Server 2003 32 bit with 2 NIC (1 interna 192.168.0.15 and 1 external > 10.0.3.6 (212.001.001.001)) > > I've change in the LOG files my external IP and my phone number. > > -- > Alby > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090118/b9bfdfeb/attachment-0001.html -------------- next part -------------- 2009-01-17 16:09:35 [INFO] switch_core_sqldb.c:487 switch_core_sqldb_start() Opening DB 2009-01-17 16:09:35 [DEBUG] switch_scheduler.c:214 switch_scheduler_add_task() Added task 1 heartbeat (core) to run at 1232204975 2009-01-17 16:09:35 [NOTICE] switch_scheduler.c:166 switch_scheduler_task_thread() Starting task thread 2009-01-17 16:09:35 [CONSOLE] switch_core.c:1270 switch_core_init_and_modload() Bringing up environment. 2009-01-17 16:09:35 [CONSOLE] switch_core.c:1271 switch_core_init_and_modload() Loading Modules. 2009-01-17 16:09:35 [INFO] switch_time.c:656 switch_load_timezones() Timezone loaded 530 definitions 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [CORE_SOFTTIMER_MODULE] 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:221 switch_loadable_module_process() Adding Timer 'soft' 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [CORE_PCM_MODULE] 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PROXY-VID' (PROXY VIDEO PASS-THROUGH) 90000hz 0ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PROXY' (PROXY PASS-THROUGH) 8000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 11025hz 40ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 22050hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 8ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 6ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 4ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 2ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 8ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 6ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 4ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 2ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 8ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 6ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 4ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 2ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 10ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 8ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 6ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 4ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 2ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 10ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 30ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 40ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 50ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 60ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 10ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 30ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 40ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 50ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 60ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 10ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 30ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 40ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 50ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 60ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 70ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 80ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 90ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 100ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 110ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 120ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 10ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 30ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 40ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 50ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 60ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 70ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 80ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 90ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 100ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 110ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMU' (G.711 ulaw) 8000hz 120ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 10ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 20ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 30ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 40ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 50ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 60ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 70ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 80ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 90ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 100ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 110ms 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'PCMA' (G.711 alaw) 8000hz 120ms 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_console] 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'console' 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_logfile] 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_enum] 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:203 switch_loadable_module_process() Adding Dialplan 'enum' 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'enum' 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'enum' 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'enum_auto' 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_cdr_csv] 2009-01-17 16:09:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_event_socket] 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'socket' 2009-01-17 16:09:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'event_sink' 2009-01-17 16:09:35 [NOTICE] sofia_reg.c:1872 sofia_reg_add_gateway() Added gateway 'messagenet.it' to profile 'external' 2009-01-17 16:09:35 [NOTICE] sofia.c:2234 config_sofia() Adding Alias [outbound] for profile [external] 2009-01-17 16:09:35 [NOTICE] sofia.c:2234 config_sofia() Adding Alias [nat] for profile [external] 2009-01-17 16:09:35 [NOTICE] sofia.c:2244 config_sofia() Started Profile external [sofia_reg_external] 2009-01-17 16:09:35 [NOTICE] sofia.c:1181 parse_domain_tag() Adding Alias [192.168.0.74] for profile [internal] 2009-01-17 16:09:35 [NOTICE] sofia.c:2234 config_sofia() Adding Alias [default] for profile [internal] 2009-01-17 16:09:35 [NOTICE] sofia.c:2244 config_sofia() Started Profile internal [sofia_reg_internal] su_socket_port_init(00DE3B58, 02491E00) called su_pthread_port_init(00DE3B58, 02491E00) called su_socket_port_init(0137A468, 02491E00) called su_pthread_port_init(0137A468, 02491E00) called soa_create("default", 0137A8D8, 00DD7F90) called soa_set_params(static::01ADEDC8, ...) called soa_set_params(static::01ADEDC8, ...) called nta_agent_create: initialized hash tables nta_agent_create: initialized transports nta_agent_create: initialized random identifiers nta_agent_create: initialized timer su_socket_port_init(01ADEFF8, 02491E00) called su_pthread_port_init(01ADEFF8, 02491E00) called nta_agent_create: initialized resolver tport_create(): 00DEC198 nta: master transport created tport(00DEC198) to */10.0.3.6:5080/sip tport(00DEC198): calling tport_listen for udp tport(00DEC198): new primary tport 00DD8D20 su_socket_port_init(01AE9DD8, 02491E00) called su_pthread_port_init(01AE9DD8, 02491E00) called soa_create("default", 01AE28C0, 01AE5580) called soa_set_params(static::01AE57C0, ...) called soa_set_params(static::01AE57C0, ...) called nta_agent_create: initialized hash tables nta_agent_create: initialized transports nta_agent_create: initialized random identifiers nta_agent_create: initialized timer nta_agent_create: initialized resolver tport_create(): 01AE9368 nta: master transport created tport(01AE9368) to */192.168.0.74:5060/sip tport(01AE9368): calling tport_listen for udp tport(01AE9368): new primary tport 01AE9650 tport(00DD8D20): listening at udp/10.0.3.6:5080/sip tport(00DEC198): calling tport_listen for tcp tport(00DEC198): new primary tport 01AE1540 tport(01AE9650): listening at udp/192.168.0.74:5060/sip tport(01AE9368): calling tport_listen for tcp tport(01AE9368): new primary tport 01AE17F8 tport(01AE17F8): listening at tcp/192.168.0.74:5060/sip nta: bound to (192.168.0.74:5060;transport=*) nta: agent_init_via: SIP/2.0/udp 192.168.0.74 (sip) nta: agent_init_via: SIP/2.0/tcp 192.168.0.74 (sip) nta: Via fields initialized nta: Contact header created nua_register: Adding contact URL '192.168.0.74' to list. tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01AE5FB8 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 01AE5FB8 (0 bytes) from udp/192.168.0.74:5060/sip next=00000000 tport(01AE1540): listening at tcp/10.0.3.6:5080/sip nta: bound to (212.001.001.001:5080;transport=*;maddr=10.0.3.6) nta: agent_init_via: SIP/2.0/udp 212.001.001.001:5080 (sip) nta: agent_init_via: SIP/2.0/tcp 212.001.001.001:5080 (sip) nta: Via fields initialized nta: Contact header created nua_register: Adding contact URL '212.001.001.001' to list. nua(00000000): sent signal r_set_params nua(00000000): recv signal r_set_params soa_set_params(static::01AE57C0, ...) called nua(00000000): event r_set_params 200 OK tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01AF0170 from (udp/10.0.3.6:5080) has 4 bytes, veclen = 1 tport(00DD8D20): bad msg 01AF0170 (0 bytes) from udp/10.0.3.6:5080/sip next=00000000 nua(00000000): sent signal r_set_params nua(00000000): recv signal r_set_params soa_set_params(static::01ADEDC8, ...) called nua(00000000): event r_set_params 200 OK 2009-01-17 16:09:36 [NOTICE] sofia_reg.c:265 sofia_reg_check_gateway() Registering messagenet.it nua(01AE32E0): sent signal r_register nua(01AE32E0): recv signal r_register soa_clone(static::01ADEDC8, 0137A8D8, 01AE32E0) called soa_set_params(static::01AEC380, ...) called soa_set_params(static::01AEC380, ...) called nta_leg_tcreate(01AED9C0) nua(01AE32E0): adding register usage nta: selecting scheme sip nta: for "sip.messagenet.it" query "_sip._udp.sip.messagenet.it" SRV nta: for "sip.messagenet.it" query "sip.messagenet.it" A nta: sip.messagenet.it. IN A 212.97.59.76 tport_tsend(00DD8D20) tpn = udp/212.97.59.76:5060 tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name udp/212.97.59.76:5060 tport_vsend(00DD8D20): 639 bytes of 639 to udp/212.97.59.76:5060 tport_vsend returned 639 send 639 bytes to udp/[212.97.59.76]:5060 at 15:09:36.803764: ------------------------------------------------------------------------ REGISTER sip:sip.messagenet.it;transport=udp SIP/2.0 Via: SIP/2.0/UDP 212.001.001.001:5080;rport;branch=z9hG4bKmD9FZQBy2FKXp Max-Forwards: 70 From: ;tag=mypD78F0vQSjc To: Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984152 REGISTER Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ nta: sent REGISTER (109984152) to udp/212.97.59.76:5060 tport_pend(00DD8D20): pending 01AF0170 for udp/10.0.3.6:5080 (already 0) nta: timer set to 32000 ms nta: timer shortened to 500 ms tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01B5D978 from (udp/10.0.3.6:5080) has 1278 bytes, veclen = 1 recv 564 bytes from udp/[212.97.59.76]:5060 at 15:09:36.803764: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.001.001.001:5080;rport=5080;branch=z9hG4bKmD9FZQBy2FKXp From: ;tag=mypD78F0vQSjc To: Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984152 REGISTER Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5060 "Noisy feedback tells: pid=15517 req_src_ip=212.001.001.001 req_src_port=5080 in_uri=sip:sip.messagenet.it;transport=udp out_uri=sip:sip.messagenet.it;transport=udp via_cnt==1" ------------------------------------------------------------------------ tport(00DD8D20): msg 01B5D978 (564 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received 100 Trying for REGISTER (109984152) nta: 100 Trying is going to a transaction nta_outgoing: RTT is 0 ms tport(00DD8D20): 01AF0170 by 01AED108 with 01B5D978 (preliminary) tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01B5D978 from (udp/10.0.3.6:5080) has 714 bytes, veclen = 1 recv 714 bytes from udp/[212.97.59.76]:5060 at 15:09:36.803764: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 212.001.001.001:5080;rport=5080;branch=z9hG4bKmD9FZQBy2FKXp From: ;tag=mypD78F0vQSjc To: ;tag=797b035a1d6b0049ed33903c5ef0eddf.062a Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984152 REGISTER WWW-Authenticate: Digest realm="sip.messagenet.it", nonce="4971f5dfd4d7b34dd71873b0bc184126b45c1700" Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5060 "Noisy feedback tells: pid=15517 req_src_ip=212.001.001.001 req_src_port=5080 in_uri=sip:sip.messagenet.it;transport=udp out_uri=sip:sip.messagenet.it;transport=udp via_cnt==1" ------------------------------------------------------------------------ tport(00DD8D20): msg 01B5D978 (714 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received 401 Unauthorized for REGISTER (109984152) nta: 401 Unauthorized is going to a transaction tport(00DD8D20): 01AF0170 by 01AED108 with 01B5D978 auth_digest_challenge_get(): got 2 nua(01AE32E0): event r_register 401 Unauthorized nua(01AE32E0): recv signal r_authenticate auth_digest_a1() has A1 = MD5(5343504:sip.messagenet.it:zanebap) = 7d86e12ae1b9db5b55b0653441352198 A2 = MD5(REGISTER:sip:sip.messagenet.it;transport=udp) auth_response: 937ac9aceb3126881d30de0ff6c9bcea = MD5(7d86e12ae1b9db5b55b0653441352198:4971f5dfd4d7b34dd71873b0bc184126b45c1700:c7eb2509efe71a9f29aa95f4d4019686) (qop=NONE) nta: selecting scheme sip nta: for "sip.messagenet.it" query "_sip._udp.sip.messagenet.it" SRV (cached) nta: for "sip.messagenet.it" query "sip.messagenet.it" A (cached) nta: sip.messagenet.it. IN A 212.97.59.76 tport_tsend(00DD8D20) tpn = udp/212.97.59.76:5060 tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name udp/212.97.59.76:5060 tport_vsend(00DD8D20): 861 bytes of 861 to udp/212.97.59.76:5060 tport_vsend returned 861 send 861 bytes to udp/[212.97.59.76]:5060 at 15:09:36.803764: ------------------------------------------------------------------------ REGISTER sip:sip.messagenet.it;transport=udp SIP/2.0 nua(01AE32E0): sent signal r_authenticate Via: SIP/2.0/UDP 212.001.001.001:5080;rport;branch=z9hG4bKNp280jv1Zr9Fj Max-Forwards: 70 From: ;tag=mypD78F0vQSjc To: Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984153 REGISTER Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Authorization: Digest username="5343504", realm="sip.messagenet.it", nonce="4971f5dfd4d7b34dd71873b0bc184126b45c1700", algorithm=MD5, uri="sip:sip.messagenet.it;transport=udp", response="937ac9aceb3126881d30de0ff6c9bcea" Content-Length: 0 ------------------------------------------------------------------------ nta: sent REGISTER (109984153) to udp/212.97.59.76:5060 tport_pend(00DD8D20): pending 01B58858 for udp/10.0.3.6:5080 (already 0) tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01B59988 from (udp/10.0.3.6:5080) has 564 bytes, veclen = 1 recv 564 bytes from udp/[212.97.59.76]:5060 at 15:09:36.819385: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.001.001.001:5080;rport=5080;branch=z9hG4bKNp280jv1Zr9Fj From: ;tag=mypD78F0vQSjc To: Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984153 REGISTER Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5060 "Noisy feedback tells: pid=15519 req_src_ip=212.001.001.001 req_src_port=5080 in_uri=sip:sip.messagenet.it;transport=udp out_uri=sip:sip.messagenet.it;transport=udp via_cnt==1" ------------------------------------------------------------------------ tport(00DD8D20): msg 01B59988 (564 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received 100 Trying for REGISTER (109984153) nta: 100 Trying is going to a transaction nta_outgoing: RTT is 15.621 ms tport(00DD8D20): 01B58858 by 01B5C388 with 01B59988 (preliminary) tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01B59988 from (udp/10.0.3.6:5080) has 715 bytes, veclen = 1 recv 715 bytes from udp/[212.97.59.76]:5060 at 15:09:36.819385: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 212.001.001.001:5080;rport=5080;branch=z9hG4bKNp280jv1Zr9Fj From: ;tag=mypD78F0vQSjc To: ;tag=797b035a1d6b0049ed33903c5ef0eddf.bdba Call-ID: c66dcbcd-e144-354e-b169-0723760063cf CSeq: 109984153 REGISTER Date: Sat, 17 Jan 2009 15:09:39 GMT Contact: ;q=0.5;expires=3600 Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 212.97.59.76:5060 "Noisy feedback tells: pid=15519 req_src_ip=212.001.001.001 req_src_port=5080 in_uri=sip:sip.messagenet.it;transport=udp out_uri=sip:sip.messagenet.it;transport=udp via_cnt==1" ------------------------------------------------------------------------ tport(00DD8D20): msg 01B59988 (715 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received 200 OK for REGISTER (109984153) nta: 200 OK is going to a transaction tport(00DD8D20): 01B58858 by 01B5C388 with 01B59988 nua(): refresh register after 2568 seconds (in [900..2700]) nua(): refresh register after 2568 seconds nua(01AE32E0): event r_register 200 OK 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_sofia] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'sofia' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sofia' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sofia_contact' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'sip' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:394 switch_loadable_module_process() Adding Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1] 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_loopback] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'loopback' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_commands] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'group_call' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'in_group' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_flush_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'md5' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'hupall' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'strftime_tz' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'originate' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'tone_detect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_kill' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_park' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'reloadacl' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'reloadxml' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'unload' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'reload' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'load' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_transfer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'pause' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'break' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'show' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'complete' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'alias' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'status' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_session_heartbeat' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_bridge' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_setvar' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_setvar_multi' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_getvar' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_dump' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'global_setvar' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'global_getvar' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_displace' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_record' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_broadcast' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_hold' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_display' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_media' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'fsctl' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'help' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'version' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sched_hangup' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sched_broadcast' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sched_transfer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'create_uuid' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sched_api' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'unsched_api' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'bgapi' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'sched_del' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'xml_wrap' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'is_lan_addr' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'cond' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'regex' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'acl' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_chat' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_deflect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'find_user_xml' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'user_exists' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'xml_locate' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'user_data' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'url_encode' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'url_decode' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'module_exists' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'domain_exists' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'uuid_send_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'eval' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'system' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'time_test' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_conference] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'conference' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'conference_set_auto_outcall' 2009-01-17 16:09:37 [CONSOLE] sofia_presence.c:621 sofia_presence_event_thread_run() Event Thread Started 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'conference' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'conf' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_dptools] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'error' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'group' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'user' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:203 switch_loadable_module_process() Adding Dialplan 'inline' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'privacy' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'flush_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'hold' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'unhold' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'transfer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'check_acl' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'verbose_events' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'sleep' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'delay_echo' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'strftime' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'phrase' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'eval' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'pre_answer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'answer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'hangup' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'set_name' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'presence' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'log' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'info' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'event' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'export' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'set' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'set_global' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'set_profile_var' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'unset' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'ring_ready' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'remove_bugs' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'break' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'detect_speech' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'ivr' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'redirect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'send_display' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'respond' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'deflect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'queue_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'send_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'sched_hangup' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'sched_broadcast' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'sched_transfer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'execute_extension' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'mkdir' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'soft_hold' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'bind_meta_app' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'unbind_meta_app' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'intercept' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'eavesdrop' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'three_way' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'set_user' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'stop_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'start_dtmf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'stop_dtmf_generate' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'start_dtmf_generate' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'stop_tone_detect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'fax_detect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'tone_detect' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'echo' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'park' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'park_state' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'gentones' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'playback' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'att_xfer' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'read' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'stop_record_session' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'record_session' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'record' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'stop_displace_session' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'displace_session' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'speak' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'clear_speech_cache' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'bridge' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'system' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'say' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'wait_for_silence' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'strepoch' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'chat' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'strftime' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'presence' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'event' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'api' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_expr] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'expr' 2009-01-17 16:09:37 [INFO] mod_fifo.c:1811 load_config() cool_fifo at 192.168.0.74 configured 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_fifo] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'fifo' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'fifo' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'fifo_member' 2009-01-17 16:09:37 [INFO] mod_voicemail.c:765 load_config() Added Profile default 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_voicemail] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'voicemail' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'voicemail' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'voicemail_inject' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'vm_boxcount' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_limit] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'limit' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'limit_hash' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'db' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'hash' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'group' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'limit_hash_usage' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'limit_usage' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'db' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'hash' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'group' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_esf] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'esf_page_group' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_fsv] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'play_fsv' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'record_fsv' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_dialplan_xml] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:203 switch_loadable_module_process() Adding Dialplan 'XML' nta: timer set next to 4485 ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_dialplan_asterisk] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'SIP' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'IAX2' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:203 switch_loadable_module_process() Adding Dialplan 'asterisk' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'Dial' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'Goto' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'AvoidingDeadlock' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_voipcodecs] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 16000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'DVI4' (ADPCM (IMA)) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-16' (G.726 16k (AAL2)) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-16' (G.726 16k) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-24' (G.726 24k (AAL2)) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-32' (G.726 32k (AAL2)) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-32' (G.726 32k) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AAL2-G726-40' (G.726 40k (AAL2)) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G726-40' (G.726 40k) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G722' (G.722) 16000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'GSM' (GSM) 8000hz 120ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'LPC' (LPC-10) 8000hz 90ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_g723_1] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G723' (G.723.1 6.3k) 8000hz 30ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_g729] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 10ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 40ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 50ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 60ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 70ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 80ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 90ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 100ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 110ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 120ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_amr] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'AMR' (AMR) 8000hz 20ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_ilbc] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC' (iLBC) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC' (iLBC) 8000hz 30ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC20ms' (iLBC) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC102' (iLBC) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC' (iLBC) 8000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'iLBC' (iLBC) 8000hz 20ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_speex] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'SPEEX' (Speex) 32000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'SPEEX' (Speex) 16000hz 20ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'SPEEX' (Speex) 8000hz 20ms 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_h26x] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'H264' (H.264 Video (passthru)) 90000hz 0ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'H263' (H.263 Video (passthru)) 90000hz 0ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'H263-1998' (H.263+ Video (passthru)) 90000hz 0ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'H263-2000' (H.263++ Video (passthru)) 90000hz 0ms 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'H261' (H.261 Video (passthru)) 90000hz 0ms 2009-01-17 16:09:37 [INFO] mod_sndfile.c:308 setup_formats() LibSndFile Version : libsndfile-1.0.12 Supported Formats ================================================================================ AIFF (Apple/SGI) (extension "aiff") AU (Sun/NeXT) (extension "au") AVR (Audio Visual Research) (extension "avr") CAF (Apple Core Audio File) (extension "caf") HTK (HMM Tool Kit) (extension "htk") IFF (Amiga IFF/SVX8/SV16) (extension "iff") MAT4 (GNU Octave 2.0 / Matlab 4.2) (extension "mat") MAT5 (GNU Octave 2.1 / Matlab 5.0) (extension "mat") PAF (Ensoniq PARIS) (extension "paf") PVF (Portable Voice Format) (extension "pvf") RAW (header-less) (extension "raw") SD2 (Sound Designer II) (extension "sd2") SDS (Midi Sample Dump Standard) (extension "sds") SF (Berkeley/IRCAM/CARL) (extension "sf") VOC (Creative Labs) (extension "voc") W64 (SoundFoundry WAVE 64) (extension "w64") WAV (Microsoft) (extension "wav") WAV (NIST Sphere) (extension "wav") WAVEX (Microsoft) (extension "wav") XI (FastTracker 2) (extension "xi") ================================================================================ 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_sndfile] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'aiff' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'au' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'avr' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'caf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'htk' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'iff' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'mat' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'mat' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'paf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'pvf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'raw' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'sd2' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'sds' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'sf' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'voc' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'w64' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'wav' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'wav' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'wav' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'xi' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'r8' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'r16' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'r24' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'r32' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'gsm' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'ul' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'al' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_native_file] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'H263' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'iLBC20ms' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'AMR' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'H263-1998' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'SPEEX' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G729' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G723' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G726-16' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'H261' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'AAL2-G726-16' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'DVI4' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'PCMA' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'PCMU' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'L16' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'iLBC20ms' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'PROXY' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'PROXY-VID' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'AAL2-G726-24' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'AAL2-G726-32' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'H263-2000' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'H264' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G726-32' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'iLBC20ms' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G722' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'AAL2-G726-40' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'G726-40' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'GSM' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'LPC' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_local_stream] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'stop_local_stream' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'start_local_stream' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'local_stream' 2009-01-17 16:09:37 [CONSOLE] mod_local_stream.c:142 read_stream_thread() Can't open directory: C:\Program Files\FreeSwitch/sounds/music/16000 2009-01-17 16:09:37 [CONSOLE] mod_local_stream.c:142 read_stream_thread() Can't open directory: C:\Program Files\FreeSwitch/sounds/music/32000 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_tone_stream] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'tone_stream' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding File Format 'silence_stream' 2009-01-17 16:09:37 [CONSOLE] mod_spidermonkey.c:948 sm_load_file() Successfully Loaded [C:\Program Files\FreeSwitch\mod\mod_spidermonkey_teletone.dll] 2009-01-17 16:09:37 [CONSOLE] mod_spidermonkey.c:948 sm_load_file() Successfully Loaded [C:\Program Files\FreeSwitch\mod\mod_spidermonkey_core_db.dll] 2009-01-17 16:09:37 [CONSOLE] mod_spidermonkey.c:948 sm_load_file() Successfully Loaded [C:\Program Files\FreeSwitch\mod\mod_spidermonkey_socket.dll] 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_spidermonkey] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'javascript' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'jsrun' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'jsapi' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_lua] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'lua' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'luarun' 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'lua' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_say_en] 2009-01-17 16:09:37 [NOTICE] switch_loadable_module.c:372 switch_loadable_module_process() Adding Say interface 'en' 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:119 switch_loadable_module_runtime() Starting runtime thread for CORE_SOFTTIMER_MODULE 2009-01-17 16:09:37 [CONSOLE] switch_loadable_module.c:119 switch_loadable_module_runtime() Starting runtime thread for mod_event_socket 2009-01-17 16:09:37 [CONSOLE] switch_core.c:865 switch_load_network_lists() Created ip list dl-candidates default (allow) 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 10.0.0.0/8 (deny) to list dl-candidates 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 172.16.0.0/12 (deny) to list dl-candidates 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 192.168.0.0/16 (deny) to list dl-candidates 2009-01-17 16:09:37 [CONSOLE] switch_core.c:865 switch_load_network_lists() Created ip list rfc1918 default (deny) 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 10.0.0.0/8 (allow) to list rfc1918 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 172.16.0.0/12 (allow) to list rfc1918 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 192.168.0.0/16 (allow) to list rfc1918 2009-01-17 16:09:37 [CONSOLE] switch_core.c:865 switch_load_network_lists() Created ip list lan default (allow) 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 192.168.42.0/24 (deny) to list lan 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 192.168.42.42/32 (allow) to list lan 2009-01-17 16:09:37 [CONSOLE] switch_core.c:865 switch_load_network_lists() Created ip list strict default (deny) 2009-01-17 16:09:37 [NOTICE] switch_core.c:948 switch_load_network_lists() Adding 208.102.123.124/32 (allow) to list strict 2009-01-17 16:09:37 [CONSOLE] switch_core.c:865 switch_load_network_lists() Created ip list domains default (deny) 2009-01-17 16:09:37 [CONSOLE] switch_core.c:1287 switch_core_init_and_modload() FreeSWITCH Version 1.0.trunk (UNKNOWN) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] freeswitch at inetmail2> nta: timer K fired, terminate REGISTER (109984152) outgoing_reclaim_all(00000000, 00000000, 027FFEC0) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 1 ms nta: timer K fired, terminate REGISTER (109984153) outgoing_reclaim_all(00000000, 00000000, 027FFEC0) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01B59988 from (udp/192.168.0.74:5060) has 543 bytes, veclen = 1 recv 543 bytes from udp/[192.168.0.52]:63214 at 15:09:57.844578: ------------------------------------------------------------------------ REGISTER sip:192.168.0.74 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-10626c213e281d27-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Alberto" From: "Alberto";tag=544adf7f Call-ID: ZjNlNGY5ZGE5OGI5MjFkNTc3Y2RjMzhhMDc4MmM1NDc. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 01B59988 (543 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received REGISTER sip:192.168.0.74 SIP/2.0 (CSeq 1) nta: canonizing sip:192.168.0.74 with contact nta: REGISTER (1) going to a default leg soa_clone(static::01AE57C0, 01AE28C0, 01B6FBD8) called soa_set_params(static::01B5EF70, ...) called nua(01B6FBD8): event i_register 100 Trying nua(01B6FBD8): sent signal r_respond nua(01B6FBD8): recv signal r_respond 401 Unauthorized soa_set_params(static::01B5EF70, ...) called nua(01B6FBD8): sent signal r_destroy tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 667 bytes of 667 to udp/192.168.0.52:63214 tport_vsend returned 667 send 667 bytes to udp/[192.168.0.52]:63214 at 15:09:57.844578: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-10626c213e281d27-1---d8754z-;rport=63214 From: "Alberto";tag=544adf7f To: "Alberto" ;tag=mypD78F0vQSjc Call-ID: ZjNlNGY5ZGE5OGI5MjFkNTc3Y2RjMzhhMDc4MmM1NDc. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="192.168.0.74", nonce="8d55768a-5992-1c4d-912e-92961f1f4e5e", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ nta: sent 401 Unauthorized for REGISTER (1) nta: timer set to 32000 ms nua(01B6FBD8): recv signal r_destroy nta_leg_destroy(00000000) soa_destroy(static::01B5EF70) called tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01B5D978 from (udp/192.168.0.74:5060) has 792 bytes, veclen = 1 recv 792 bytes from udp/[192.168.0.52]:63214 at 15:09:58.047644: ------------------------------------------------------------------------ REGISTER sip:192.168.0.74 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-53075b0a800fd422-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Alberto" From: "Alberto";tag=544adf7f Call-ID: ZjNlNGY5ZGE5OGI5MjFkNTc3Y2RjMzhhMDc4MmM1NDc. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="1008",realm="192.168.0.74",nonce="8d55768a-5992-1c4d-912e-92961f1f4e5e",uri="sip:192.168.0.74",response="55d0982a97d14c1b35e41e9617710f4f",cnonce="d4204f159a249cc80d1ea728bac88ad2",nc=00000001,qop=auth,algorithm=MD5 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 01B5D978 (792 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received REGISTER sip:192.168.0.74 SIP/2.0 (CSeq 2) nta: canonizing sip:192.168.0.74 with contact nta: REGISTER (2) going to a default leg soa_clone(static::01AE57C0, 01AE28C0, 01B6FBD8) called soa_set_params(static::01B75A20, ...) called nua(01B6FBD8): event i_register 100 Trying nua(01B6FBD8): sent signal r_respond nua(01B6FBD8): recv signal r_respond 200 OK soa_set_params(static::01B75A20, ...) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 654 bytes of 654 to udp/192.168.0.52:63214 tport_vsend returned 654 nua(01B6FBD8): sent signal r_destroy send 654 bytes to udp/[192.168.0.52]:63214 at 15:09:58.047644: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-53075b0a800fd422-1---d8754z-;rport=63214 From: "Alberto";tag=544adf7f To: "Alberto" ;tag=N7F68303S0F5Q Call-ID: ZjNlNGY5ZGE5OGI5MjFkNTc3Y2RjMzhhMDc4MmM1NDc. CSeq: 2 REGISTER Contact: ;expires=3600 Date: Sat, 17 Jan 2009 15:09:58 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ nta: sent 200 OK for REGISTER (2) nua(01B6FBD8): recv signal r_destroy nta_leg_destroy(00000000) soa_destroy(static::01B75A20) called nua(01B6FBD8): sent signal r_notify nua(01B6FBD8): recv signal r_notify soa_clone(static::01AE57C0, 01AE28C0, 01B6FBD8) called soa_set_params(static::01B75A20, ...) called soa_set_params(static::01B75A20, ...) called nta_leg_tcreate(00DD7010) nua(01B6FBD8): adding notify usage with event message-summary nta: selecting scheme sip tport_tsend(01AE9650) tpn = */192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name */192.168.0.52:63214 tport_vsend(01AE9650): 887 bytes of 887 to udp/192.168.0.52:63214 tport_vsend returned 887 send 887 bytes to udp/[192.168.0.52]:63214 at 15:09:58.141367: ------------------------------------------------------------------------ NOTIFY sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.74;rport;branch=z9hG4bKmD9FZQBy2FKXp Max-Forwards: 70 From: ;tag=pg9yaZH7p95QK To: Call-ID: beeb792e-5f4b-122c-2780-39a48cb53b8d CSeq: 109984163 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;timeout Content-Type: application/simple-message-summary Content-Length: 64 Messages-Waiting: no Message-Account: sip:1008 at 192.168.0.74 ------------------------------------------------------------------------ nta: sent NOTIFY (109984163) to */192.168.0.52:63214 tport_pend(01AE9650): pending 01B72C38 for udp/192.168.0.74:5060 (already 0) nta: timer shortened to 500 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F2FC50 from (udp/192.168.0.74:5060) has 546 bytes, veclen = 1 recv 546 bytes from udp/[192.168.0.52]:63214 at 15:09:58.141367: ------------------------------------------------------------------------ SUBSCRIBE sip:1008 at 192.168.0.74 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-f3490746570af404-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Alberto" From: "Alberto";tag=d11f6f44 Call-ID: ZDY2ZWMwN2M0MTkwZDA1NDJkOGNhZmNmYTg5NDgyNGY. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Event: message-summary Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F2FC50 (546 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received SUBSCRIBE sip:1008 at 192.168.0.74 SIP/2.0 (CSeq 1) nta: canonizing sip:1008 at 192.168.0.74 with contact nta: SUBSCRIBE (1) going to a default leg soa_clone(static::01AE57C0, 01AE28C0, 01B72720) called soa_set_params(static::01B5EF70, ...) called nta_leg_tcreate(01B6F530) nua(01B72720): adding notify usage with event message-summary nua(01B72720): event i_subscribe 100 Trying nua(): refresh notify after 600 seconds (in [600..600]) nua(): refresh notify after 600 seconds nua(01B72720): recv signal r_respond 202 Accepted soa_set_params(static::01B5EF70, ...) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 771 bytes of 771 to udp/192.168.0.52:63214 tport_vsend returned 771 send 771 bytes to udp/[192.168.0.52]:63214 at 15:09:58.141367: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-f3490746570af404-1---d8754z-;rport=63214 From: "Alberto";tag=d11f6f44 To: "Alberto" ;tag=QS2Qct2amjvaF Call-ID: ZDY2ZWMwN2M0MTkwZDA1NDJkOGNhZmNmYTg5NDgyNGY.nua(01B72720): sent signal r_respond CSeq: 1 SUBSCRIBE Contact: "Alberto" Expires: 300 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=300 Content-Length: 0 ------------------------------------------------------------------------ nta: sent 202 Accepted for SUBSCRIBE (1) tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01D71768 from (udp/192.168.0.74:5060) has 350 bytes, veclen = 1 recv 350 bytes from udp/[192.168.0.52]:63214 at 15:09:58.141367: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.74;rport=5060;branch=z9hG4bKmD9FZQBy2FKXp Contact: To: ;tag=321bea76 From: ;tag=pg9yaZH7p95QK Call-ID: beeb792e-5f4b-122c-2780-39a48cb53b8d CSeq: 109984163 NOTIFY User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 01D71768 (350 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received 200 OK for NOTIFY (109984163) nta: 200 OK is going to a transaction nta_outgoing: RTT is 15.621 ms tport(01AE9650): 01B72C38 by 01AED108 with 01D71768 nua(01B6FBD8): event r_notify 200 OK nua(01B6FBD8): removing notify usage with event message-summary nta_leg_destroy(00DD7010) nua(01B6FBD8): sent signal r_destroy nua(01B6FBD8): recv signal r_destroy nta_leg_destroy(00000000) soa_destroy(static::01B75A20) called nta: timer set next to 4484 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F4BC48 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 02F4BC48 (4 bytes) from udp/192.168.0.38:5060/sip next=00000000 nta: timer K fired, terminate NOTIFY (109984163) outgoing_reclaim_all(00000000, 00000000, 028FFEC0) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer set next to 26689 ms tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 01D71768 from (udp/10.0.3.6:5080) has 1168 bytes, veclen = 1 recv 1168 bytes from udp/[212.97.59.76]:5060 at 15:10:08.076005: ------------------------------------------------------------------------ INVITE sip:5343504 at 212.001.001.001:5080;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKe61f.f220cd02.0 Via: SIP/2.0/UDP 212.97.59.87:5060;branch=z9hG4bK57267a99;rport=5060 From: "+39020000000" ;tag=as302aa103 To: Contact: Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 102 INVITE User-Agent: alpha Max-Forwards: 69 Date: Sat, 17 Jan 2009 15:10:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 395 v=0 o=root 3859 3859 IN IP4 212.97.59.87 s=session c=IN IP4 212.97.59.91 t=0 0 m=audio 38196 RTP/AVP 18 3 97 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes ------------------------------------------------------------------------ tport(00DD8D20): msg 01D71768 (1168 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received INVITE sip:5343504 at 212.001.001.001:5080;transport=udp SIP/2.0 (CSeq 102) nta: canonizing sip:5343504 at 212.001.001.001:5080 with contact nta: INVITE (102) going to a default leg nta: timer set to 200 ms soa_clone(static::01ADEDC8, 0137A8D8, 02EDF6E0) called soa_set_params(static::01B75A20, ...) called nta_leg_tcreate(02F004E0) soa_init_offer_answer(static::01B75A20) called soa_set_remote_sdp(static::01B75A20, 00000000, 02F47CAD, 395) called nua(02EDF6E0): adding session usage tport_tsend(00DD8D20) tpn = UDP/212.97.59.76:5060 tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name UDP/212.97.59.76:5060 tport_vsend(00DD8D20): 546 bytes of 546 to udp/212.97.59.76:5060 tport_vsend returned 546 send 546 bytes to udp/[212.97.59.76]:5060 at 15:10:08.076005: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKe61f.f220cd02.0 Via: SIP/2.0/UDP 212.97.59.87:5060;branch=z9hG4bK57267a99;rport=5060 Record-Route: Record-Route: From: "+39020000000" ;tag=as302aa103 To: Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (102) nua(02EDF6E0): event i_invite 100 Trying nua(02EDF6E0): call state changed: init -> received, received offer soa_get_remote_sdp(static::01B75A20, [027FFC14], [027FFC10], [00000000]) called nua(02EDF6E0): event i_state 100 Trying 2009-01-17 16:10:08 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/external/+39020000000 at sip.messagenet.it [711a2124-077d-f442-ae20-e63b2273f326] 2009-01-17 16:10:08 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing +39020000000->5343504 in context public 2009-01-17 16:10:08 [NOTICE] switch_ivr.c:1255 switch_ivr_session_transfer() Transfer sofia/external/+39020000000 at sip.messagenet.it to XML[1008 at default] 2009-01-17 16:10:08 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing +39020000000->1008 in context default 2009-01-17 16:10:08 [ERR] switch_regex.c:94 switch_regex_perform() COMPILE ERROR: 1 [nothing to repeat][^+39020000000$] 2009-01-17 16:10:08 [INFO] switch_ivr_async.c:1644 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-01-17 16:10:08 [INFO] switch_ivr_async.c:1644 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::C:Program FilesFreeSwitch/recordings/+39020000000.2009-01-17-16-10-08.wav 2009-01-17 16:10:08 [INFO] switch_ivr_async.c:1644 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2009-01-17 16:10:08 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 [698623ef-686c-9c4d-93c7-3cc6db803902] nua(02EDF468): sent signal r_invite nua(02EDF468): recv signal r_invite soa_clone(static::01AE57C0, 01AE28C0, 02EDF468) called soa_set_params(static::01D7FE78, ...) called soa_set_params(static::01D7FE78, ...) called soa_set_user_sdp(static::01D7FE78, 00000000, 012733C2, -1) called soa_set_capability_sdp(static::01D7FE78, 00000000, 012733C2, -1) called nta_leg_tcreate(02F47280) nua(02EDF468): adding session usage soa_init_offer_answer(static::01D7FE78) called soa_generate_offer(static::01D7FE78, 0) called soa_static_offer_answer_action(01D7FE78, soa_generate_offer): called soa_static(01D7FE78, soa_generate_offer): generating local description soa_static(01D7FE78, soa_generate_offer): upgrade with local description soa_sdp_mode_set(028FDCAC, 00000000, ""): called soa_static(01D7FE78, soa_generate_offer): storing local description soa_get_local_sdp(static::01D7FE78, [00000000], [028FFDC4], [028FFDC0]) called nta: selecting scheme sip tport_tsend(01AE9650) tpn = */192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name */192.168.0.52:63214 tport_vsend(01AE9650): 1261 bytes of 1261 to udp/192.168.0.52:63214 tport_vsend returned 1261 send 1261 bytes to udp/[192.168.0.52]:63214 at 15:10:08.216590: ------------------------------------------------------------------------ INVITE sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.74;rport;branch=z9hG4bKNp280jv1Zr9Fj Max-Forwards: 67 From: "+39020000000" ;tag=r2UgeNKeHUjXa To: Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 109984168 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 333 Remote-Party-ID: "+39020000000" ;screen=yes;privacy=off v=0 o=FreeSWITCH 6699757194367626025 7696645903169343292 IN IP4 192.168.0.74 s=FreeSWITCH c=IN IP4 192.168.0.74 t=0 0 m=audio 29688 RTP/AVP 3 9 0 8 101 13 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ nta: sent INVITE (109984168) to */192.168.0.52:63214 tport_pend(01AE9650): pending 02FC1940 for udp/192.168.0.74:5060 (already 0) nta: timer shortened to 500 ms nua(02EDF468): call state changed: init -> calling, sent offer soa_get_local_sdp(static::01D7FE78, [028FFDC4], [028FFDC0], [00000000]) called nua(02EDF468): event i_state INVITE sent nta: timer not set 2009-01-17 16:10:08 [INFO] mod_sofia.c:1294 sofia_receive_message() Asked to send early media by sofia/external/+39020000000 at sip.messagenet.it 2009-01-17 16:10:08 [INFO] mod_sofia.c:1335 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1232175320 1232175321 IN IP4 212.001.001.001 s=FreeSWITCH c=IN IP4 212.001.001.001 t=0 0 m=audio 29688 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-01-17 16:10:08 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Ring-Ready sofia/external/+39020000000 at sip.messagenet.it! 2009-01-17 16:10:08 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Pre-Answer sofia/external/+39020000000 at sip.messagenet.it! nua(02EDF6E0): sent signal r_respond nua(02EDF6E0): recv signal r_respond 183 Session Progress soa_set_params(static::01B75A20, ...) called soa_set_user_sdp(static::01B75A20, 00000000, 00DECA53, -1) called soa_set_capability_sdp(static::01B75A20, 00000000, 00DECA53, -1) called soa_generate_answer(static::01B75A20) called soa_static_offer_answer_action(01B75A20, soa_generate_answer): called soa_static(01B75A20, soa_generate_answer): generating local description soa_static(01B75A20, soa_generate_answer): upgrade with remote description soa_sdp_mode_set(027FDCF8, 01274870, ""): called soa_static(01B75A20, soa_generate_answer): storing local description soa_activate(static::01B75A20, (nil)) called soa_get_local_sdp(static::01B75A20, [00000000], [027FFE0C], [027FFE08]) called tport_tsend(00DD8D20) tpn = UDP/212.97.59.76:5060 tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name UDP/212.97.59.76:5060 tport_vsend(00DD8D20): 1170 bytes of 1170 to udp/212.97.59.76:5060 tport_vsend returned 1170 send 1170 bytes to udp/[212.97.59.76]:5060 at 15:10:08.310313: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKe61f.f220cd02.0 Via: SIP/2.0/UDP 212.97.59.87:5060;branch=z9hG4bK57267a99;rport=5060 Record-Route: Record-Route: From: "+39020000000" ;tag=as302aa103 To: ;tag=N7F68303S0F5Q Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 102 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 266 v=0 o=FreeSWITCH 1110445663877587962 6324161778166975444 IN IP4 212.001.001.001 s=FreeSWITCH c=IN IP4 212.001.001.001 t=0 0 m=audio 29688 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ nta: sent 183 Session Progress for INVITE (102) nta: timer set to 60000 ms nua(02EDF6E0): call state changed: received -> early, sent answer soa_get_local_sdp(static::01B75A20, [027FFE48], [027FFE44], [00000000]) called soa_get_params(static::01B75A20, ...) called nua(02EDF6E0): event i_state 183 Session Progress tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F4BC48 from (udp/192.168.0.74:5060) has 316 bytes, veclen = 1 recv 316 bytes from udp/[192.168.0.52]:63214 at 15:10:08.325933: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.74;rport=5060;branch=z9hG4bKNp280jv1Zr9Fj To: From: "+39020000000" ;tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 109984168 INVITE Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F4BC48 (316 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received 100 Trying for INVITE (109984168) nta: 100 Trying is going to a transaction nta_outgoing: RTT is 109.343 ms tport(01AE9650): 02FC1940 by 02F30CF8 with 02F4BC48 (preliminary) tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F4BC48 from (udp/192.168.0.74:5060) has 442 bytes, veclen = 1 recv 442 bytes from udp/[192.168.0.52]:63214 at 15:10:08.638343: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.74;rport=5060;branch=z9hG4bKNp280jv1Zr9Fj Contact: To: ;tag=8b08c857 From: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 109984168 INVITE User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F4BC48 (442 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received 180 Ringing for INVITE (109984168) nta: 180 Ringing is going to a transaction tport(01AE9650): 02FC1940 by 02F30CF8 with 02F4BC48 (preliminary) nua(02EDF468): event r_invite 180 Ringing nua(02EDF468): call state changed: calling -> proceeding nua(02EDF468): event i_state 180 Ringing 2009-01-17 16:10:08 [NOTICE] sofia.c:2627 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5! nta: timer set next to 21113 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F51A68 from (udp/192.168.0.74:5060) has 740 bytes, veclen = 1 recv 740 bytes from udp/[192.168.0.52]:63214 at 15:10:12.059233: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.74;rport=5060;branch=z9hG4bKNp280jv1Zr9Fj Contact: To: ;tag=8b08c857 From: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 109984168 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 187 v=0 o=- 6 2 IN IP4 192.168.0.52 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.52 t=0 0 m=audio 28768 RTP/AVP 3 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ tport(01AE9650): msg 02F51A68 (740 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received 200 OK for INVITE (109984168) nta: 200 OK is going to a transaction tport(01AE9650): 02FC1940 by 02F30CF8 with 02F51A68 soa_set_remote_sdp(static::01D7FE78, 00000000, 02FD0601, 187) called soa_process_answer(static::01D7FE78) called soa_static_offer_answer_action(01D7FE78, soa_process_answer): called soa_sdp_mode_set(02FC27A0, 03007728, ""): called soa_static(01D7FE78, soa_process_answer): upgrade codecs with remote description soa_static(01D7FE78, soa_process_answer): storing local description soa_activate(static::01D7FE78, (nil)) called nua(02EDF468): INVITE: processed SDP answer in 200 OK nua(02EDF468): event r_invite 200 OK soa_activate(static::01D7FE78, (nil)) called nta: selecting scheme sip tport_tsend(01AE9650) tpn = */192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name */192.168.0.52:63214 tport_vsend(01AE9650): 431 bytes of 431 to udp/192.168.0.52:63214 tport_vsend returned 431 send 431 bytes to udp/[192.168.0.52]:63214 at 15:10:12.059233: ------------------------------------------------------------------------ ACK sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.74;rport;branch=z9hG4bKpZU12DD5v1Z2D Max-Forwards: 70 From: "+39020000000" ;tag=r2UgeNKeHUjXa To: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 109984168 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ nta: sent ACK (109984168) to */192.168.0.52:63214 nua(02EDF468): call state changed: proceeding -> ready, received answer soa_get_remote_sdp(static::01D7FE78, [028FFC2C], [028FFC28], [00000000]) called soa_get_params(static::01D7FE78, ...) called nua(02EDF468): event i_state 200 OK nua(02EDF468): event i_active 200 Call active nua(02EDF6E0): recv signal r_respond 200 OK soa_set_params(static::01B75A20, ...) called soa_set_user_sdp(static::01B75A20, 00000000, 0127FA5D, -1) called soa_get_local_sdp(static::01B75A20, [00000000], [027FFE0C], [027FFE08]) called tport_tsend(00DD8D20) tpn = UDP/212.97.59.76:5060 2009-01-17 16:10:12 [NOTICE] sofia.c:3065 sofia_handle_sip_i_state() Channel [sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5] has been answered tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name UDP/212.97.59.76:5060 tport_vsend(00DD8D20): 1144 bytes of 1144 to udp/212.97.59.76:5060 tport_vsend returned 1144 send 1144 bytes to udp/[212.97.59.76]:5060 at 15:10:12.074853: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKe61f.f220cd02.0 Via: SIP/2.0/UDP 212.97.59.87:5060;branch=z9hG4bK57267a99;rport=5060 Record-Route: Record-Route: From: "+39020000000" ;tag=as302aa103 To: ;tag=N7F68303S0F5Q Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 102 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 266 v=0 o=FreeSWITCH 1110445663877587962 6324161778166975444 IN IP4 212.001.001.001 s=FreeSWITCH c=IN IP4 212.001.001.001 t=0 0 m=audio 29688 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (102) nta: timer shortened to 500 ms nua(02EDF6E0): call state changed: early -> completed, sent answer soa_get_local_sdp(static::01B75A20, [027FFE48], [027FFE44], [00000000]) called soa_get_params(static::01B75A20, ...) called nua(02EDF6E0): event i_state 200 OK nua(02EDF6E0): sent signal r_respond 2009-01-17 16:10:12 [NOTICE] switch_ivr_originate.c:1597 switch_ivr_originate() Channel [sofia/external/+39020000000 at sip.messagenet.it] has been answered tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 02F87F58 from (udp/10.0.3.6:5080) has 628 bytes, veclen = 1 recv 628 bytes from udp/[212.97.59.76]:5060 at 15:10:12.106094: ------------------------------------------------------------------------ ACK sip:mod_sofia at 212.001.001.001:5080;transport=udp SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKe61f.f220cd02.2 Via: SIP/2.0/UDP 212.97.59.87:5060;branch=z9hG4bK0498006e;rport=5060 From: "+39020000000" ;tag=as302aa103 To: ;tag=N7F68303S0F5Q Contact: Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 102 ACK User-Agent: alpha Max-Forwards: 69 Content-Length: 0 ------------------------------------------------------------------------ tport(00DD8D20): msg 02F87F58 (628 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received ACK sip:mod_sofia at 212.001.001.001:5080;transport=udp SIP/2.0 (CSeq 102) nta: ACK (102) is going to INVITE (102) soa_clear_remote_sdp(static::01B75A20) called nua(02EDF6E0): event i_ack 200 OK nua(02EDF6E0): call state changed: completed -> ready nua(02EDF6E0): event i_state 200 OK nua(02EDF6E0): event i_active 200 Call active tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F87F58 from (udp/192.168.0.74:5060) has 574 bytes, veclen = 1 recv 574 bytes from udp/[192.168.0.38]:51091 at 15:10:12.527848: ------------------------------------------------------------------------ SUBSCRIBE sip:1002 at 192.168.0.74 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.38:51091;branch=z9hG4bK-d8754z-08453f6c6e106020-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Alessandro Ceccarelli" From: "Alessandro Ceccarelli";tag=333d701e Call-ID: NTU2NDA2N2VjMzZlZTY4ODJlYmUwN2Y1MDc3NmFiOGY. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Event: message-summary Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F87F58 (574 bytes) from udp/192.168.0.38:5060/sip next=00000000 nta: received SUBSCRIBE sip:1002 at 192.168.0.74 SIP/2.0 (CSeq 1) nta: canonizing sip:1002 at 192.168.0.74 with contact nta: SUBSCRIBE (1) going to a default leg soa_clone(static::01AE57C0, 01AE28C0, 02F49AA0) called soa_set_params(static::02F88ED0, ...) called nta_leg_tcreate(02F039B0) nua(02F49AA0): adding notify usage with event message-summary nua(02F49AA0): event i_subscribe 100 Trying nua(): refresh notify after 600 seconds (in [600..600]) nua(): refresh notify after 600 seconds nua(02F49AA0): sent signal r_respond nua(02F49AA0): recv signal r_respond 202 Accepted soa_set_params(static::02F88ED0, ...) called tport_tsend(01AE9650) tpn = UDP/192.168.0.38:51091 tport_resolve addrinfo = 192.168.0.38:51091 tport(01AE9650): not found by name UDP/192.168.0.38:51091 tport_vsend(01AE9650): 813 bytes of 813 to udp/192.168.0.38:51091 tport_vsend returned 813 send 813 bytes to udp/[192.168.0.38]:51091 at 15:10:12.527848: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.0.38:51091;branch=z9hG4bK-d8754z-08453f6c6e106020-1---d8754z-;rport=51091 From: "Alessandro Ceccarelli";tag=333d701e To: "Alessandro Ceccarelli" ;tag=SBN9Fg4He48Fp Call-ID: NTU2NDA2N2VjMzZlZTY4ODJlYmUwN2Y1MDc3NmFiOGY. CSeq: 1 SUBSCRIBE Contact: "Alessandro Ceccarelli" Expires: 300 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=300 Content-Length: 0 ------------------------------------------------------------------------ nta: sent 202 Accepted for SUBSCRIBE (1) nta: timer set next to 4453 ms nta: timer I fired, terminate 200 response incoming_reclaim_all(00000000, 00000000, 027FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01D71768 from (udp/192.168.0.74:5060) has 801 bytes, veclen = 1 recv 801 bytes from udp/[192.168.0.52]:63214 at 15:10:19.104078: ------------------------------------------------------------------------ INVITE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4e76a82c441aae66-1---d8754z-;rport Max-Forwards: 70 Contact: To: "+39020000000";tag=r2UgeNKeHUjXa From: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 182 v=0 o=- 6 3 IN IP4 192.168.0.52 s=CounterPath X-Lite 3.0 c=IN IP4 0.0.0.0 t=0 0 m=audio 28768 RTP/AVP 3 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendonly ------------------------------------------------------------------------ tport(01AE9650): msg 01D71768 (801 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received INVITE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 (CSeq 2) nta: canonizing sip:mod_sofia at 192.168.0.74:5060 with contact nta: INVITE (2) going to existing leg nta: timer shortened to 200 ms soa_init_offer_answer(static::01D7FE78) called soa_set_remote_sdp(static::01D7FE78, 00000000, 01279F8B, 182) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 402 bytes of 402 to udp/192.168.0.52:63214 tport_vsend returned 402 send 402 bytes to udp/[192.168.0.52]:63214 at 15:10:19.119699: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4e76a82c441aae66-1---d8754z-;rport=63214 From: ;tag=8b08c857 To: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (2) nua(02EDF468): event i_invite 100 Trying nua(02EDF468): ready call updated: received received offer soa_get_remote_sdp(static::01D7FE78, [028FFC14], [028FFC10], [00000000]) called nua(02EDF468): event i_state 100 Trying nta: timer set next to 10538 ms nua(02EDF468): sent signal r_respond nua(02EDF468): recv signal r_respond 200 OK soa_set_params(static::01D7FE78, ...) called soa_set_user_sdp(static::01D7FE78, 00000000, 01B8170C, -1) called soa_generate_answer(static::01D7FE78) called soa_static_offer_answer_action(01D7FE78, soa_generate_answer): called soa_static(01D7FE78, soa_generate_answer): upgrade with remote description soa_static(01D7FE78, soa_generate_answer): marking rejected media soa_sdp_mode_set(028FDCF8, 01D705C8, ""): called soa_sdp_mode_set(028FDCF8, 01D705C8, ""): called soa_static(01D7FE78, soa_generate_answer): storing local description soa_activate(static::01D7FE78, (nil)) called soa_get_local_sdp(static::01D7FE78, [00000000], [028FFE0C], [028FFE08]) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 984 bytes of 984 to udp/192.168.0.52:63214 tport_vsend returned 984 send 984 bytes to udp/[192.168.0.52]:63214 at 15:10:19.369627: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4e76a82c441aae66-1---d8754z-;rport=63214 From: ;tag=8b08c857 To: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 276 v=0 o=FreeSWITCH 6699757194367626025 7696645903169343293 IN IP4 192.168.0.74 s=FreeSWITCH c=IN IP4 192.168.0.74 t=0 0 m=audio 29688 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=recvonly a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (2) nta: timer shortened to 500 ms nua(02EDF468): ready call updated: completed sent answer soa_get_local_sdp(static::01D7FE78, [028FFE48], [028FFE44], [00000000]) called soa_get_params(static::01D7FE78, ...) called nua(02EDF468): event i_state 200 OK tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F4BC48 from (udp/192.168.0.74:5060) has 497 bytes, veclen = 1 recv 497 bytes from udp/[192.168.0.52]:63214 at 15:10:19.478970: ------------------------------------------------------------------------ ACK sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-0a230c1b27666c30-1---d8754z-;rport Max-Forwards: 70 Contact: To: "+39020000000";tag=r2UgeNKeHUjXa From: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 2 ACK User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F4BC48 (497 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received ACK sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 (CSeq 2) nta: ACK (2) is going to INVITE (2) soa_clear_remote_sdp(static::01D7FE78) called nua(02EDF468): event i_ack 200 OK nua(02EDF468): ready call updated: ready nua(02EDF468): event i_state 200 OK nua(02EDF468): event i_active 200 Call active nta: timer set next to 4594 ms nta: timer I fired, terminate 200 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/5 term, 1/5 free nta: timer set next to 5352 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01D71768 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 01D71768 (4 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: timer J fired, terminate 401 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/4 term, 1/4 free nta: timer set next to 197 ms nta: timer J fired, terminate 200 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free nta: timer set next to 87 ms nta: timer J fired, terminate 202 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 13911 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01D71768 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 01D71768 (4 bytes) from udp/192.168.0.38:5060/sip next=00000000 tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 01D71768 from (udp/192.168.0.74:5060) has 802 bytes, veclen = 1 recv 802 bytes from udp/[192.168.0.52]:63214 at 15:10:39.504451: ------------------------------------------------------------------------ INVITE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4b5f9a669369bb1f-1---d8754z-;rport Max-Forwards: 70 Contact: To: "+39020000000";tag=r2UgeNKeHUjXa From: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 3 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 183 v=0 o=- 6 4 IN IP4 192.168.0.52 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.52 t=0 0 m=audio 28768 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ tport(01AE9650): msg 01D71768 (802 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received INVITE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 (CSeq 3) nta: canonizing sip:mod_sofia at 192.168.0.74:5060 with contact nta: INVITE (3) going to existing leg nta: timer shortened to 200 ms soa_init_offer_answer(static::01D7FE78) called soa_set_remote_sdp(static::01D7FE78, 00000000, 01279F8B, 183) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 402 bytes of 402 to udp/192.168.0.52:63214 tport_vsend returned 402 send 402 bytes to udp/[192.168.0.52]:63214 at 15:10:39.504451: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4b5f9a669369bb1f-1---d8754z-;rport=63214 From: ;tag=8b08c857 To: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 3 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (3) nua(02EDF468): event i_invite 100 Trying nua(02EDF468): ready call updated: received received offer soa_get_remote_sdp(static::01D7FE78, [028FFC14], [028FFC10], [00000000]) called nua(02EDF468): event i_state 100 Trying nta: timer set next to 4351 ms nua(02EDF468): sent signal r_respond nua(02EDF468): recv signal r_respond 200 OK soa_set_params(static::01D7FE78, ...) called soa_set_user_sdp(static::01D7FE78, 00000000, 01B8170C, -1) called soa_generate_answer(static::01D7FE78) called soa_static_offer_answer_action(01D7FE78, soa_generate_answer): called soa_static(01D7FE78, soa_generate_answer): upgrade with remote description soa_sdp_mode_set(028FDCF8, 00DE0118, ""): called soa_static(01D7FE78, soa_generate_answer): storing local description soa_activate(static::01D7FE78, (nil)) called soa_get_local_sdp(static::01D7FE78, [00000000], [028FFE0C], [028FFE08]) called tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 972 bytes of 972 to udp/192.168.0.52:63214 tport_vsend returned 972 send 972 bytes to udp/[192.168.0.52]:63214 at 15:10:39.785620: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-4b5f9a669369bb1f-1---d8754z-;rport=63214 From: ;tag=8b08c857 To: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 3 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 264 v=0 o=FreeSWITCH 6699757194367626025 7696645903169343294 IN IP4 192.168.0.74 s=FreeSWITCH c=IN IP4 192.168.0.74 t=0 0 m=audio 29688 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (3) nta: timer shortened to 500 ms nua(02EDF468): ready call updated: completed sent answer soa_get_local_sdp(static::01D7FE78, [028FFE48], [028FFE44], [00000000]) called soa_get_params(static::01D7FE78, ...) called nua(02EDF468): event i_state 200 OK tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F04E20 from (udp/192.168.0.74:5060) has 497 bytes, veclen = 1 recv 497 bytes from udp/[192.168.0.52]:63214 at 15:10:39.894964: ------------------------------------------------------------------------ ACK sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-b131c97e6c16e653-1---d8754z-;rport Max-Forwards: 70 Contact: To: "+39020000000";tag=r2UgeNKeHUjXa From: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 3 ACK User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F04E20 (497 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received ACK sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 (CSeq 3) nta: ACK (3) is going to INVITE (3) soa_clear_remote_sdp(static::01D7FE78) called nua(02EDF468): event i_ack 200 OK nua(02EDF468): ready call updated: ready nua(02EDF468): event i_state 200 OK nua(02EDF468): event i_active 200 Call active nta: timer set next to 3758 ms tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F04E20 from (udp/192.168.0.74:5060) has 537 bytes, veclen = 1 recv 537 bytes from udp/[192.168.0.52]:63214 at 15:10:42.316141: ------------------------------------------------------------------------ BYE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-5f47b655ea7fe925-1---d8754z-;rport Max-Forwards: 70 Contact: To: "+39020000000";tag=r2UgeNKeHUjXa From: ;tag=8b08c857 Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 4 BYE User-Agent: X-Lite release 1100l stamp 47546 Reason: SIP;description="User Hung Up" Content-Length: 0 ------------------------------------------------------------------------ tport(01AE9650): msg 02F04E20 (537 bytes) from udp/192.168.0.52:5060/sip next=00000000 nta: received BYE sip:mod_sofia at 192.168.0.74:5060 SIP/2.0 (CSeq 4) nta: canonizing sip:mod_sofia at 192.168.0.74:5060 with contact nta: BYE (4) going to existing leg tport_tsend(01AE9650) tpn = UDP/192.168.0.52:63214 tport_resolve addrinfo = 192.168.0.52:63214 tport(01AE9650): not found by name UDP/192.168.0.52:63214 tport_vsend(01AE9650): 560 bytes of 560 to udp/192.168.0.52:63214 tport_vsend returned 560 send 560 bytes to udp/[192.168.0.52]:63214 at 15:10:42.316141: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.52:63214;branch=z9hG4bK-d8754z-5f47b655ea7fe925-1---d8754z-;rport=63214 From: ;tag=8b08c857 To: "+39020000000";tag=r2UgeNKeHUjXa Call-ID: c4ecd490-5f4b-122c-2780-39a48cb53b8d CSeq: 4 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ nta: sent 200 OK for BYE (4) nua(02EDF468): event i_bye 200 Session Terminated nua(02EDF468): removing session usage nua(02EDF468): call state changed: ready -> terminated nua(02EDF468): event i_state 200 Session Terminated nua(02EDF468): event i_terminated 200 Session Terminated soa_destroy(static::01D7FE78) called nta_leg_destroy(02F47280) 2009-01-17 16:10:42 [NOTICE] sofia.c:3124 sofia_handle_sip_i_state() Hangup sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] nua(02EDF468): sent signal r_destroy nua(02EDF468): event i_terminated dropped nua(02EDF468): recv signal r_destroy nta_leg_destroy(00000000) 2009-01-17 16:10:42 [NOTICE] switch_ivr_bridge.c:955 switch_ivr_multi_threaded_bridge() Hangup sofia/external/+39020000000 at sip.messagenet.it [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-17 16:10:42 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 2 (sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5) Ended 2009-01-17 16:10:42 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/sip:1008 at 192.168.0.52:63214;rinstance=a8691dfbc11775d5 [CS_HANGUP] nua(02EDF6E0): recv signal r_bye soa_set_params(static::01B75A20, ...) called soa_terminate(static::01B75A20) called soa_init_offer_answer(static::01B75A20) called nta: selecting scheme sip tport_tsend(00DD8D20) tpn = */212.97.59.76:5060 nua(02EDF6E0): sent signal r_bye tport_resolve addrinfo = 212.97.59.76:5060 tport(00DD8D20): not found by name */212.97.59.76:5060 tport_vsend(00DD8D20): 796 bytes of 796 to udp/212.97.59.76:5060 tport_vsend returned 796 send 796 bytes to udp/[212.97.59.76]:5060 at 15:10:42.331762: ------------------------------------------------------------------------ BYE sip:+39020000000 at 212.97.59.87 SIP/2.0 Via: SIP/2.0/UDP 212.001.001.001:5080;rport;branch=z9hG4bKpZU12DD5v1Z2D Route: Route: Max-Forwards: 70 From: ;tag=N7F68303S0F5Q To: "+39020000000" ;tag=as302aa103 Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 109984185 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ nta: sent BYE (109984185) to */212.97.59.76:5060 tport_pend(00DD8D20): pending 02F4B840 for udp/10.0.3.6:5080 (already 0) nta: timer set to 32000 ms nta: timer shortened to 500 ms 2009-01-17 16:10:42 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 1 (sofia/external/+39020000000 at sip.messagenet.it) Ended 2009-01-17 16:10:42 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/external/+39020000000 at sip.messagenet.it [CS_HANGUP] tport_wakeup_pri(00DD8D20): events IN tport_recv_event(00DD8D20) tport(00DD8D20) msg 02F8AAA8 from (udp/10.0.3.6:5080) has 612 bytes, veclen = 1 recv 612 bytes from udp/[212.97.59.76]:5060 at 15:10:42.347382: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 212.001.001.001:5080;rport=5080;branch=z9hG4bKpZU12DD5v1Z2D Record-Route: Record-Route: From: ;tag=N7F68303S0F5Q To: "+39020000000" ;tag=as302aa103 Call-ID: 0991575e53c716557b287c0f4e34a842 at sip.messagenet.it CSeq: 109984185 BYE User-Agent: alpha Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 ------------------------------------------------------------------------ tport(00DD8D20): msg 02F8AAA8 (612 bytes) from udp/212.97.59.76:5080/sip next=00000000 nta: received 200 OK for BYE (109984185) nta: 200 OK is going to a transaction nta_outgoing: RTT is 31.241 ms tport(00DD8D20): 02F4B840 by 01B70810 with 02F8AAA8 nua(02EDF6E0): event r_bye 200 OK nua(02EDF6E0): call state changed: terminating -> terminated nua(02EDF6E0): event i_state 200 to BYE nua(02EDF6E0): event i_terminated 200 to BYE nua(02EDF6E0): removing session usage soa_destroy(static::01B75A20) called nta_leg_destroy(02F004E0) nua: terminated session 02EDF6E0 nua(02EDF6E0): recv signal r_destroy nta_leg_destroy(00000000) nua(02EDF6E0): sent signal r_destroy nta: timer set next to 4500 ms nta: timer D fired, terminate INVITE (109984168) nta: timer F fired, terminating ACK (109984168) outgoing_reclaim_all(00000000, 00000000, 028FFEC0) nta_outgoing_timer: 0/0 resent, 1/1 tout, 1/1 term, 2/2 free nta: timer set next to 477 ms nta: timer J fired, terminate 202 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free nta: timer set next to 345 ms nta: timer I fired, terminate 200 response incoming_reclaim_all(00000000, 00000000, 028FFEBC) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 29407 ms nta: timer K fired, terminate BYE (109984185) outgoing_reclaim_all(00000000, 00000000, 027FFEC0) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set 2009-01-17 16:10:57 [CONSOLE] switch_console.c:229 switch_console_process() Bye! 2009-01-17 16:10:57 [CONSOLE] switch_core.c:1459 switch_core_destroy() End existing sessions 2009-01-17 16:10:57 [CONSOLE] switch_core.c:1461 switch_core_destroy() Clean up modules. 2009-01-17 16:10:57 [CONSOLE] switch_core_memory.c:443 switch_core_memory_stop() Stopping memory pool queue. 2009-01-17 16:10:57 [CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim() Returning 3 recycled memory pool(s) 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'lua' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'luarun' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'lua' 2009-01-17 16:10:57 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_lua 2009-01-17 16:10:57 [NOTICE] switch_event.c:1189 switch_event_unbind() Event Binding deleted for mod_lua:MESSAGE_QUERY 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'limit' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'limit_hash' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'db' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'hash' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'group' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'limit_hash_usage' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'limit_usage' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'db' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'hash' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'group' 2009-01-17 16:10:57 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_limit has no shutdown routine 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'fifo' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'fifo' 2009-01-17 16:10:57 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'fifo_member' 2009-01-17 16:10:57 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_fifo 2009-01-17 16:10:57 [NOTICE] switch_event.c:1189 switch_event_unbind() Event Binding deleted for mod_fifo:PRESENCE_PROBE 2009-01-17 16:10:57 [NOTICE] switch_event.c:374 switch_event_free_subclass_detailed() Subclass reservation deleted for .\mod_fifo.c:fifo::info tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F053A0 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 02F053A0 (4 bytes) from udp/192.168.0.52:5060/sip next=00000000 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'expr' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_expr has no shutdown routine 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'aiff' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'au' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'avr' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'caf' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'htk' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'iff' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'mat' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'mat' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'paf' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'pvf' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'raw' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'sd2' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'sds' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'sf' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'voc' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'w64' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'wav' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'wav' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'wav' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'xi' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'r8' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'r16' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'r24' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'r32' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'gsm' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'ul' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:575 switch_loadable_module_unprocess() Deleting File Format 'al' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_sndfile 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:700 switch_loadable_module_unprocess() Deleting Say interface 'en' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_say_en has no shutdown routine 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:434 switch_loadable_module_unprocess() Deleting Endpoint 'loopback' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_loopback 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'conference' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'conference_set_auto_outcall' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'conference' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:677 switch_loadable_module_unprocess() Deleting Chat interface 'conf' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_conference 2009-01-17 16:10:58 [NOTICE] switch_event.c:1189 switch_event_unbind() Event Binding deleted for mod_conference:PRESENCE_PROBE 2009-01-17 16:10:58 [NOTICE] switch_event.c:374 switch_event_free_subclass_detailed() Subclass reservation deleted for .\mod_conference.c:conference::maintenance 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PROXY-VID' (PROXY VIDEO PASS-THROUGH) 90000hz 0ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PROXY' (PROXY PASS-THROUGH) 8000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 11025hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 22050hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 8ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 6ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 4ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 2ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 8ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 6ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 4ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 2ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 8ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 6ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 4ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 2ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 8ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 6ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 4ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 48000hz 2ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 30ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 50ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 32000hz 60ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 30ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 50ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 16000hz 60ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 30ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 50ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 60ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 70ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 80ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 90ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 100ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 110ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'L16' (RAW Signed Linear (16 bit)) 8000hz 120ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 30ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 50ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 60ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 70ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 80ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 90ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 100ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 110ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMU' (G.711 ulaw) 8000hz 120ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 10ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 20ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 30ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 40ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 50ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 60ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 70ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 80ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 90ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 100ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 110ms 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'PCMA' (G.711 alaw) 8000hz 120ms 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: CORE_PCM_MODULE 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'socket' 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'event_sink' 2009-01-17 16:10:58 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_event_socket 2009-01-17 16:10:58 [NOTICE] switch_event.c:1189 switch_event_unbind() Event Binding deleted for mod_event_socket:ALL 2009-01-17 16:10:58 [NOTICE] mod_event_socket.c:2079 mod_event_socket_runtime() Shutting Down 2009-01-17 16:10:58 [NOTICE] switch_loadable_module.c:95 switch_loadable_module_exec() Thread ended for mod_event_socket 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'AMR' (AMR) 8000hz 20ms 2009-01-17 16:10:59 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_amr has no shutdown routine 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:434 switch_loadable_module_unprocess() Deleting Endpoint 'error' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:434 switch_loadable_module_unprocess() Deleting Endpoint 'group' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:434 switch_loadable_module_unprocess() Deleting Endpoint 'user' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:479 switch_loadable_module_unprocess() Deleting Dialplan 'inline' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'privacy' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'flush_dtmf' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'hold' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'unhold' 2009-01-17 16:10:59 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'transfer' tport_wakeup_pri(01AE9650): events IN tport_recv_event(01AE9650) tport(01AE9650) msg 02F053A0 from (udp/192.168.0.74:5060) has 4 bytes, veclen = 1 tport(01AE9650): bad msg 02F053A0 (4 bytes) from udp/192.168.0.38:5060/sip next=00000000 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'check_acl' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'verbose_events' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'sleep' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'delay_echo' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'strftime' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'phrase' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'eval' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'pre_answer' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'answer' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'hangup' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'set_name' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'presence' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'log' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'info' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'event' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'export' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'set' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'set_global' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'set_profile_var' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'unset' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'ring_ready' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'remove_bugs' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'break' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'detect_speech' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'ivr' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'redirect' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'send_display' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'respond' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'deflect' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'queue_dtmf' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'send_dtmf' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'sched_hangup' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'sched_broadcast' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'sched_transfer' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'execute_extension' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'mkdir' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'soft_hold' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'bind_meta_app' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'unbind_meta_app' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'intercept' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'eavesdrop' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'three_way' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'set_user' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'stop_dtmf' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'start_dtmf' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'stop_dtmf_generate' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'start_dtmf_generate' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'stop_tone_detect' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'fax_detect' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'tone_detect' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'echo' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'park' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'park_state' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'gentones' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'playback' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'att_xfer' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'read' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'stop_record_session' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'record_session' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'record' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'stop_displace_session' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'displace_session' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'speak' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'clear_speech_cache' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'bridge' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'system' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'say' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'wait_for_silence' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'strepoch' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'chat' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'strftime' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'presence' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:677 switch_loadable_module_unprocess() Deleting Chat interface 'event' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:677 switch_loadable_module_unprocess() Deleting Chat interface 'api' 2009-01-17 16:11:03 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_dptools has no shutdown routine 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:455 switch_loadable_module_unprocess() Deleting Codec 'G723' (G.723.1 6.3k) 8000hz 30ms 2009-01-17 16:11:03 [CONSOLE] switch_loadable_module.c:1234 do_shutdown() mod_g723_1 has no shutdown routine 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'console' 2009-01-17 16:11:03 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_logfile 2009-01-17 16:11:03 [NOTICE] switch_event.c:1189 switch_event_unbind() Event Binding deleted for mod_logfile:TRAP 2009-01-17 16:11:03 [INFO] mod_logfile.c:391 mod_logfile_shutdown() Closing C:\Program Files\FreeSwitch\log\freeswitch.log 2009-01-17 16:11:03 [DEBUG] switch_loadable_module.c:427 switch_loadable_module_unprocess() Write lock interface 'sofia' to wait for existing references. 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:434 switch_loadable_module_unprocess() Deleting Endpoint 'sofia' 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'sofia' 2009-01-17 16:11:03 [DEBUG] switch_loadable_module.c:538 switch_loadable_module_unprocess() Write lock interface 'sofia' to wait for existing references. 2009-01-17 16:11:03 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'sofia_contact' 2009-01-17 16:11:03 [DEBUG] switch_loadable_module.c:538 switch_loadable_module_unprocess() Write lock interface 'sofia_contact' to wait for existing references. 2009-01-17 16:11:03 [DEBUG] switch_loadable_module.c:669 switch_loadable_module_unprocess() Write lock interface 'sip' to wait for existing referen -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.log Type: application/octet-stream Size: 148830 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090118/b9bfdfeb/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: tport_sip.log Type: application/octet-stream Size: 21119 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090118/b9bfdfeb/attachment-0003.obj From brian at freeswitch.org Sun Jan 18 18:33:56 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 18 Jan 2009 20:33:56 -0600 Subject: [Freeswitch-users] Dialing Out Problem via Gateway In-Reply-To: <94c121cc0901170802i18c883cdv104660fc8f132315@mail.gmail.com> References: <94c121cc0901170802i18c883cdv104660fc8f132315@mail.gmail.com> Message-ID: You modified the Local_Extension in default.xml 2009-01-17 16:10:08 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [Local_Extension] destination_number(1008) =~ /^+39020000000$/ 2009-01-17 16:10:08 [ERR] switch_regex.c:94 switch_regex_perform() COMPILE ERROR: 1 [nothing to repeat][^+39020000000$] 2009-01-17 16:10:08 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch Remove that match for the /^+39020000000$/ and put it back to the default value... it should work fine. /b On Jan 17, 2009, at 10:02 AM, Alberto Ceccarelli wrote: > From helmut.kuper at ewetel.de Mon Jan 19 00:42:32 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 19 Jan 2009 09:42:32 +0100 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> Message-ID: <49743CF8.6050804@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Michael, I'm currently on my way to put all those shell and perl stuff into C code. As soon as that works I try to put it in ozmod_isdn. So after that FS will be able to enable Q931Pcapfile generation. After that I will send the patch to FS. Hope this is ok for you. What do you mean with "help with jira"? I thought it is just a bug tracker... If it's also a feature request tool, I will feed it with documentation about what I did. regards helmut Am 17.01.2009 21:44, schrieb Michael S Collins: > Guys this is awesome! Helmut, if you need any help with jira just let > me know. > -MC > > Sent from my iPhone > > On Jan 17, 2009, at 9:20 AM, Brian West wrote: > >> Maybe open a jira with this info? Maybe it can all be done as a one >> step process in the ozmod_isdn ;) >> >> /b >> >> On Jan 17, 2009, at 10:58 AM, Helmut Kuper wrote: >> >>> Hi Brian, >>> >>> currently it's a simple perl script which greps all Q931 hexdumps >>> from >>> FS logfile converting them to a TPKT packet, and writing those to a >>> separate local file (wireshark's text2pcap file format). This 1st >>> step >>> is easy to put right into FS ozmod_isdn debug code. >>> The 2nd step is to convert the new file into a .pcap file with adding >>> TCP/IP dummy packet in front of each TPKT packet. This is done via >>> test2pcap. This .pcap file is ready to be decoded by wireshark. >>> >>> I put both steps into a little shell script and added a 3rd step to >>> get >>> those .pcap file emailed from the Server to my Desktop. >>> >>> regards >>> helmut >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl0PPgACgkQ4tZeNddg3dwKMACfYYHglospD7FJeY4Ne2Q8qBWJ pNIAoKtEXQ+RnVg2ahySjd5zKfUcOowQ =1abg -----END PGP SIGNATURE----- From sias at cpdata.co.za Mon Jan 19 01:13:13 2009 From: sias at cpdata.co.za (Sias Mey) Date: Mon, 19 Jan 2009 11:13:13 +0200 Subject: [Freeswitch-users] Dialling from a conferance Message-ID: <20090119091313.GA7773@cpdata.co.za> Hi, I am building a call handleing framework that uses freeswitch for CRM like activities. To this end I an actually trying to use conferances for meeting points between diffirent parties. I would like to be able to add/drop poeple from the calls at will and with great flexibility (hence freeswitch). I have only one question whith implemeting all this (and that is a huge up to the FS dev for keeping the API constant accross seperate interfaces) Is it possible to play ringing into the conferance while using the conferance dial command to make a new call? I was thinking that I would like the one party to be ready and waiting in the conferance and then dial out from there. However by default it seems that there would be almost no feedback to the party in the conferance about what is happening. It has been a while since I last tested this, but I am busy doing an svn up in the background as I type this (South African bandwidth makes it take a while) and I havent seen anything that could be related in any of the documentation I have read. Thanks in advance. From juanbackson at gmail.com Mon Jan 19 02:23:23 2009 From: juanbackson at gmail.com (Juan Backson) Date: Mon, 19 Jan 2009 18:23:23 +0800 Subject: [Freeswitch-users] strange 503 error thrown by freeswitch Message-ID: <27c25bc40901190223q5de210a2j6ed83bb2c7eacb27@mail.gmail.com> Hi, I am running a test for a set of 290 calls to be fired to freeswitch once every second. The console logs show the following for one of those 290 calls. Could someone please help me out to fix this problem? What could be causing this? I recalled being able to run thousands of calls without any error with an older release. 2009-01-19 13:07:19 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file 2009-01-19 13:07:19 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10286 at 192.168.1.122:7001Execute hangup() 2009-01-19 13:07:19 [NOTICE] mod_dptools.c:566 hangup_function() Hangup sofia/internal/10286 at 192.168.1.122:7001 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-19 13:07:19 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/internal/ 10286 at 192.168.1.122:7001 [KILL] 2009-01-19 13:07:19 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/ 10286 at 192.168.1.122:7001 [BREAK] 2009-01-19 13:07:19 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/internal/10286 at 192.168.1.122:7001) State EXECUTE going to sleep 2009-01-19 13:07:19 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/internal/10286 at 192.168.1.122:7001) Running State Change CS_HANGUP 2009-01-19 13:07:19 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/internal/10286 at 192.168.1.122:7001) State HANGUP 2009-01-19 13:07:19 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/internal/10286 at 192.168.1.122:7001 Overriding SIP cause 480 with 503 from the other leg 2009-01-19 13:07:19 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/10286 at 192.168.1.122:7001 hanging up, cause: NORMAL_CLEARING 2009-01-19 13:07:19 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 503 2009-01-19 13:07:20 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/10286 at 192.168.1.122:7001Standard HANGUP, cause: NORMAL_CLEARING 2009-01-19 13:07:20 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/internal/10286 at 192.168.1.122:7001) State HANGUP going to sleep 2009-01-19 13:07:20 [DEBUG] switch_core_session.c:938 switch_core_session_thread() Session 575 (sofia/internal/ 10286 at 192.168.1.122:7001) Locked, Waiting on external entities 2009-01-19 13:07:20 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 575 (sofia/internal/ 10286 at 192.168.1.122:7001) Ended 2009-01-19 13:07:20 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/internal/ 10286 at 192.168.1.122:7001 [CS_HANGUP] freeswitch returns 503 U 192.168.1.116:5070 -> 192.168.1.122:5060 SIP/2.0 503 Service Unavailable. Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bK27f5.22865cc6.0. Via: SIP/2.0/UDP 192.168.1.103:7001. From: 10286 ;tag=287. To: 0010286 >;tag=pSHaHm69m41Xa. Call-ID: 287-23175 at 192.168.1.103. CSeq: 2 INVITE. User-Agent: YHT Media Gateway. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/5d132d3d/attachment.html From regs at kinetix.gr Mon Jan 19 02:31:01 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Mon, 19 Jan 2009 12:31:01 +0200 Subject: [Freeswitch-users] Outbound call - choose profile In-Reply-To: References: <497082DC.8020904@kinetix.gr> <49709A5B.2060806@freeswitch.org> <49709F7B.4030905@kinetix.gr> <191c3a030901160915n250d791m4451815341b8d60a@mail.gmail.com> <4971F96C.6050406@kinetix.gr> Message-ID: <49745665.5090405@kinetix.gr> Thanks for the clarification Brian West wrote: > The external profile is setup to parse the gateways... internal is > setup to parse the domains and apply an alias to the internal profile > for all domains in the directory. > > On internal you have: > > > > > > > > > > Then on external you have: > > > > > > Notice this tells internal to add an alias for every domain to > internal, while external is told to parse for gateways. > > /b > > On Jan 17, 2009, at 9:29 AM, Apostolos Pantsiopoulos wrote: > > >> Yes, I am aware of that fact : insteaad of the pre-configured >> internal/external profiles I coulf have myprofile1/myprofile2. >> But since I am using the pre-configured profiles (without changing >> the ports) >> and since the internal profile initiated the parsing I would expect >> that all the gateways defined in my directory user's xml files would >> belong to the internal profile (which is preconfigured to use port >> 5060). >> But when I am capturing the packets sent to that gateway I can see >> that >> the port used from FS is 5080 (which is the preconfigured port of >> the internal profile). >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/5de7f0e1/attachment.html From juanbackson at gmail.com Mon Jan 19 03:06:17 2009 From: juanbackson at gmail.com (Juan Backson) Date: Mon, 19 Jan 2009 19:06:17 +0800 Subject: [Freeswitch-users] strange 503 error thrown by freeswitch In-Reply-To: <27c25bc40901190223q5de210a2j6ed83bb2c7eacb27@mail.gmail.com> References: <27c25bc40901190223q5de210a2j6ed83bb2c7eacb27@mail.gmail.com> Message-ID: <27c25bc40901190306l91cb7dbr2ace115dd561646c@mail.gmail.com> Here the detail trace for this problem, please notice the highlighted line. Thanks alot for all your help. 2009-01-19 13:40:30 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/internal/10110 at 192.168.1.122:7001 entering state [ready] recv 787 bytes from udp/[192.168.1.122]:5060 at 18:40:30.915571: ------------------------------------------------------------------------ INVITE sip:0010111 at 192.168.1.116:5070 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bK855f.d0dc6ad1.0 Via: SIP/2.0/UDP 192.168.1.103:7001 From: 10111 ;tag=112 To: 0010111 > Call-ID: 112-26869 at 192.168.1.103 CSeq: 2 INVITE Contact: Max-Forwards: 69 User-Agent: Performance Test Content-Type: application/sdp Content-Length: 276 P-hint: inbound->inbound v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.103 s=- t=0 0 c=IN IP4 192.168.1.103 m=audio 6444 RTP/AVP 0 9 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ send 364 bytes to udp/[192.168.1.122]:5060 at 18:40:30.916041: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bK855f.d0dc6ad1.0 Via: SIP/2.0/UDP 192.168.1.103:7001 Record-Route: From: 10111 ;tag=112 To: 0010111 > Call-ID: 112-26869 at 192.168.1.103 CSeq: 2 INVITE User-Agent: FREESWITCH Media Gateway Content-Length: 0 ------------------------------------------------------------------------ 2009-01-19 13:40:30 [DEBUG] sofia.c:3762 sofia_handle_sip_i_invite() IP 192.168.1.122 Approved by acl "lan[]". Access Granted. 2009-01-19 13:40:30 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/10111 at 192.168.1.122:7001[a61933bc-e658-11dd-b62c-5db88ae44bdf] 2009-01-19 13:40:30 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/internal/10111 at 192.168.1.122:7001 entering state [received] 2009-01-19 13:40:30 [DEBUG] sofia.c:2533 sofia_handle_sip_i_state() Remote SDP: v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.103 s=- c=IN IP4 192.168.1.103 t=0 0 m=audio 6444 RTP/AVP 0 9 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-01-19 13:40:30 [DEBUG] sofia_glue.c:2409 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[G722:9:8000] 2009-01-19 13:40:30 [DEBUG] sofia_glue.c:2409 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2009-01-19 13:40:30 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/internal/10111 at 192.168.1.122:7001 PCMU/8000 20 ms 160 samples 2009-01-19 13:40:30 [DEBUG] sofia_glue.c:2373 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-01-19 13:40:30 [DEBUG] sofia.c:2685 sofia_handle_sip_i_state() (sofia/internal/10111 at 192.168.1.122:7001) State Change CS_NEW -> CS_INIT 2009-01-19 13:40:30 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/ 10111 at 192.168.1.122:7001 [BREAK] 2009-01-19 13:40:30 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) Running State Change CS_INIT 2009-01-19 13:40:30 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) State INIT 2009-01-19 13:40:30 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/ 10111 at 192.168.1.122:7001 SOFIA INIT 2009-01-19 13:40:30 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/ 10111 at 192.168.1.122:7001) State Change CS_INIT -> CS_ROUTING 2009-01-19 13:40:30 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/ 10111 at 192.168.1.122:7001 [BREAK] 2009-01-19 13:40:30 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) State INIT going to sleep 2009-01-19 13:40:30 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) Running State Change CS_ROUTING 2009-01-19 13:40:30 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) State ROUTING 2009-01-19 13:40:30 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/10111 at 192.168.1.122:7001 SOFIA ROUTING 2009-01-19 13:40:30 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_routing() sofia/internal/10111 at 192.168.1.122:7001Standard ROUTING 2009-01-19 13:40:30 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 10111->0010111 in context public 2009-01-19 13:40:31 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML response is in /tmp/a619aae0-e658-11dd-b62c-5db88ae44bdf.tmp.xml 2009-01-19 13:40:31 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [test9] destination_number(0010111) =~ /^(.*)$/ 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:100 switch_core_standard_on_routing() (sofia/internal/10111 at 192.168.1.122:7001) State Change CS_ROUTING -> CS_EXECUTE 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/ 10111 at 192.168.1.122:7001 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) State ROUTING going to sleep 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) Running State Change CS_EXECUTE 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) State EXECUTE 2009-01-19 13:40:31 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/10111 at 192.168.1.122:7001 SOFIA EXECUTE 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:137 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Standard EXECUTE 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute export(hold_music=silence) 2009-01-19 13:40:31 [DEBUG] mod_dptools.c:819 export_function() EXPORT [hold_music]=[silence] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute set(call_timeout=120) 2009-01-19 13:40:31 [DEBUG] mod_dptools.c:681 set_function() sofia/internal/ 10111 at 192.168.1.122:7001 SET [call_timeout]=[120] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute set(hangup_after_bridge=true) 2009-01-19 13:40:31 [DEBUG] mod_dptools.c:681 set_function() sofia/internal/ 10111 at 192.168.1.122:7001 SET [hangup_after_bridge]=[true] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute set(continue_on_fail=true) 2009-01-19 13:40:31 [DEBUG] mod_dptools.c:681 set_function() sofia/internal/ 10111 at 192.168.1.122:7001 SET [continue_on_fail]=[true] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute set(language=zh) 2009-01-19 13:40:31 [DEBUG] mod_dptools.c:681 set_function() sofia/internal/ 10111 at 192.168.1.122:7001 SET [language]=[zh] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute set(ringback=%(2000, 4000, 440.0, 480.0)) 2009-01-19 13:40:31 [DEBUG] mod_dptools.c:681 set_function() sofia/internal/ 10111 at 192.168.1.122:7001 SET [ringback]=[%(2000, 4000, 440.0, 480.0)] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute bind_meta_app(1 a s execute_extension::a_record XML features) 2009-01-19 13:40:31 [INFO] switch_ivr_async.c:1570 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 1 execute_extension::a_record XML features 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute bind_meta_app(2 a s execute_extension::a_stoprecord XML features) 2009-01-19 13:40:31 [INFO] switch_ivr_async.c:1570 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 2 execute_extension::a_stoprecord XML features 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute bind_meta_app(3 a s execute_extension::a_att_xfer XML features) 2009-01-19 13:40:31 [INFO] switch_ivr_async.c:1570 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 3 execute_extension::a_att_xfer XML features 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute bind_meta_app(1 b s execute_extension::b_record XML features) 2009-01-19 13:40:31 [INFO] switch_ivr_async.c:1577 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::b_record XML features 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute bind_meta_app(2 b s execute_extension::b_stoprecord XML features) 2009-01-19 13:40:31 [INFO] switch_ivr_async.c:1577 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 execute_extension::b_stoprecord XML features 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute bind_meta_app(3 b s execute_extension::b_att_xfer XML features) 2009-01-19 13:40:31 [INFO] switch_ivr_async.c:1577 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::b_att_xfer XML features 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/10111 at 192.168.1.122:7001Execute bridge(sofia/gateway/openser/10111) 2009-01-19 13:40:31 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/10111 [a6274970-e658-11dd-b62c-5db88ae44bdf] 2009-01-19 13:40:31 [DEBUG] mod_sofia.c:2495 sofia_outgoing_channel() (sofia/internal/10111) State Change CS_NEW -> CS_INIT 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/10111 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/internal/10111) Running State Change CS_INIT 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/internal/10111) State INIT 2009-01-19 13:40:31 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/10111 SOFIA INIT 2009-01-19 13:40:31 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/10111) State Change CS_INIT -> CS_ROUTING 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/10111 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/internal/10111) State INIT going to sleep 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/internal/10111) Running State Change CS_ROUTING 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/internal/10111) State ROUTING 2009-01-19 13:40:31 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/10111 SOFIA ROUTING 2009-01-19 13:40:31 [DEBUG] switch_ivr_originate.c:52 originate_on_routing() (sofia/internal/10111) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/10111 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/internal/10111) State ROUTING going to sleep 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/internal/10111) Running State Change CS_CONSUME_MEDIA 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/internal/10111) State CONSUME_MEDIA 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/internal/10111) State CONSUME_MEDIA going to sleep send 1229 bytes to udp/[192.168.1.122]:5060 at 18:40:31.010618: ------------------------------------------------------------------------ INVITE sip:10111 at 192.168.1.122:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:5070;rport;branch=z9hG4bKDvK1NF3F0K70m Max-Forwards: 68 From: "10111" ;tag=3cZNea7Be27cB To: Call-ID: 7d85cf3d-60fb-122c-f480-001517871e28 CSeq: 110076879 INVITE Contact: User-Agent: FREESWITCH Media Gateway Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 335 P-hint: inbound->inbound Remote-Party-ID: "10111" ;screen=yes;privacy=off v=0 o=FreeSWITCH 5124756122584252723 8222947971594082779 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12396 RTP/AVP 0 9 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2009-01-19 13:40:31 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/internal/10111 entering state [calling] recv 354 bytes from udp/[192.168.1.122]:5060 at 18:40:31.031915: ------------------------------------------------------------------------ SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.1.116:5070;rport=5070;branch=z9hG4bKDvK1NF3F0K70m From: "10111" ;tag=3cZNea7Be27cB To: Call-ID: 7d85cf3d-60fb-122c-f480-001517871e28 CSeq: 110076879 INVITE Server: OpenSIPS (1.4.3-notls (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ recv 382 bytes from udp/[192.168.1.122]:5060 at 18:40:31.035869: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.116:5070 ;received=192.168.1.116;rport=5070;branch=z9hG4bKDvK1NF3F0K70m From: "10111" ;tag=3cZNea7Be27cB To: ;tag=117 Call-ID: 7d85cf3d-60fb-122c-f480-001517871e28 CSeq: 110076879 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ recv 688 bytes from udp/[192.168.1.122]:5060 at 18:40:31.036002: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.116:5070 ;received=192.168.1.116;rport=5070;branch=z9hG4bKDvK1NF3F0K70m From: "10111" ;tag=3cZNea7Be27cB To: ;tag=117 Call-ID: 7d85cf3d-60fb-122c-f480-001517871e28 CSeq: 110076879 INVITE Contact: Content-Type: application/sdp Content-Length: 276 v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.107 s=- c=IN IP4 192.168.1.107 t=0 0 m=audio 6000 RTP/AVP 0 9 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 2009-01-19 13:40:31 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/internal/10111 entering state [proceeding] ------------------------------------------------------------------------ 2009-01-19 13:40:31 [NOTICE] sofia.c:2583 sofia_handle_sip_i_state() Ring-Ready sofia/internal/10111! send 414 bytes to udp/[192.168.1.107]:7000 at 18:40:31.036398: ------------------------------------------------------------------------ ACK sip:192.168.1.107:7000;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:5070;rport;branch=z9hG4bKe5ctQamKXvXKg Max-Forwards: 70 From: "10111" ;tag=3cZNea7Be27cB To: ;tag=117 Call-ID: 7d85cf3d-60fb-122c-f480-001517871e28 CSeq: 110076879 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2009-01-19 13:40:31 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/internal/10111 entering state [ready] 2009-01-19 13:40:31 [DEBUG] sofia.c:2533 sofia_handle_sip_i_state() Remote SDP: v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.107 s=- c=IN IP4 192.168.1.107 t=0 0 m=audio 6000 RTP/AVP 0 9 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-01-19 13:40:31 [DEBUG] sofia_glue.c:2409 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2009-01-19 13:40:31 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/internal/10111 PCMU/8000 20 ms 160 samples 2009-01-19 13:40:31 [DEBUG] sofia_glue.c:2373 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-01-19 13:40:31 [DEBUG] sofia_glue.c:1825 sofia_glue_activate_rtp() AUDIO RTP [sofia/internal/10111] 192.168.1.116 port 12396 -> 192.168.1.107 port 6000 codec: 0 ms: 20 2009-01-19 13:40:31 [DEBUG] switch_rtp.c:859 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2009-01-19 13:40:31 [DEBUG] switch_channel.c:1710 switch_channel_perform_mark_answered() Send signal sofia/internal/ 10111 at 192.168.1.122:7001 [BREAK] 2009-01-19 13:40:31 [NOTICE] sofia.c:3018 sofia_handle_sip_i_state() Channel [sofia/internal/10111] has been answered 2009-01-19 13:40:31 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/10111 receive message [SWITCH_MESSAGE_INDICATE_AUDIO_SYNC] 2009-01-19 13:40:31 [DEBUG] switch_channel.c:1631 switch_channel_perform_pre_answer() sofia/internal/10111 at 192.168.1.122:7001receive message [SWITCH_MESSAGE_INDICATE_PROGRESS] 2009-01-19 13:40:31 [INFO] mod_sofia.c:1272 sofia_receive_message() Asked to send early media by sofia/internal/10111 at 192.168.1.122:7001 2009-01-19 13:40:31 [DEBUG] sofia_glue.c:1825 sofia_glue_activate_rtp() AUDIO RTP [sofia/internal/10111 at 192.168.1.122:7001] 192.168.1.116 port 12410 -> 192.168.1.103 port 6444 codec: 0 ms: 20 2009-01-19 13:40:31 [DEBUG] switch_rtp.c:859 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2009-01-19 13:40:31 [INFO] mod_sofia.c:1313 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1232378021 1232378022 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12410 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-01-19 13:40:31 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Ring-Ready sofia/internal/10111 at 192.168.1.122:7001! 2009-01-19 13:40:31 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Pre-Answer sofia/internal/10111 at 192.168.1.122:7001! 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/internal/ 10111 at 192.168.1.122:7001 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_ivr_originate.c:1287 switch_ivr_originate() Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2009-01-19 13:40:31 [DEBUG] switch_ivr_originate.c:1345 switch_ivr_originate() Play Ringback Tone [%(2000, 4000, 440.0, 480.0)] send 1094 bytes to udp/[192.168.1.122]:5060 at 18:40:31.121578: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bK855f.d0dc6ad1.0 Via: SIP/2.0/UDP 192.168.1.103:7001 Record-Route: From: 10111 ;tag=112 To: 0010111 >;tag=235vcFp8gSHtF Call-ID: 112-26869 at 192.168.1.103 CSeq: 2 INVITE Contact: User-Agent: FREESWITCH Media Gateway Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 267 v=0 o=FreeSWITCH 3481863127924572429 7277142438139379053 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12410 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2009-01-19 13:40:31 [DEBUG] switch_channel.c:1768 switch_channel_perform_answer() sofia/internal/10111 at 192.168.1.122:7001receive message [SWITCH_MESSAGE_INDICATE_ANSWER] 2009-01-19 13:40:31 [DEBUG] mod_sofia.c:503 sofia_answer_channel() Local SDP sofia/internal/10111 at 192.168.1.122:7001: v=0 o=FreeSWITCH 1232378021 1232378023 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12410 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/internal/ 10111 at 192.168.1.122:7001 [BREAK] 2009-01-19 13:40:31 [NOTICE] switch_ivr_originate.c:1581 switch_ivr_originate() Channel [sofia/internal/10111 at 192.168.1.122:7001] has been answered 2009-01-19 13:40:31 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/10111 at 192.168.1.122:7001 receive message [SWITCH_MESSAGE_INDICATE_AUDIO_SYNC] 2009-01-19 13:40:31 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/internal/10111 at 192.168.1.122:7001 entering state [early] 2009-01-19 13:40:31 [DEBUG] switch_ivr_originate.c:1621 switch_ivr_originate() Originate Resulted in Success: [sofia/internal/10111] 2009-01-19 13:40:31 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/internal/10111 at 192.168.1.122:7001 entering state [completed] 2009-01-19 13:40:31 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/10111 receive message [SWITCH_MESSAGE_INDICATE_AUDIO_SYNC] 2009-01-19 13:40:31 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/10111 at 192.168.1.122:7001 receive message [SWITCH_MESSAGE_INDICATE_AUDIO_SYNC] 2009-01-19 13:40:31 [DEBUG] switch_ivr_bridge.c:862 switch_ivr_multi_threaded_bridge() sofia/internal/10111 receive message [SWITCH_MESSAGE_INDICATE_BRIDGE] 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/internal/10111 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_ivr_bridge.c:869 switch_ivr_multi_threaded_bridge() sofia/internal/10111 at 192.168.1.122:7001receive message [SWITCH_MESSAGE_INDICATE_BRIDGE] 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/internal/ 10111 at 192.168.1.122:7001 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_ivr_bridge.c:913 switch_ivr_multi_threaded_bridge() (sofia/internal/10111) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/10111 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/internal/10111) Running State Change CS_EXCHANGE_MEDIA 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:445 switch_core_session_run() (sofia/internal/10111) State EXCHANGE_MEDIA 2009-01-19 13:40:31 [DEBUG] mod_sofia.c:404 sofia_on_exchange_media() SOFIA LOOPBACK 2009-01-19 13:40:31 [DEBUG] switch_ivr_bridge.c:293 audio_bridge_thread() sofia/internal/10111 at 192.168.1.122:7001 receive message [SWITCH_MESSAGE_INDICATE_RINGING] 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/internal/ 10111 at 192.168.1.122:7001 [BREAK] send 658 bytes to udp/[192.168.1.122]:5060 at 18:40:31.154213: ------------------------------------------------------------------------ BYE sip:10111 at 192.168.1.103:7001 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:5070;rport;branch=z9hG4bKFe6jS54pt5K6B Route: Max-Forwards: 70 From: 0010111 >;tag=235vcFp8gSHtF To: 10111 ;tag=112 Call-ID: 112-26869 at 192.168.1.103 CSeq: 110076879 BYE Contact: User-Agent: FREESWITCH Media Gateway Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Reason: SIP;cause=408;text="ACK Timeout" Content-Length: 0 ------------------------------------------------------------------------ 2009-01-19 13:40:31 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/internal/10111 at 192.168.1.122:7001 entering state [terminating] send 316 bytes to udp/[192.168.1.122]:5060 at 18:40:31.154944: ------------------------------------------------------------------------ SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bK855f.d0dc6ad1.0 Via: SIP/2.0/UDP 192.168.1.103:7001 From: 10111 ;tag=112 To: 0010111 >;tag=235vcFp8gSHtF Call-ID: 112-26869 at 192.168.1.103 CSeq: 2 INVITE Content-Length: 0 ------------------------------------------------------------------------ 2009-01-19 13:40:31 [DEBUG] switch_ivr_bridge.c:360 audio_bridge_thread() sofia/internal/10111 at 192.168.1.122:7001 ending bridge by request from read function 2009-01-19 13:40:31 [DEBUG] switch_ivr_bridge.c:409 audio_bridge_thread() sofia/internal/10111 at 192.168.1.122:7001 receive message [SWITCH_MESSAGE_INDICATE_UNBRIDGE] 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/internal/ 10111 at 192.168.1.122:7001 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_ivr_bridge.c:435 audio_bridge_thread() BRIDGE THREAD DONE [sofia/internal/10111 at 192.168.1.122:7001] 2009-01-19 13:40:31 [DEBUG] switch_ivr_bridge.c:439 audio_bridge_thread() Send signal sofia/internal/10111 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_ivr_bridge.c:409 audio_bridge_thread() sofia/internal/10111 receive message [SWITCH_MESSAGE_INDICATE_UNBRIDGE] 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/internal/10111 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_ivr_bridge.c:435 audio_bridge_thread() BRIDGE THREAD DONE [sofia/internal/10111] 2009-01-19 13:40:31 [DEBUG] switch_ivr_bridge.c:439 audio_bridge_thread() Send signal sofia/internal/10111 at 192.168.1.122:7001 [BREAK] 2009-01-19 13:40:31 [NOTICE] switch_ivr_bridge.c:470 audio_bridge_on_exchange_media() Hangup sofia/internal/10111 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-01-19 13:40:31 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/internal/10111 [KILL] 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/10111 [BREAK] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:445 switch_core_session_run() (sofia/internal/10111) State EXCHANGE_MEDIA going to sleep 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/internal/10111) Running State Change CS_HANGUP 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/internal/10111) State HANGUP 2009-01-19 13:40:31 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/10111 hanging up, cause: NORMAL_CLEARING 2009-01-19 13:40:31 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending BYE to sofia/internal/10111 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/10111 Standard HANGUP, cause: NORMAL_CLEARING 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/internal/10111) State HANGUP going to sleep 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:938 switch_core_session_thread() Session 805 (sofia/internal/10111) Locked, Waiting on external entities recv 351 bytes from udp/[192.168.1.122]:5060 at 18:40:31.177548: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.116:5070 ;received=192.168.1.116;rport=5070;branch=z9hG4bKFe6jS54pt5K6B From: 0010111 >;tag=235vcFp8gSHtF To: 10111 ;tag=112 Call-ID: 112-26869 at 192.168.1.103 CSeq: 110076879 BYE Contact: Content-Length: 0 ------------------------------------------------------------------------ 2009-01-19 13:40:31 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/internal/10111 at 192.168.1.122:7001 entering state [terminated] send 657 bytes to udp/[192.168.1.107]:7000 at 18:40:31.178007: ------------------------------------------------------------------------ BYE sip:192.168.1.107:7000;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:5070;rport;branch=z9hG4bKgQZBU0NtQeaSQ Max-Forwards: 70 From: "10111" ;tag=3cZNea7Be27cB To: ;tag=117 Call-ID: 7d85cf3d-60fb-122c-f480-001517871e28 CSeq: 110076880 BYE Contact: User-Agent: FREESWITCH Media Gateway Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ recv 354 bytes from udp/[192.168.1.122]:5060 at 18:40:31.178123: ------------------------------------------------------------------------ ACK sip:0010111 at 192.168.1.116:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bK855f.d0dc6ad1.0 From: 10111 ;tag=112 Call-ID: 112-26869 at 192.168.1.103 To: 0010111 >;tag=235vcFp8gSHtF CSeq: 2 ACK Max-Forwards: 70 User-Agent: OpenSIPS (1.4.3-notls (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ 2009-01-19 13:40:31 [NOTICE] switch_ivr_bridge.c:954 switch_ivr_multi_threaded_bridge() Hangup sofia/internal/ 10111 at 192.168.1.122:7001 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-19 13:40:31 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/internal/ 10111 at 192.168.1.122:7001 [KILL] 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/ 10111 at 192.168.1.122:7001 [BREAK] 2009-01-19 13:40:31 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 805 (sofia/internal/10111) Ended 2009-01-19 13:40:31 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/internal/10111 [CS_HANGUP] 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) State EXECUTE going to sleep 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) Running State Change CS_HANGUP 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) State HANGUP 2009-01-19 13:40:31 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/10111 at 192.168.1.122:7001 hanging up, cause: NORMAL_CLEARING 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/10111 at 192.168.1.122:7001Standard HANGUP, cause: NORMAL_CLEARING 2009-01-19 13:40:31 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/internal/10111 at 192.168.1.122:7001) State HANGUP going to sleep 2009-01-19 13:40:31 [DEBUG] switch_core_session.c:938 switch_core_session_thread() Session 804 (sofia/internal/ 10111 at 192.168.1.122:7001) Locked, Waiting on external entities 2009-01-19 13:40:31 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 804 (sofia/internal/ 10111 at 192.168.1.122:7001) Ended 2009-01-19 13:40:31 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/internal/ 10111 at 192.168.1.122:7001 [CS_HANGUP] send 657 bytes to udp/[192.168.1.107]:7000 at 18:40:31.677674: ------------------------------------------------------------------------ BYE sip:192.168.1.107:7000;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:5070;rport;branch=z9hG4bKgQZBU0NtQeaSQ Max-Forwards: 70 From: "10111" ;tag=3cZNea7Be27cB To: ;tag=117 Call-ID: 7d85cf3d-60fb-122c-f480-001517871e28 CSeq: 110076880 BYE Contact: User-Agent: FREESWITCH Media Gateway Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/874124a7/attachment-0001.html From michal.bielicki at voiceworks.pl Mon Jan 19 05:53:16 2009 From: michal.bielicki at voiceworks.pl (Michal Bielicki) Date: Mon, 19 Jan 2009 14:53:16 +0100 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <49743CF8.6050804@ewetel.de> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> Message-ID: <497485CC.6060906@voiceworks.pl> It is both :) Helmut Kuper schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello Michael, > > I'm currently on my way to put all those shell and perl stuff into C > code. As soon as that works I try to put it in ozmod_isdn. So after that > FS will be able to enable Q931Pcapfile generation. After that I will > send the patch to FS. > > Hope this is ok for you. > > > What do you mean with "help with jira"? I thought it is just a bug > tracker... If it's also a feature request tool, I will feed it with > documentation about what I did. > > > regards > helmut > > Am 17.01.2009 21:44, schrieb Michael S Collins: > >> Guys this is awesome! Helmut, if you need any help with jira just let >> me know. >> -MC >> >> Sent from my iPhone >> >> On Jan 17, 2009, at 9:20 AM, Brian West wrote: >> >> >>> Maybe open a jira with this info? Maybe it can all be done as a one >>> step process in the ozmod_isdn ;) >>> >>> /b >>> >>> On Jan 17, 2009, at 10:58 AM, Helmut Kuper wrote: >>> >>> >>>> Hi Brian, >>>> >>>> currently it's a simple perl script which greps all Q931 hexdumps >>>> from >>>> FS logfile converting them to a TPKT packet, and writing those to a >>>> separate local file (wireshark's text2pcap file format). This 1st >>>> step >>>> is easy to put right into FS ozmod_isdn debug code. >>>> The 2nd step is to convert the new file into a .pcap file with adding >>>> TCP/IP dummy packet in front of each TPKT packet. This is done via >>>> test2pcap. This .pcap file is ready to be decoded by wireshark. >>>> >>>> I put both steps into a little shell script and added a 3rd step to >>>> get >>>> those .pcap file emailed from the Server to my Desktop. >>>> >>>> regards >>>> helmut >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkl0PPgACgkQ4tZeNddg3dwKMACfYYHglospD7FJeY4Ne2Q8qBWJ > pNIAoKtEXQ+RnVg2ahySjd5zKfUcOowQ > =1abg > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/52b65f06/attachment.html From anthony.minessale at gmail.com Mon Jan 19 05:53:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 19 Jan 2009 07:53:17 -0600 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <49743CF8.6050804@ewetel.de> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> Message-ID: <191c3a030901190553h4dcf607es33e58d08eb0ec9f6@mail.gmail.com> jira is an issue tracker in general. We use it for bugs, features, bounties and anything else that is an ongoing process that we want to do a workflow on. There is an issue type of new feature you can file such things under and assign it to yourself so we can track and complete the task in a managed way. On Mon, Jan 19, 2009 at 2:42 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello Michael, > > I'm currently on my way to put all those shell and perl stuff into C > code. As soon as that works I try to put it in ozmod_isdn. So after that > FS will be able to enable Q931Pcapfile generation. After that I will > send the patch to FS. > > Hope this is ok for you. > > > What do you mean with "help with jira"? I thought it is just a bug > tracker... If it's also a feature request tool, I will feed it with > documentation about what I did. > > > regards > helmut > > Am 17.01.2009 21:44, schrieb Michael S Collins: > > Guys this is awesome! Helmut, if you need any help with jira just let > > me know. > > -MC > > > > Sent from my iPhone > > > > On Jan 17, 2009, at 9:20 AM, Brian West wrote: > > > >> Maybe open a jira with this info? Maybe it can all be done as a one > >> step process in the ozmod_isdn ;) > >> > >> /b > >> > >> On Jan 17, 2009, at 10:58 AM, Helmut Kuper wrote: > >> > >>> Hi Brian, > >>> > >>> currently it's a simple perl script which greps all Q931 hexdumps > >>> from > >>> FS logfile converting them to a TPKT packet, and writing those to a > >>> separate local file (wireshark's text2pcap file format). This 1st > >>> step > >>> is easy to put right into FS ozmod_isdn debug code. > >>> The 2nd step is to convert the new file into a .pcap file with adding > >>> TCP/IP dummy packet in front of each TPKT packet. This is done via > >>> test2pcap. This .pcap file is ready to be decoded by wireshark. > >>> > >>> I put both steps into a little shell script and added a 3rd step to > >>> get > >>> those .pcap file emailed from the Server to my Desktop. > >>> > >>> regards > >>> helmut > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkl0PPgACgkQ4tZeNddg3dwKMACfYYHglospD7FJeY4Ne2Q8qBWJ > pNIAoKtEXQ+RnVg2ahySjd5zKfUcOowQ > =1abg > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/9c776158/attachment.html From regs at kinetix.gr Mon Jan 19 05:57:57 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Mon, 19 Jan 2009 15:57:57 +0200 Subject: [Freeswitch-users] mod_opal first unsuccessful test In-Reply-To: <04fb01c9783d$6a065d70$3e131850$@jongbloed@bigpond.com> References: <496B66F3.9000902@kinetix.gr> <191c3a030901121119o688c722er745106b014367569@mail.gmail.com> <024701c97510$cb9a1890$62ce49b0$@jongbloed@bigpond.com> <496C7EFC.8060401@kinetix.gr> <496C834E.9070608@kinetix.gr> <02d101c975d1$aebd0e50$0c372af0$@jongbloed@bigpond.com> <496DCC18.9040006@kinetix.gr> <49706258.50501@kinetix.gr> <04fb01c9783d$6a065d70$3e131850$@jongbloed@bigpond.com> Message-ID: <497486E5.7080008@kinetix.gr> Sorry. I am attaching it know Robert Jongbloed wrote: > > Um, there is nothing attached .... > > > > Robert Jongbloed > > OPAL/OpenH323/PTLib Architect and Co-founder. > > > > *From:* Apostolos Pantsiopoulos [mailto:regs at kinetix.gr] > *Sent:* Friday, 16 January 2009 9:33 PM > *To:* Robert Jongbloed; freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_opal first unsuccessful test > > > > Any news regarding this issue? > > Apostolos Pantsiopoulos wrote: > > I am attaching the wireshark capture. Openphone is on xxx.xxx.xxx.202 and > FS is on xxx.xxx.xxx.212 > > Robert Jongbloed wrote: > > Can you send me a WireShark capture? > > > > Robert Jongbloed > > OPAL/OpenH323/PTLib Architect and Co-founder. > > > > *From:* Apostolos Pantsiopoulos [mailto:regs at kinetix.gr] > *Sent:* Tuesday, 13 January 2009 11:05 PM > *To:* freeswitch-users at lists.freeswitch.org > > *Cc:* Robert Jongbloed > *Subject:* Re: [Freeswitch-users] mod_opal first unsuccessful test > > > > I also tried using Ekiga - which is OPAL based - and got the same > behavior. No audio - although I can see RTP packets. > > Apostolos Pantsiopoulos wrote: > > Hi, > > Yes, openPhone is working with my soundcard. I am using it > every day for testing purposes. I use the 1.8.1 version. Is there a newer > version that uses OPAL? I didn't know that. Where can I get it from? > > Robert Jongbloed wrote: > > Hi guys, > > > > I was using the OpenPhone that you build with OPAL for my testing. So > that is identical (I think) to you. > > > > I have not (yet) do any third party client testing. > > > > Two ALERTING messages are fine, perfectly legal and OPAL can handle it. > > > > You say you can see the RTP packets flowing so that implies that the > mod_opal is actually working, so let's look somewhere else. Have you > confirmed that OpenPhone is using the sound card correctly? Made a > call between two machines JUST using OpenPhone for example? > > > > > > Robert Jongbloed > > OPAL/OpenH323/PTLib Architect and Co-founder. > > > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Tuesday, 13 January 2009 6:20 AM > *To:* Robert Jongbloed > *Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test > > > > Heh, > what client are you using in your tests that are working? > > > > ---------- Forwarded message ---------- > From: *Apostolos Pantsiopoulos* > > Date: Mon, Jan 12, 2009 at 9:51 AM > Subject: [Freeswitch-users] mod_opal first unsuccessful test > To: freeswitch-users at lists.freeswitch.org > > > > Hi, > > I successfully compiled mod_opal using the latest svn for both opal > and ptlib as Brian suggested. > > When I try to establish a call using h323 from my openphone client > I get no audio although I can see RTP packets in both directions when I am > doing a capture. Some details : > > I am using the 11094 revision of the FS trunk. > I am using the PCMU codec. > I tried dialing the default IVR (5000) and other testing extensions > (freeswitch conference, echo test etc.) > I tried using fast start on and off , h245 tunneling on and off, h245 in > SETUP on and off. > > In my captures I have also noticed a strange behavior : FS sends to > my client 2 "alerting" packets > for no apparent reason. Could this be a cause of the problem? > > Had anyone any success with mod_opal lately? If yes, could you > please reply quoting your config > options (both on FS and on your client)? > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > ------------------------------------------------------------------------ > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/0e33c2e4/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: tech02.cap Type: application/octet-stream Size: 103881 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/0e33c2e4/attachment-0001.obj From anthony.minessale at gmail.com Mon Jan 19 06:04:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 19 Jan 2009 08:04:43 -0600 Subject: [Freeswitch-users] mod_opal first unsuccessful test In-Reply-To: <497486E5.7080008@kinetix.gr> References: <496B66F3.9000902@kinetix.gr> <191c3a030901121119o688c722er745106b014367569@mail.gmail.com> <496C7EFC.8060401@kinetix.gr> <496C834E.9070608@kinetix.gr> <496DCC18.9040006@kinetix.gr> <49706258.50501@kinetix.gr> <497486E5.7080008@kinetix.gr> Message-ID: <191c3a030901190604w10296892uc696f9b1237825c5@mail.gmail.com> robert is not on this email list so I never see his replies nor does he probably see mine. Can you please move this thread to be between our 3 email addresses directly? On Mon, Jan 19, 2009 at 7:57 AM, Apostolos Pantsiopoulos wrote: > Sorry. I am attaching it know > > Robert Jongbloed wrote: > > Um, there is nothing attached .... > > > > Robert Jongbloed > > OPAL/OpenH323/PTLib Architect and Co-founder. > > > > *From:* Apostolos Pantsiopoulos [mailto:regs at kinetix.gr ] > > *Sent:* Friday, 16 January 2009 9:33 PM > *To:* Robert Jongbloed; freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] mod_opal first unsuccessful test > > > > Any news regarding this issue? > > Apostolos Pantsiopoulos wrote: > > I am attaching the wireshark capture. Openphone is on xxx.xxx.xxx.202 and > FS is on xxx.xxx.xxx.212 > > Robert Jongbloed wrote: > > Can you send me a WireShark capture? > > > > Robert Jongbloed > > OPAL/OpenH323/PTLib Architect and Co-founder. > > > > *From:* Apostolos Pantsiopoulos [mailto:regs at kinetix.gr ] > > *Sent:* Tuesday, 13 January 2009 11:05 PM > *To:* freeswitch-users at lists.freeswitch.org > *Cc:* Robert Jongbloed > *Subject:* Re: [Freeswitch-users] mod_opal first unsuccessful test > > > > I also tried using Ekiga - which is OPAL based - and got the same > behavior. No audio - although I can see RTP packets. > > Apostolos Pantsiopoulos wrote: > > Hi, > > Yes, openPhone is working with my soundcard. I am using it > every day for testing purposes. I use the 1.8.1 version. Is there a newer > version that uses OPAL? I didn't know that. Where can I get it from? > > Robert Jongbloed wrote: > > Hi guys, > > > > I was using the OpenPhone that you build with OPAL for my testing. So that > is identical (I think) to you. > > > > I have not (yet) do any third party client testing. > > > > Two ALERTING messages are fine, perfectly legal and OPAL can handle it. > > > > You say you can see the RTP packets flowing so that implies that the > mod_opal is actually working, so let's look somewhere else. Have you > confirmed that OpenPhone is using the sound card correctly? Made a call > between two machines JUST using OpenPhone for example? > > > > > > Robert Jongbloed > > OPAL/OpenH323/PTLib Architect and Co-founder. > > > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > > *Sent:* Tuesday, 13 January 2009 6:20 AM > *To:* Robert Jongbloed > *Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test > > > > Heh, > what client are you using in your tests that are working? > > > > ---------- Forwarded message ---------- > From: *Apostolos Pantsiopoulos* > Date: Mon, Jan 12, 2009 at 9:51 AM > Subject: [Freeswitch-users] mod_opal first unsuccessful test > To: freeswitch-users at lists.freeswitch.org > > > Hi, > > I successfully compiled mod_opal using the latest svn for both opal > and ptlib as Brian suggested. > > When I try to establish a call using h323 from my openphone client > I get no audio although I can see RTP packets in both directions when I am > doing a capture. Some details : > > I am using the 11094 revision of the FS trunk. > I am using the PCMU codec. > I tried dialing the default IVR (5000) and other testing extensions > (freeswitch conference, echo test etc.) > I tried using fast start on and off , h245 tunneling on and off, h245 in > SETUP on and off. > > In my captures I have also noticed a strange behavior : FS sends to > my client 2 "alerting" packets > for no apparent reason. Could this be a cause of the problem? > > Had anyone any success with mod_opal lately? If yes, could you > please reply quoting your config > options (both on FS and on your client)? > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > > > -- > > ------------------------------------------- > > Apostolos Pantsiopoulos > > Kinetix Tele.com R & D > > email: regs at kinetix.gr > > ------------------------------------------- > > > > ------------------------------ > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > > ------------------------------------------- > > Apostolos Pantsiopoulos > > Kinetix Tele.com R & D > > email: regs at kinetix.gr > > ------------------------------------------- > > > > > -- > > ------------------------------------------- > > Apostolos Pantsiopoulos > > Kinetix Tele.com R & D > > email: regs at kinetix.gr > > ------------------------------------------- > > > > > -- > > ------------------------------------------- > > Apostolos Pantsiopoulos > > Kinetix Tele.com R & D > > email: regs at kinetix.gr > > ------------------------------------------- > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/6eb85131/attachment.html From regs at kinetix.gr Mon Jan 19 06:16:14 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Mon, 19 Jan 2009 16:16:14 +0200 Subject: [Freeswitch-users] enforcing codec in gateway (or call) level Message-ID: <49748B2E.9010909@kinetix.gr> I am using xml_curl to create my dialplans on the fly. I want to be able to use deferent codec preferences on each call (depending on various conditions i.e. gateway,destination etc.) I understand that I have to set inbound-late-negotiation=true in my profile in order for the dialplan to be processed before the codec negotiation. The only thing that I cannot figure out is how to "tell" FS to use a specific codec (or series of codecs) before bridging a call. Which app do I use or which variable do I set (without of course changing any global variable or profile's "codec-prefs" variable)? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From regs at kinetix.gr Mon Jan 19 06:26:55 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Mon, 19 Jan 2009 16:26:55 +0200 Subject: [Freeswitch-users] enforcing codec in gateway (or call) level (SOLVED) In-Reply-To: <49748B2E.9010909@kinetix.gr> References: <49748B2E.9010909@kinetix.gr> Message-ID: <49748DAF.4040401@kinetix.gr> Sorry I just found that you can achieve that by codec_string and absolute_codec_string by grepping the code :) Apostolos Pantsiopoulos wrote: > I am using xml_curl to create my dialplans on the fly. > I want to be able to use deferent codec preferences on > each call (depending on various conditions i.e. gateway,destination etc.) > I understand that I have to set inbound-late-negotiation=true in my profile > in order for the dialplan to be processed before the codec negotiation. > > The only thing that I cannot figure out is how to "tell" FS to use > a specific codec (or series of codecs) before bridging a call. Which > app do I use or which variable do I set (without of course > changing any global variable or profile's "codec-prefs" variable)? > > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From brian at freeswitch.org Mon Jan 19 06:37:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 19 Jan 2009 08:37:40 -0600 Subject: [Freeswitch-users] Dialling from a conferance In-Reply-To: <20090119091313.GA7773@cpdata.co.za> References: <20090119091313.GA7773@cpdata.co.za> Message-ID: if the far end provides ringback when you use conference dial then you should hear it. Have you not tried that? /b On Jan 19, 2009, at 3:13 AM, Sias Mey wrote: > I have only one question whith implemeting all this (and that is a > huge > up to the FS dev for keeping the API constant accross seperate > interfaces) Is it possible to play ringing into the conferance while > using the conferance dial command to make a new call? From brian at freeswitch.org Mon Jan 19 06:38:54 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 19 Jan 2009 08:38:54 -0600 Subject: [Freeswitch-users] strange 503 error thrown by freeswitch In-Reply-To: <27c25bc40901190306l91cb7dbr2ace115dd561646c@mail.gmail.com> References: <27c25bc40901190223q5de210a2j6ed83bb2c7eacb27@mail.gmail.com> <27c25bc40901190306l91cb7dbr2ace115dd561646c@mail.gmail.com> Message-ID: <8950B814-E582-4C3A-9EFD-51078DD11C29@freeswitch.org> You have NAT issues... ACK Timeout. /b On Jan 19, 2009, at 5:06 AM, Juan Backson wrote: > BYE sip:10111 at 192.168.1.103:7001 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.1.116:5070;rport;branch=z9hG4bKFe6jS54pt5K6B > Route: > Max-Forwards: 70 > From: 0010111 ;tag=235vcFp8gSHtF > To: 10111 ;tag=112 > Call-ID: 112-26869 at 192.168.1.103 > CSeq: 110076879 BYE > Contact: > User-Agent: FREESWITCH Media Gateway > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Reason: SIP;cause=408;text="ACK Timeout" > Content-Length: 0 > > > ------------------------------------------------------------------------ > 2009-01-19 13:40:31 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() > Channel sofia/internal/10111 at 192.168.1.122:7001 entering state > [terminating] > send 316 bytes to udp/[192.168.1.122]:5060 at 18:40:31.154944: > > ------------------------------------------------------------------------ > SIP/2.0 500 Internal Server Error > Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bK855f.d0dc6ad1.0 > Via: SIP/2.0/UDP 192.168.1.103:7001 > From: 10111 ;tag=112 > To: 0010111 ;tag=235vcFp8gSHtF > Call-ID: 112-26869 at 192.168.1.103 > CSeq: 2 INVITE > Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/88d1ce42/attachment.html From pablosaro at gmail.com Mon Jan 19 06:49:24 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Mon, 19 Jan 2009 12:49:24 -0200 Subject: [Freeswitch-users] Conference auto-record not working Message-ID: <247f8100901190649g4086b3d8x7cbedf6a87be3ff2@mail.gmail.com> Hi there, I've set up in my default conference context auto-record as follows: but recording never starts. I've checked this by starting FS in console mode but no errors or useful messages to figure out what is going on. My conference extension is set up as follows: Could you please help me finding some missing configuration or error? Let me know if you need further information. Thanks in advance. Pablo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/f3ce2109/attachment.html From brian at freeswitch.org Mon Jan 19 06:58:56 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 19 Jan 2009 08:58:56 -0600 Subject: [Freeswitch-users] Conference auto-record not working In-Reply-To: <247f8100901190649g4086b3d8x7cbedf6a87be3ff2@mail.gmail.com> References: <247f8100901190649g4086b3d8x7cbedf6a87be3ff2@mail.gmail.com> Message-ID: What SVN Rev are you working on? /b On Jan 19, 2009, at 8:49 AM, Pablo Hernan Saro wrote: > Hi there, > > I've set up in my default conference context auto-record as follows: > > > > but recording never starts. I've checked this by starting FS in > console mode but no errors or useful messages to figure out what is > going on. > My conference extension is set up as follows: > > > > > > > > > > > Could you please help me finding some missing configuration or error? > Let me know if you need further information. > Thanks in advance. > > Pablo > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From juanbackson at gmail.com Mon Jan 19 07:06:20 2009 From: juanbackson at gmail.com (Juan Backson) Date: Mon, 19 Jan 2009 23:06:20 +0800 Subject: [Freeswitch-users] strange 503 error thrown by freeswitch In-Reply-To: <8950B814-E582-4C3A-9EFD-51078DD11C29@freeswitch.org> References: <27c25bc40901190223q5de210a2j6ed83bb2c7eacb27@mail.gmail.com> <27c25bc40901190306l91cb7dbr2ace115dd561646c@mail.gmail.com> <8950B814-E582-4C3A-9EFD-51078DD11C29@freeswitch.org> Message-ID: <27c25bc40901190706q31ab7b6bqf688b677813a7c67@mail.gmail.com> Hi, I don't think so because I am testing everything within LAN. Also, amount 290 calls, only 1 gets that error. JB On Mon, Jan 19, 2009 at 10:38 PM, Brian West wrote: > You have NAT issues... ACK Timeout. > /b > > On Jan 19, 2009, at 5:06 AM, Juan Backson wrote: > > BYE sip:10111 at 192.168.1.103:7001 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.116:5070;rport;branch=z9hG4bKFe6jS54pt5K6B > Route: > Max-Forwards: 70 > From: 0010111 > >;tag=235vcFp8gSHtF > To: 10111 ;tag=112 > Call-ID: 112-26869 at 192.168.1.103 > CSeq: 110076879 BYE > Contact: > User-Agent: FREESWITCH Media Gateway > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Reason: SIP;cause=408;text="ACK Timeout" > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-01-19 13:40:31 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel > sofia/internal/10111 at 192.168.1.122:7001 entering state [terminating] > send 316 bytes to udp/[192.168.1.122]:5060 at 18:40:31.154944: > ------------------------------------------------------------------------ > SIP/2.0 500 Internal Server Error > Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bK855f.d0dc6ad1.0 > Via: SIP/2.0/UDP 192.168.1.103:7001 > From: 10111 ;tag=112 > To: 0010111 > >;tag=235vcFp8gSHtF > Call-ID: 112-26869 at 192.168.1.103 > CSeq: 2 INVITE > Content-Length: 0 > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/4006e88b/attachment.html From brian at freeswitch.org Mon Jan 19 07:09:58 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 19 Jan 2009 09:09:58 -0600 Subject: [Freeswitch-users] Conference auto-record not working In-Reply-To: <247f8100901190649g4086b3d8x7cbedf6a87be3ff2@mail.gmail.com> References: <247f8100901190649g4086b3d8x7cbedf6a87be3ff2@mail.gmail.com> Message-ID: Pablo, I tested this on the latest SVN and it works fine. You'll need to update there was a point where it was broken for a little bit. /b On Jan 19, 2009, at 8:49 AM, Pablo Hernan Saro wrote: > Hi there, > > I've set up in my default conference context auto-record as follows: > > > > but recording never starts. I've checked this by starting FS in > console mode but no errors or useful messages to figure out what is > going on. > My conference extension is set up as follows: > > > > > > > > > > > Could you please help me finding some missing configuration or error? > Let me know if you need further information. > Thanks in advance. > > Pablo > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sias at cpdata.co.za Mon Jan 19 07:10:39 2009 From: sias at cpdata.co.za (Sias Mey) Date: Mon, 19 Jan 2009 17:10:39 +0200 Subject: [Freeswitch-users] Dialling from a conferance In-Reply-To: References: <20090119091313.GA7773@cpdata.co.za> Message-ID: <20090119151039.GA24098@cpdata.co.za> Hmm all testing was done to another extension on the same FS server... but as I said I am busy reinstalling and setting up the new version. Ill make sure to try some additional tests asap. I just had it in my mind that I couldent get it to ring last time and havent seen anything said about it on the list so... I asked in case someone like you who knows more than I do can save me a weeks worth of wondering about it. Thanks you very much. Sias On Mon, Jan 19, 2009 at 08:37:40AM -0600, Brian West wrote: > if the far end provides ringback when you use conference dial then you > should hear it. Have you not tried that? > > /b > > On Jan 19, 2009, at 3:13 AM, Sias Mey wrote: > > > I have only one question whith implemeting all this (and that is a > > huge > > up to the FS dev for keeping the API constant accross seperate > > interfaces) Is it possible to play ringing into the conferance while > > using the conferance dial command to make a new call? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Jan 19 07:11:26 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 19 Jan 2009 09:11:26 -0600 Subject: [Freeswitch-users] strange 503 error thrown by freeswitch In-Reply-To: <27c25bc40901190706q31ab7b6bqf688b677813a7c67@mail.gmail.com> References: <27c25bc40901190223q5de210a2j6ed83bb2c7eacb27@mail.gmail.com> <27c25bc40901190306l91cb7dbr2ace115dd561646c@mail.gmail.com> <8950B814-E582-4C3A-9EFD-51078DD11C29@freeswitch.org> <27c25bc40901190706q31ab7b6bqf688b677813a7c67@mail.gmail.com> Message-ID: <2820B56A-BA16-4990-94F4-B92BF984F662@freeswitch.org> Well one isn't acking the call. So it times out and we hang it up. If its just one then its a fluke or your device is broken. If you can narrow down the cause please open a jira. Thanks, /b On Jan 19, 2009, at 9:06 AM, Juan Backson wrote: > SIP;cause=408;text="ACK Timeout" From pablosaro at gmail.com Mon Jan 19 07:15:42 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Mon, 19 Jan 2009 13:15:42 -0200 Subject: [Freeswitch-users] Conference auto-record not working In-Reply-To: References: <247f8100901190649g4086b3d8x7cbedf6a87be3ff2@mail.gmail.com> Message-ID: <247f8100901190715w1b6f833fne9b7f161e5e6e535@mail.gmail.com> sorry. it's 1.0.1 stable (tarrball) On Mon, Jan 19, 2009 at 12:58 PM, Brian West wrote: > What SVN Rev are you working on? > > /b > > On Jan 19, 2009, at 8:49 AM, Pablo Hernan Saro wrote: > > > Hi there, > > > > I've set up in my default conference context auto-record as follows: > > > > > > > > but recording never starts. I've checked this by starting FS in > > console mode but no errors or useful messages to figure out what is > > going on. > > My conference extension is set up as follows: > > > > > > > > > > > > > > > > > > > > > > Could you please help me finding some missing configuration or error? > > Let me know if you need further information. > > Thanks in advance. > > > > Pablo > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/10c9e117/attachment.html From pablosaro at gmail.com Mon Jan 19 07:16:31 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Mon, 19 Jan 2009 13:16:31 -0200 Subject: [Freeswitch-users] Conference auto-record not working In-Reply-To: References: <247f8100901190649g4086b3d8x7cbedf6a87be3ff2@mail.gmail.com> Message-ID: <247f8100901190716w3c19cdadi63cf9a878fb23e64@mail.gmail.com> Thanks Brian! I'll try the latest stable. Have a nice day On Mon, Jan 19, 2009 at 1:09 PM, Brian West wrote: > Pablo, > I tested this on the latest SVN and it works fine. You'll need to > update there was a point where it was broken for a little bit. > > /b > > On Jan 19, 2009, at 8:49 AM, Pablo Hernan Saro wrote: > > > Hi there, > > > > I've set up in my default conference context auto-record as follows: > > > > > > > > but recording never starts. I've checked this by starting FS in > > console mode but no errors or useful messages to figure out what is > > going on. > > My conference extension is set up as follows: > > > > > > > > > > > > > > > > > > > > > > Could you please help me finding some missing configuration or error? > > Let me know if you need further information. > > Thanks in advance. > > > > Pablo > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/2d1d3ceb/attachment.html From brian at freeswitch.org Mon Jan 19 07:22:52 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 19 Jan 2009 09:22:52 -0600 Subject: [Freeswitch-users] Conference auto-record not working In-Reply-To: <247f8100901190716w3c19cdadi63cf9a878fb23e64@mail.gmail.com> References: <247f8100901190649g4086b3d8x7cbedf6a87be3ff2@mail.gmail.com> <247f8100901190716w3c19cdadi63cf9a878fb23e64@mail.gmail.com> Message-ID: Please try the latest Trunk... Even 1.0.2 has bugs that are already fixed in trunk. Expect releases more often. Btw we never call a release stable. /b On Jan 19, 2009, at 9:16 AM, Pablo Hernan Saro wrote: > Thanks Brian! I'll try the latest stable. > Have a nice day From pablosaro at gmail.com Mon Jan 19 07:59:56 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Mon, 19 Jan 2009 13:59:56 -0200 Subject: [Freeswitch-users] Conference auto-record not working In-Reply-To: References: <247f8100901190649g4086b3d8x7cbedf6a87be3ff2@mail.gmail.com> <247f8100901190716w3c19cdadi63cf9a878fb23e64@mail.gmail.com> Message-ID: <247f8100901190759n2deab0c6tc7aa07af4654bdb7@mail.gmail.com> Thanks again. Then, I'll try the latest trunk. Pablo On Mon, Jan 19, 2009 at 1:22 PM, Brian West wrote: > Please try the latest Trunk... Even 1.0.2 has bugs that are already > fixed in trunk. Expect releases more often. Btw we never call a > release stable. > > /b > > On Jan 19, 2009, at 9:16 AM, Pablo Hernan Saro wrote: > > > Thanks Brian! I'll try the latest stable. > > Have a nice day > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/cc3012ad/attachment.html From krzysiez at go2.pl Mon Jan 19 02:05:14 2009 From: krzysiez at go2.pl (=?UTF-8?Q?Krzysztof_Zimnicki?=) Date: Mon, 19 Jan 2009 11:05:14 +0100 Subject: [Freeswitch-users] =?utf-8?q?Problem_with_digium_te220p?= Message-ID: Hi, We would like to configure our digium te220p card with freeswitch. Configured OpenZap and install zaptel drivers, but card doesn't work with our freeswitch. When freeswitch start put error on console: 2009-01-17 19:41:06 [NOTICE] zap_io.c:2517 zap_global_init() Modules configured: 1 2009-01-17 19:41:06 [INFO] zap_io.c:2341 zap_load_module() Loading IO from /usr/local/freeswitch/mod/ozmod_zt.so 2009-01-17 19:41:06 [INFO] zap_io.c:2127 load_config() auto-loaded 'zt' 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:38 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 2 as OpenZAP device 1:2 fd:39 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 3 as OpenZAP device 1:3 fd:40 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 4 as OpenZAP device 1:4 fd:41 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 5 as OpenZAP device 1:5 fd:42 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 6 as OpenZAP device 1:6 fd:43 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 7 as OpenZAP device 1:7 fd:44 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 8 as OpenZAP device 1:8 fd:45 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 9 as OpenZAP device 1:9 fd:46 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 10 as OpenZAP device 1:10 fd:47 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 11 as OpenZAP device 1:11 fd:48 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 12 as OpenZAP device 1:12 fd:49 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 13 as OpenZAP device 1:13 fd:50 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 14 as OpenZAP device 1:14 fd:51 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 15 as OpenZAP device 1:15 fd:52 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 16 as OpenZAP device 1:16 fd:53 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 17 as OpenZAP device 1:17 fd:54 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 18 as OpenZAP device 1:18 fd:55 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 19 as OpenZAP device 1:19 fd:56 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 20 as OpenZAP device 1:20 fd:57 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 21 as OpenZAP device 1:21 fd:58 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 22 as OpenZAP device 1:22 fd:59 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 23 as OpenZAP device 1:23 fd:60 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 24 as OpenZAP device 1:24 fd:61 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 25 as OpenZAP device 1:25 fd:62 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 26 as OpenZAP device 1:26 fd:63 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 27 as OpenZAP device 1:27 fd:64 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 28 as OpenZAP device 1:28 fd:65 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 29 as OpenZAP device 1:29 fd:66 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 30 as OpenZAP device 1:30 fd:67 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 31 as OpenZAP device 1:31 fd:68 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 32 as OpenZAP device 2:1 fd:69 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 33 as OpenZAP device 2:2 fd:70 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 34 as OpenZAP device 2:3 fd:71 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 35 as OpenZAP device 2:4 fd:72 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 36 as OpenZAP device 2:5 fd:73 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 37 as OpenZAP device 2:6 fd:74 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 38 as OpenZAP device 2:7 fd:75 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 39 as OpenZAP device 2:8 fd:76 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 40 as OpenZAP device 2:9 fd:77 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 41 as OpenZAP device 2:10 fd:78 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 42 as OpenZAP device 2:11 fd:79 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 43 as OpenZAP device 2:12 fd:80 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 44 as OpenZAP device 2:13 fd:81 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 45 as OpenZAP device 2:14 fd:82 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 46 as OpenZAP device 2:15 fd:83 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 47 as OpenZAP device 2:16 fd:84 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 48 as OpenZAP device 2:17 fd:85 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 49 as OpenZAP device 2:18 fd:86 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 50 as OpenZAP device 2:19 fd:87 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 51 as OpenZAP device 2:20 fd:88 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 52 as OpenZAP device 2:21 fd:89 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 53 as OpenZAP device 2:22 fd:90 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 54 as OpenZAP device 2:23 fd:91 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 55 as OpenZAP device 2:24 fd:92 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 56 as OpenZAP device 2:25 fd:93 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 57 as OpenZAP device 2:26 fd:94 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 58 as OpenZAP device 2:27 fd:95 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 59 as OpenZAP device 2:28 fd:96 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 60 as OpenZAP device 2:29 fd:97 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 61 as OpenZAP device 2:30 fd:98 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 62 as OpenZAP device 2:31 fd:99 2009-01-17 19:41:06 [INFO] zap_io.c:2265 load_config() Configured 62 channel(s) 2009-01-17 19:41:06 [INFO] zap_io.c:2358 zap_load_module() Loading SIG from /usr/local/freeswitch/mod/ozmod_analog.so 2009-01-17 19:41:06 [INFO] zap_io.c:2474 zap_configure_span() auto-loaded 'analog' 2009-01-17 19:41:06 [ERR] mod_openzap.c:1895 load_config() Error finding OpenZAP span id: name:PRI_1 2009-01-17 19:41:06 [ERR] mod_openzap.c:1895 load_config() Error finding OpenZAP span id: name:PRI_2 2009-01-17 19:41:06 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] 2009-01-17 19:41:06 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'openzap' openzap.conf.xml: zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span = 2,0,0,ccs,hdb3,crc4 bchan = 32-46,48-62 dchan = 47 loadzone = ru defaultzone=ru freeswitch.log: 2009-01-17 18:26:57 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/modules.conf. 2009-01-17 18:26:57 [NOTICE] zap_io.c:2517 zap_global_init() Modules configured: 1 2009-01-17 18:26:57 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/openzap.conf. 2009-01-17 18:26:57 [DEBUG] zap_io.c:2110 load_config() found config for span 2009-01-17 18:26:57 [INFO] zap_io.c:2341 zap_load_module() Loading IO from /usr/local/freeswitch/mod/ozmod_wanpipe.so 2009-01-17 18:26:57 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/wanpipe.conf. 2009-01-17 18:26:57 [INFO] zap_io.c:2127 load_config() auto-loaded 'wanpipe' 2009-01-17 18:26:57 [DEBUG] zap_io.c:2163 load_config() created span 1 (span1) of type wanpipe 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 1 [name]=[OpenZAP] 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 1 [number]=[1] 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 1 [fxs-channel]=[1:3-4] 2009-01-17 18:26:57 [DEBUG] zap_io.c:2216 load_config() setting trunk type to 'FXS' start(KEWL) 2009-01-17 18:26:57 [ERR] ozmod_wanpipe.c:440 wp_open_range() failure configuring device s1c3 2009-01-17 18:26:57 [ERR] ozmod_wanpipe.c:440 wp_open_range() failure configuring device s1c4 2009-01-17 18:26:57 [DEBUG] zap_io.c:2110 load_config() found config for span 2009-01-17 18:26:57 [DEBUG] zap_io.c:2163 load_config() created span 2 (span2) of type wanpipe 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 2 [fxo-channel]=[1:1-2] 2009-01-17 18:26:57 [DEBUG] zap_io.c:2205 load_config() setting trunk type to 'FXO' start(KEWL) 2009-01-17 18:26:57 [ERR] ozmod_wanpipe.c:440 wp_open_range() failure configuring device s1c1 2009-01-17 18:26:57 [ERR] ozmod_wanpipe.c:440 wp_open_range() failure configuring device s1c2 2009-01-17 18:26:57 [DEBUG] zap_io.c:2110 load_config() found config for span 2009-01-17 18:26:57 [INFO] zap_io.c:2341 zap_load_module() Loading IO from /usr/local/freeswitch/mod/ozmod_zt.so 2009-01-17 18:26:57 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/zt.conf. 2009-01-17 18:26:57 [INFO] zap_io.c:2127 load_config() auto-loaded 'zt' 2009-01-17 18:26:57 [DEBUG] zap_io.c:2163 load_config() created span 3 (span3) of type zt 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 3 [name]=[OpenZAP] 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 3 [number]=[2] 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 3 [fxs-channel]=[1] 2009-01-17 18:26:57 [DEBUG] zap_io.c:2216 load_config() setting trunk type to 'FXS' start(KEWL) 2009-01-17 18:26:57 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 3:1 fd:40 2009-01-17 18:26:57 [DEBUG] zap_io.c:2110 load_config() found config for span 2009-01-17 18:26:57 [DEBUG] zap_io.c:2163 load_config() created span 4 (span4) of type zt 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 4 [name]=[OpenZAP] 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 4 [number]=[2] 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 4 [fxo-channel]=[3] 2009-01-17 18:26:57 [DEBUG] zap_io.c:2205 load_config() setting trunk type to 'FXO' start(KEWL) 2009-01-17 18:26:57 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 3 as OpenZAP device 4:1 fd:41 2009-01-17 18:26:57 [INFO] zap_io.c:2265 load_config() Configured 2 channel(s) 2009-01-17 18:26:57 [INFO] zap_io.c:2358 zap_load_module() Loading SIG from /usr/local/freeswitch/mod/ozmod_analog.so 2009-01-17 18:26:57 [INFO] zap_io.c:2474 zap_configure_span() auto-loaded 'analog' 2009-01-17 18:26:57 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/tones.conf. 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [dial] = [v=-7;%(1000,0,350,440)] 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [dial] = [350,440] 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [ring] = [v=-7;%(2000,4000,440,480)] 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [ring] = [440,480] 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [busy] = [v=-7;%(500,500,480,620)] 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [busy] = [480,620] 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [attn] = [v=0;%(100,100,1400,2060,2450,2600)] 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [attn] = [1400,2060,2450,2600] 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [callwaiting-sas] = [v=0;%(300,0,440)] 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [callwaiting-sas] = [440] 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [callwaiting-cas] = [v=0;%(80,0,2750,2130)] 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [callwaiting-cas] = [2750,2130] 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [fail1] = [913.8] 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [fail2] = [1370.6] 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [fail3] = [776.7] 2009-01-17 18:26:57 [ERR] mod_openzap.c:1895 load_config() Error finding OpenZAP span id: name:PRI_1 2009-01-17 18:26:57 [ERR] mod_openzap.c:1895 load_config() Error finding OpenZAP span id: name:PRI_2 2009-01-17 18:26:57 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'openzap' 2009-01-17 18:26:57 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'disable_ec' 2009-01-17 18:26:57 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'oz' 2009-01-17 18:26:57 [DEBUG] ozmod_analog.c:875 zap_analog_run() ANALOG thread starting. freeswitch command line: freeswitch at voipgw> oz list API CALL [oz(list)] output: +OK span: 1 (span1) type: analog chan_count: 31 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none freeswitch at voipgw> oz dump 1 1 API CALL [oz(dump 1 1)] output: span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE Any idee where is the problem ? From msc at freeswitch.org Mon Jan 19 09:31:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Jan 2009 09:31:45 -0800 Subject: [Freeswitch-users] Problem with digium te220p In-Reply-To: References: Message-ID: <87f2f3b90901190931t15034affse754fb74cd93d77a@mail.gmail.com> Can you post your openzap.conf file? -MC On Mon, Jan 19, 2009 at 2:05 AM, Krzysztof Zimnicki wrote: > Hi, > > We would like to configure our digium te220p card with freeswitch. > > Configured OpenZap and install zaptel drivers, but card doesn't work with our freeswitch. > > When freeswitch start put error on console: > > 2009-01-17 19:41:06 [NOTICE] zap_io.c:2517 zap_global_init() Modules configured: 1 > 2009-01-17 19:41:06 [INFO] zap_io.c:2341 zap_load_module() Loading IO from /usr/local/freeswitch/mod/ozmod_zt.so > 2009-01-17 19:41:06 [INFO] zap_io.c:2127 load_config() auto-loaded 'zt' > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:38 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 2 as OpenZAP device 1:2 fd:39 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 3 as OpenZAP device 1:3 fd:40 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 4 as OpenZAP device 1:4 fd:41 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 5 as OpenZAP device 1:5 fd:42 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 6 as OpenZAP device 1:6 fd:43 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 7 as OpenZAP device 1:7 fd:44 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 8 as OpenZAP device 1:8 fd:45 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 9 as OpenZAP device 1:9 fd:46 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 10 as OpenZAP device 1:10 fd:47 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 11 as OpenZAP device 1:11 fd:48 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 12 as OpenZAP device 1:12 fd:49 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 13 as OpenZAP device 1:13 fd:50 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 14 as OpenZAP device 1:14 fd:51 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 15 as OpenZAP device 1:15 fd:52 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 16 as OpenZAP device 1:16 fd:53 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 17 as OpenZAP device 1:17 fd:54 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 18 as OpenZAP device 1:18 fd:55 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 19 as OpenZAP device 1:19 fd:56 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 20 as OpenZAP device 1:20 fd:57 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 21 as OpenZAP device 1:21 fd:58 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 22 as OpenZAP device 1:22 fd:59 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 23 as OpenZAP device 1:23 fd:60 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 24 as OpenZAP device 1:24 fd:61 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 25 as OpenZAP device 1:25 fd:62 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 26 as OpenZAP device 1:26 fd:63 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 27 as OpenZAP device 1:27 fd:64 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 28 as OpenZAP device 1:28 fd:65 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 29 as OpenZAP device 1:29 fd:66 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 30 as OpenZAP device 1:30 fd:67 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 31 as OpenZAP device 1:31 fd:68 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 32 as OpenZAP device 2:1 fd:69 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 33 as OpenZAP device 2:2 fd:70 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 34 as OpenZAP device 2:3 fd:71 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 35 as OpenZAP device 2:4 fd:72 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 36 as OpenZAP device 2:5 fd:73 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 37 as OpenZAP device 2:6 fd:74 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 38 as OpenZAP device 2:7 fd:75 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 39 as OpenZAP device 2:8 fd:76 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 40 as OpenZAP device 2:9 fd:77 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 41 as OpenZAP device 2:10 fd:78 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 42 as OpenZAP device 2:11 fd:79 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 43 as OpenZAP device 2:12 fd:80 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 44 as OpenZAP device 2:13 fd:81 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 45 as OpenZAP device 2:14 fd:82 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 46 as OpenZAP device 2:15 fd:83 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 47 as OpenZAP device 2:16 fd:84 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 48 as OpenZAP device 2:17 fd:85 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 49 as OpenZAP device 2:18 fd:86 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 50 as OpenZAP device 2:19 fd:87 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 51 as OpenZAP device 2:20 fd:88 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 52 as OpenZAP device 2:21 fd:89 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 53 as OpenZAP device 2:22 fd:90 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 54 as OpenZAP device 2:23 fd:91 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 55 as OpenZAP device 2:24 fd:92 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 56 as OpenZAP device 2:25 fd:93 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 57 as OpenZAP device 2:26 fd:94 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 58 as OpenZAP device 2:27 fd:95 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 59 as OpenZAP device 2:28 fd:96 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 60 as OpenZAP device 2:29 fd:97 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 61 as OpenZAP device 2:30 fd:98 > 2009-01-17 19:41:06 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 62 as OpenZAP device 2:31 fd:99 > 2009-01-17 19:41:06 [INFO] zap_io.c:2265 load_config() Configured 62 channel(s) > 2009-01-17 19:41:06 [INFO] zap_io.c:2358 zap_load_module() Loading SIG from /usr/local/freeswitch/mod/ozmod_analog.so > 2009-01-17 19:41:06 [INFO] zap_io.c:2474 zap_configure_span() auto-loaded 'analog' > 2009-01-17 19:41:06 [ERR] mod_openzap.c:1895 load_config() Error finding OpenZAP span id: name:PRI_1 > 2009-01-17 19:41:06 [ERR] mod_openzap.c:1895 load_config() Error finding OpenZAP span id: name:PRI_2 > 2009-01-17 19:41:06 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] > 2009-01-17 19:41:06 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'openzap' > > openzap.conf.xml: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > zaptel.conf: > > span=1,1,0,ccs,hdb3,crc4 > bchan=1-15,17-31 > dchan=16 > > span = 2,0,0,ccs,hdb3,crc4 > bchan = 32-46,48-62 > dchan = 47 > > loadzone = ru > defaultzone=ru > > > freeswitch.log: > 2009-01-17 18:26:57 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/modules.conf. > 2009-01-17 18:26:57 [NOTICE] zap_io.c:2517 zap_global_init() Modules configured: 1 > 2009-01-17 18:26:57 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/openzap.conf. > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2110 load_config() found config for span > 2009-01-17 18:26:57 [INFO] zap_io.c:2341 zap_load_module() Loading IO from /usr/local/freeswitch/mod/ozmod_wanpipe.so > 2009-01-17 18:26:57 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/wanpipe.conf. > 2009-01-17 18:26:57 [INFO] zap_io.c:2127 load_config() auto-loaded 'wanpipe' > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2163 load_config() created span 1 (span1) of type wanpipe > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 1 [name]=[OpenZAP] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 1 [number]=[1] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 1 [fxs-channel]=[1:3-4] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2216 load_config() setting trunk type to 'FXS' start(KEWL) > 2009-01-17 18:26:57 [ERR] ozmod_wanpipe.c:440 wp_open_range() failure configuring device s1c3 > 2009-01-17 18:26:57 [ERR] ozmod_wanpipe.c:440 wp_open_range() failure configuring device s1c4 > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2110 load_config() found config for span > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2163 load_config() created span 2 (span2) of type wanpipe > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 2 [fxo-channel]=[1:1-2] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2205 load_config() setting trunk type to 'FXO' start(KEWL) > 2009-01-17 18:26:57 [ERR] ozmod_wanpipe.c:440 wp_open_range() failure configuring device s1c1 > 2009-01-17 18:26:57 [ERR] ozmod_wanpipe.c:440 wp_open_range() failure configuring device s1c2 > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2110 load_config() found config for span > 2009-01-17 18:26:57 [INFO] zap_io.c:2341 zap_load_module() Loading IO from /usr/local/freeswitch/mod/ozmod_zt.so > 2009-01-17 18:26:57 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/zt.conf. > 2009-01-17 18:26:57 [INFO] zap_io.c:2127 load_config() auto-loaded 'zt' > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2163 load_config() created span 3 (span3) of type zt > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 3 [name]=[OpenZAP] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 3 [number]=[2] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 3 [fxs-channel]=[1] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2216 load_config() setting trunk type to 'FXS' start(KEWL) > 2009-01-17 18:26:57 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 3:1 fd:40 > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2110 load_config() found config for span > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2163 load_config() created span 4 (span4) of type zt > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 4 [name]=[OpenZAP] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 4 [number]=[2] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2176 load_config() span 4 [fxo-channel]=[3] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:2205 load_config() setting trunk type to 'FXO' start(KEWL) > 2009-01-17 18:26:57 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 3 as OpenZAP device 4:1 fd:41 > 2009-01-17 18:26:57 [INFO] zap_io.c:2265 load_config() Configured 2 channel(s) > 2009-01-17 18:26:57 [INFO] zap_io.c:2358 zap_load_module() Loading SIG from /usr/local/freeswitch/mod/ozmod_analog.so > 2009-01-17 18:26:57 [INFO] zap_io.c:2474 zap_configure_span() auto-loaded 'analog' > 2009-01-17 18:26:57 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/tones.conf. > 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [dial] = [v=-7;%(1000,0,350,440)] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [dial] = [350,440] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [ring] = [v=-7;%(2000,4000,440,480)] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [ring] = [440,480] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [busy] = [v=-7;%(500,500,480,620)] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [busy] = [480,620] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [attn] = [v=0;%(100,100,1400,2060,2450,2600)] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [attn] = [1400,2060,2450,2600] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [callwaiting-sas] = [v=0;%(300,0,440)] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [callwaiting-sas] = [440] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:411 zap_span_load_tones() added tone generation [callwaiting-cas] = [v=0;%(80,0,2750,2130)] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [callwaiting-cas] = [2750,2130] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [fail1] = [913.8] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [fail2] = [1370.6] > 2009-01-17 18:26:57 [DEBUG] zap_io.c:409 zap_span_load_tones() added tone detect [fail3] = [776.7] > 2009-01-17 18:26:57 [ERR] mod_openzap.c:1895 load_config() Error finding OpenZAP span id: name:PRI_1 > 2009-01-17 18:26:57 [ERR] mod_openzap.c:1895 load_config() Error finding OpenZAP span id: name:PRI_2 > 2009-01-17 18:26:57 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'openzap' > 2009-01-17 18:26:57 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'disable_ec' > 2009-01-17 18:26:57 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'oz' > 2009-01-17 18:26:57 [DEBUG] ozmod_analog.c:875 zap_analog_run() ANALOG thread starting. > > > freeswitch command line: > > freeswitch at voipgw> oz list > API CALL [oz(list)] output: > +OK > span: 1 (span1) > type: analog > chan_count: 31 > dialplan: XML > context: default > dial_regex: > fail_dial_regex: > hold_music: > analog_options none > > > > > freeswitch at voipgw> oz dump 1 1 > API CALL [oz(dump 1 1)] output: > span_id: 1 > chan_id: 1 > physical_span_id: 1 > physical_chan_id: 1 > type: B > state: DOWN > last_state: DOWN > cid_date: > cid_name: > cid_num: > ani: > aniII: > dnis: > rdnis: > cause: NONE > > > Any idee where is the problem ? > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon Jan 19 09:46:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Jan 2009 09:46:41 -0800 Subject: [Freeswitch-users] [FS-es] Spanish Freeswitch community In-Reply-To: <8c1b00b30901171114u64d8942cyb8d05c4efae07e85@mail.gmail.com> References: <8c1b00b30901171114u64d8942cyb8d05c4efae07e85@mail.gmail.com> Message-ID: <87f2f3b90901190946o19f363a1nf21b2bfa62e43edb@mail.gmail.com> If you or anyone else would like to work on translating some of the more important wiki pages into Spanish please let me know. -MC On Sat, Jan 17, 2009 at 11:14 AM, Andr?s Mart?n - martyn wrote: > Hi. > > I think that is a good idea expan the freeswitch project arround the wolrd, > btw this site is specially to hispan community [1], and the irc channel is > #freeswitch-es :D. > > [1] http://www.freeswitch.es > > Regards. > > -- > Andr?s Mart?n Ochoa; > passport: andresmartin at linuxmail.org; > Linux Registered User #436420; > Asterisk User Number: 1000; > PBX: (57) 1 578 20 30; > Ext: 106 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From ajlong at worldlink.net Mon Jan 19 09:51:44 2009 From: ajlong at worldlink.net (Adam Long) Date: Mon, 19 Jan 2009 12:51:44 -0500 Subject: [Freeswitch-users] ODBC in place of db/core.db Message-ID: <008b01c97a5e$97f685c0$c7e39140$@net> Hi Guys, I have read the documentation on the wiki and have successfully compiled FreeSwtich with odbc core support. I am able to get my SIP Profiles and Voicemail databases to load, create, and utilize the ODBC database tables successfully. However, I cannot seem to figure out how to make FreeSwitch use the ODBC database instead of the SQL Lite db/core.db I'm sure it's just a matter of properly placing the following line in one of the configs. If anyone can point me in the right direction I would greatly appreciate it. Thank you! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/1f82dd21/attachment.html From msc at freeswitch.org Mon Jan 19 09:52:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Jan 2009 09:52:09 -0800 Subject: [Freeswitch-users] OpenZap detect tones question In-Reply-To: <4971AA12.10706@novatex.com.au> References: <4971AA12.10706@novatex.com.au> Message-ID: <87f2f3b90901190952k1540de7ct51df55a799c596eb@mail.gmail.com> Scott, Did you get the answer to this question yet? Either way, could you post your openzap.conf and tones.conf files? I'd like to get this one documented on the wiki so that non-US PSTN tones are properly documented. -MC On Sat, Jan 17, 2009 at 1:51 AM, Scott Ellis wrote: > Quick question, when specifying a "detect-busy" tone in the tones.conf > file - is the cadence used? (The US examples to not have cadence) > > In the tests I have does it does not seem to be. > > This is a problem in Australia, as we have managed to have our busy tone > 425Hz, 375ms on 375ms off, also in our dial tone 400+425+450. > > So when I go to dial a call, I often get the Zap channel hanging up > again as it thinks the line is busy. > > Scott > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Mon Jan 19 10:04:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 19 Jan 2009 12:04:08 -0600 Subject: [Freeswitch-users] ODBC in place of db/core.db In-Reply-To: <008b01c97a5e$97f685c0$c7e39140$@net> References: <008b01c97a5e$97f685c0$c7e39140$@net> Message-ID: <191c3a030901191004g5e63c8fbj60863c592ba515c8@mail.gmail.com> it's not possible, the core is only sqlite. On Mon, Jan 19, 2009 at 11:51 AM, Adam Long wrote: > Hi Guys, > > > > I have read the documentation on the wiki and have successfully compiled > FreeSwtich with odbc core support. > > I am able to get my SIP Profiles and Voicemail databases to load, create, > and utilize the ODBC database tables successfully. > > > > However, I cannot seem to figure out how to make FreeSwitch use the ODBC > database instead of the SQL Lite db/core.db > > > > I'm sure it's just a matter of properly placing the following line in one > of the configs? > > > > > > If anyone can point me in the right direction I would greatly appreciate > it. > > > > Thank you! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/3beb486d/attachment.html From scott.ellis at novatex.com.au Mon Jan 19 10:19:52 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Tue, 20 Jan 2009 05:19:52 +1100 Subject: [Freeswitch-users] OpenZap detect tones question In-Reply-To: <87f2f3b90901190952k1540de7ct51df55a799c596eb@mail.gmail.com> References: <4971AA12.10706@novatex.com.au> <87f2f3b90901190952k1540de7ct51df55a799c596eb@mail.gmail.com> Message-ID: <4974C448.4000501@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/b35c8444/attachment.html From msc at freeswitch.org Mon Jan 19 10:21:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Jan 2009 10:21:39 -0800 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <496F98B9.7040403@gmx.net> References: <496E2493.2000207@gmx.net> <87f2f3b90901141307m3d4e0e60hce337ad36b121369@mail.gmail.com> <496EF1C2.8020607@gmx.net> <496F1449.6000407@ewetel.de> <496F18C3.3030807@gmx.net> <496F2C73.9030400@ewetel.de> <496F7413.704@gmx.net> <496F98B9.7040403@gmx.net> Message-ID: <87f2f3b90901191021y4e4c93e2vd97d64889711bf77@mail.gmail.com> Peter, I believe we are in a bit of a holding pattern right now with OpenZAP PRI stuff. We have a super user, Stefan, who is working on some Q931 timers and such but he is working on it in spare time and there's no hard date. If you know someone with serious PRI and C programming skillz who can assist then we'd definitely be willing to have some help. "Patches welcome" as it were. :) Thanks, MC On Thu, Jan 15, 2009 at 12:12 PM, Peter P GMX wrote: > I did some more tests. When I sequentially setup calls (only one > simultaneous call at a time), it works for hundreds of calls. > As soon as I setup 2 calls in parallel ist fails aber a number of calls. > > Please find another debug ouput (now with Q.921 debug also). > The log starts with the latest hangup of a successfull call. After this > one I receive a > "2009-01-15 20:26:46 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received > Release with no matching channel 0" > and later > "2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 > parse error [-3012] [Q931E_INVALID_CRV]" > > Is there anyone to fix it? May I donate some money for fixing that? > > Best regards > Peter > > > Debug: > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame > ----------------- Q.921 Packet [Outgoing] --------------- > SAPI: 0, TEI: 0, C/R: Command (0) > > Type: S Frame, SV: RR (Receive Ready) > P/F: 0, N(R): 81 [V(A): 80, V(R): 81, V(S): 80] > > Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] > ---------------------------------------------- > > 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:813 state_advance() 2:3 STATE > [TERMINATING] > 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:1121 state_advance() > Terminating: Direction Inbound > 2009-01-15 20:26:44 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() > got clear channel sig [STOP] > 2009-01-15 20:26:44 [NOTICE] mod_openzap.c:1437 > on_clear_channel_signal() Hangup OpenZAP/2:3/21658519 [CS_EXECUTE] > [NORMAL_CLEARING] > 2009-01-15 20:26:44 [DEBUG] switch_channel.c:1513 > switch_channel_perform_hangup() Send signal OpenZAP/2:3/21658519 [KILL] > 2009-01-15 20:26:44 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > OpenZAP/2:3/21658519 [BREAK] > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Receiving message from Layer4 > (size: 184, type: 77) > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Sending message to Q.921 > (size: 184) > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Creating Q.931 Message Header: > ProtDisc 8 (0x8), CRV 126 (0x7e), CRVflag: 1 (0x1), MesType: 77 (0x4d) > 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 > -------------------------------------------------------------------------------- > [08 02 80 7e 4d] > > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Got frame from Q.931, type: > 4, tei: 0, size: 9 > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Enqueueing I frame for TEI 0 [0] > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame > ----------------- Q.921 Packet [Outgoing] --------------- > SAPI: 0, TEI: 0, C/R: Command (0) > > Type: I Frame > P/F: 0, N(S): 80, N(R): 81 [V(A): 80, V(R): 81, V(S): 80] > > Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] > ---------------------------------------------- > > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T200 (timeout: 1000 msecs) > started for TEI 0 > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T203 stopped for TEI 0 > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Q931Rx43 return code: 1 > 2009-01-15 20:26:44 [DEBUG] mod_event_socket.c:1922 listener_run() > Session complete, waiting for children > 2009-01-15 20:26:44 [DEBUG] mod_event_socket.c:1946 listener_run() > Connection Closed > 2009-01-15 20:26:44 [DEBUG] switch_ivr_play_say.c:1222 > switch_ivr_play_file() done playing file > 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (OpenZAP/2:3/21658519) State EXECUTE going to > sleep > 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/2:3/21658519) Running State Change > CS_HANGUP > 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/2:3/21658519) State HANGUP > 2009-01-15 20:26:44 [DEBUG] mod_openzap.c:472 channel_on_hangup() > OpenZAP/2:3/21658519 CHANNEL HANGUP > 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() OpenZAP/2:3/21658519 Standard HANGUP, > cause: NORMAL_CLEARING > 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/2:3/21658519) State HANGUP going to sleep > 2009-01-15 20:26:44 [DEBUG] switch_core_session.c:939 > switch_core_session_thread() Session 251 (OpenZAP/2:3/21658519) Locked, > Waiting on external entities > 2009-01-15 20:26:44 [NOTICE] switch_core_session.c:957 > switch_core_session_thread() Session 251 (OpenZAP/2:3/21658519) Ended > 2009-01-15 20:26:44 [NOTICE] switch_core_session.c:959 > switch_core_session_thread() Close Channel OpenZAP/2:3/21658519 [CS_HANGUP] > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() New packet received (4 bytes) > ----------------- Q.921 Packet [Incoming] --------------- > SAPI: 0, TEI: 0, C/R: Response (0) > > Type: S Frame, SV: RR (Receive Ready) > P/F: 0, N(R): 81 [V(A): 80, V(R): 81, V(S): 81] > > Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] > ---------------------------------------------- > > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T200 stopped for TEI 0 > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T203 (timeout: 10000 msecs) > restarted for TEI 0 > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() New packet received (9 bytes) > ----------------- Q.921 Packet [Incoming] --------------- > SAPI: 0, TEI: 0, C/R: Command (1) > > Type: I Frame > P/F: 0, N(S): 81, N(R): 81 [V(A): 81, V(R): 81, V(S): 81] > > Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] > ---------------------------------------------- > > 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 > -------------------------------------------------------------------------------- > [08 02 00 7e 5a] > > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Received message from Q.921 > (ind 4, tei 0, size 9) > MesType: 90, CRVFlag 0 (Originator), CRV 126 (Dialect: 0) > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Sending message to Layer4 > (size: 103) > 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[5a] Size:[103] CRV: 126 (0x7e, CTX: Originator) > 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan > e0020270 (2:3) source isdn_data->channels_remote_crv[0x7e] > 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() Changing > state on 2:3 from TERMINATING to DOWN > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame > ----------------- Q.921 Packet [Outgoing] --------------- > SAPI: 0, TEI: 0, C/R: Response (1) > > Type: S Frame, SV: RR (Receive Ready) > P/F: 0, N(R): 82 [V(A): 81, V(R): 82, V(S): 81] > > Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] > ---------------------------------------------- > > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T200 stopped for TEI 0 > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T203 (timeout: 10000 msecs) > restarted for TEI 0 > 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame > ----------------- Q.921 Packet [Outgoing] --------------- > SAPI: 0, TEI: 0, C/R: Command (0) > > Type: S Frame, SV: RR (Receive Ready) > P/F: 0, N(R): 82 [V(A): 81, V(R): 82, V(S): 81] > > Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] > ---------------------------------------------- > > 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:813 state_advance() 2:3 STATE > [DOWN] > 2009-01-15 20:26:44 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done 2:3 > 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() New packet received (16 bytes) > ----------------- Q.921 Packet [Incoming] --------------- > SAPI: 0, TEI: 0, C/R: Command (1) > > Type: I Frame > P/F: 0, N(S): 82, N(R): 81 [V(A): 81, V(R): 82, V(S): 81] > > Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] > ---------------------------------------------- > > 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 12 > -------------------------------------------------------------------------------- > [08 02 00 7d 4d 08 05 82 e6 33 30 33] > > 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Received message from Q.921 > (ind 4, tei 0, size 16) > MesType: 77, CRVFlag 0 (Originator), CRV 125 (Dialect: 0) > 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Sending message to Layer4 > (size: 110) > 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got > an event! Type:[4d] Size:[110] CRV: 125 (0x7d, CTX: Originator) > 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 > (-1:-1) source isdn_data->channels_remote_crv[0x7d] > 2009-01-15 20:26:46 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received > Release with no matching channel 0 > 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Sending message to Q.921 > (size: 110) > 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Creating Q.931 Message Header: > ProtDisc 8 (0x8), CRV 125 (0x7d), CRVflag: 1 (0x1), MesType: 90 (0x5a) > 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse > error [-3012] [Q931E_INVALID_CRV] > 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() Sending frame > ----------------- Q.921 Packet [Outgoing] --------------- > SAPI: 0, TEI: 0, C/R: Response (1) > > Type: S Frame, SV: RR (Receive Ready) > P/F: 0, N(R): 83 [V(A): 81, V(R): 83, V(S): 81] > > Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] > ---------------------------------------------- > > 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() T200 stopped for TEI 0 > 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() T203 (timeout: 10000 msecs) > restarted for TEI 0 > 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() Sending frame > ----------------- Q.921 Packet [Outgoing] --------------- > SAPI: 0, TEI: 0, C/R: Command (0) > > Type: S Frame, SV: RR (Receive Ready) > P/F: 0, N(R): 83 [V(A): 81, V(R): 83, V(S): 81] > > Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] > ---------------------------------------------- > > > > > > Peter P GMX schrieb: >> Thanks Helmut, >> >> I cross-checked with our provider. They use national numbering plan for >> our lines. So this didn't solve our problem. >> I also ensured that the local language is DE and ZAP timing is dedicated >> to span 1. >> >> I changed the configs to debug mode for OpenZAP, so I hopefully will get >> some more info on the next failure. >> >> Best regards >> Peter >> >> Helmut Kuper schrieb: >> >>> Hi Peter, >>> >>> it was simply a change in our TDM Voice Switch. It used a different >>> numbering plan and we changed it to "national" to get it work with FS >>> and openzap in Q921/Q931 mode. >>> >>> What I still search is a way to configure the numberplan in FS. >>> >>> To make it clear: In my case it didn't work from the second FS starts >>> up. So this differs from your problem. >>> >>> >>> To get an idea what's going on on the TDM link I used a TDM D-Channel >>> monitoring device and traced the d-channel messages exchanged between FS >>> and TDM. That should make it easier to see what's wrong when the >>> problems occur. >>> But you can also increase FS debug level to debug and trace the Q921 >>> and Q931 messages in FS console via fs_cli during runtime. You have to >>> set this in openzap.conf.xml: >>> >>> >>> >>> >>> Unfortunately FS doesn't decode the whole Q931 messages, but it shows a >>> hex representation of the message, so you can manually decode it with >>> this documents: >>> >>> Q.931: http://www.itu.int/rec/T-REC-Q.931-199805-I/en >>> Q.921: http://www.itu.int/rec/T-REC-Q.921-199709-I/en >>> >>> >>> I think for numberingplan issues you only have to track the Q.931 >>> messages. >>> >>> >>> The last idea I have to get some light into your problem and to avoid >>> manually decoding, try to convert FS's q931 hexdump into wiresharks pcap >>> format. Wireshark should be able to decode it :) >>> http://wiki.wireshark.org/Q.931 >>> >>> Maybe it's a good idea to implement a wireshark export for those >>> messages in FS. This will make debugging easy and cheap. >>> >>> >>> >>> Hope it helps a bit. >>> >>> >>> regards >>> helmut >>> >>> Am 15.01.2009 12:06, schrieb Peter P GMX: >>> >>>> Helmut, >>>> >>>> can you give me a hint, how you worked around this? >>>> >>>> Best regards >>>> Peter >>>> >>>> Helmut Kuper schrieb: >>>> >>>>> Hi Michael, >>>>> >>>>> it must not be the case here, but I had the same error, when incomming >>>>> calles used a wrong numbering plan resp not the one, FS expected. >>>>> >>>>> Just a hint. >>>>> >>>>> regards >>>>> Helmut >>>>> >>>>> >>>>> Am 15.01.2009 09:20, schrieb Peter P GMX: >>>>> >>>>>> 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 at lists.freeswitch.org/msg05792.html >>> >>>>>> 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 >>>>>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ajlong at worldlink.net Mon Jan 19 10:25:09 2009 From: ajlong at worldlink.net (Adam Long) Date: Mon, 19 Jan 2009 13:25:09 -0500 Subject: [Freeswitch-users] ODBC in place of db/core.db In-Reply-To: <191c3a030901191004g5e63c8fbj60863c592ba515c8@mail.gmail.com> References: <008b01c97a5e$97f685c0$c7e39140$@net> <191c3a030901191004g5e63c8fbj60863c592ba515c8@mail.gmail.com> Message-ID: <00bd01c97a63$43309c60$c991d520$@net> OK, thanks for the heads up.. Reason I had asked in the first place, I was planning on tinkering with an auto-failover setup that would be capable of maintaining state and not dropping calls during failover. Has anyone had any success with this that you know of, or is it just not possible yet? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, January 19, 2009 1:04 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] ODBC in place of db/core.db it's not possible, the core is only sqlite. On Mon, Jan 19, 2009 at 11:51 AM, Adam Long wrote: Hi Guys, I have read the documentation on the wiki and have successfully compiled FreeSwtich with odbc core support. I am able to get my SIP Profiles and Voicemail databases to load, create, and utilize the ODBC database tables successfully. However, I cannot seem to figure out how to make FreeSwitch use the ODBC database instead of the SQL Lite db/core.db I'm sure it's just a matter of properly placing the following line in one of the configs. If anyone can point me in the right direction I would greatly appreciate it. Thank you! _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/fe13e885/attachment-0001.html From msc at freeswitch.org Mon Jan 19 10:25:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Jan 2009 10:25:47 -0800 Subject: [Freeswitch-users] OpenZap detect tones question In-Reply-To: <4974C448.4000501@novatex.com.au> References: <4971AA12.10706@novatex.com.au> <87f2f3b90901190952k1540de7ct51df55a799c596eb@mail.gmail.com> <4974C448.4000501@novatex.com.au> Message-ID: <87f2f3b90901191025r3c742f23g8eb13ef7f541270d@mail.gmail.com> Thanks. It might be a good idea to open a jira and assign it to yourself so that we can keep track of this issue. Let me know if you need any help. I'm the resident OpenZAP enthusiast so I'll be happy assist wherever I can. -MC (mercutioviz) On Mon, Jan 19, 2009 at 10:19 AM, Scott Ellis wrote: > No, I need to look through the code to work it out. Will get back to you > when I have. > > Scott > > Michael Collins wrote: > > Scott, > > Did you get the answer to this question yet? Either way, could you > post your openzap.conf and tones.conf files? I'd like to get this one > documented on the wiki so that non-US PSTN tones are properly > documented. > > -MC > > On Sat, Jan 17, 2009 at 1:51 AM, Scott Ellis > wrote: > > > Quick question, when specifying a "detect-busy" tone in the tones.conf > file - is the cadence used? (The US examples to not have cadence) > > In the tests I have does it does not seem to be. > > This is a problem in Australia, as we have managed to have our busy tone > 425Hz, 375ms on 375ms off, also in our dial tone 400+425+450. > > So when I go to dial a call, I often get the Zap channel hanging up > again as it thinks the line is busy. > > Scott > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From scott.ellis at novatex.com.au Mon Jan 19 10:27:07 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Tue, 20 Jan 2009 05:27:07 +1100 Subject: [Freeswitch-users] Interesting Problem with SPA 3000 Message-ID: <4974C5FB.3020509@novatex.com.au> I have an interesting problem with a set up using two (or more) SPA 3000's. I make a call out on the unit to a PSTN line - great. I then try and make another call out on that line - it fails and moves on to the next one in the bridge call statement. It then goes through on the second unit. Almost always 1:30 later, I get an inbound call from the PSTN showing up in FreeSwitch, which goes to an extension - and when answered bridges that extension onto the existing call. Now I am sure that this behaviour from the SPA is a little odd, but does anyone have any tips for dealing with it from the dialplan? Most obvious being a way to not call the first unit when it already is active... I do not have it defined as a gateway - just using sofia/internal/$1 at 10.0.0.18:5061 so make a call. Scott From mike at jerris.com Mon Jan 19 10:18:23 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jan 2009 13:18:23 -0500 Subject: [Freeswitch-users] ODBC in place of db/core.db In-Reply-To: <191c3a030901191004g5e63c8fbj60863c592ba515c8@mail.gmail.com> References: <008b01c97a5e$97f685c0$c7e39140$@net> <191c3a030901191004g5e63c8fbj60863c592ba515c8@mail.gmail.com> Message-ID: You can disable the core db using -nosql or if you want to remotely have access to an odbc db of calls you can make an event socket listener to make your own db. Mike On Jan 19, 2009, at 1:04 PM, "Anthony Minessale" wrote: > it's not possible, the core is only sqlite. > > > On Mon, Jan 19, 2009 at 11:51 AM, Adam Long > wrote: > Hi Guys, > > > > I have read the documentation on the wiki and have successfully > compiled FreeSwtich with odbc core support. > > I am able to get my SIP Profiles and Voicemail databases to load, > create, and utilize the ODBC database tables successfully. > > > > However, I cannot seem to figure out how to make FreeSwitch use the > ODBC database instead of the SQL Lite db/core.db > > > > I'm sure it's just a matter of properly placing the following line > in one of the configs? > > > > > > If anyone can point me in the right direction I would greatly > appreciate it. > > > > Thank you! > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/a91928e2/attachment.html From saigop at gmail.com Mon Jan 19 10:42:34 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Tue, 20 Jan 2009 00:12:34 +0530 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <497485CC.6060906@voiceworks.pl> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> <497485CC.6060906@voiceworks.pl> Message-ID: <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> How can I capture Q931 packets by separating the D channel and B channel? On Mon, Jan 19, 2009 at 7:23 PM, Michal Bielicki wrote: > It is both :) > > Helmut Kuper schrieb: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello Michael, > > I'm currently on my way to put all those shell and perl stuff into C > code. As soon as that works I try to put it in ozmod_isdn. So after that > FS will be able to enable Q931Pcapfile generation. After that I will > send the patch to FS. > > Hope this is ok for you. > > > What do you mean with "help with jira"? I thought it is just a bug > tracker... If it's also a feature request tool, I will feed it with > documentation about what I did. > > > regards > helmut > > Am 17.01.2009 21:44, schrieb Michael S Collins: > > > Guys this is awesome! Helmut, if you need any help with jira just let > me know. > -MC > > Sent from my iPhone > > On Jan 17, 2009, at 9:20 AM, Brian West wrote: > > > > Maybe open a jira with this info? Maybe it can all be done as a one > step process in the ozmod_isdn ;) > > /b > > On Jan 17, 2009, at 10:58 AM, Helmut Kuper wrote: > > > > Hi Brian, > > currently it's a simple perl script which greps all Q931 hexdumps > from > FS logfile converting them to a TPKT packet, and writing those to a > separate local file (wireshark's text2pcap file format). This 1st > step > is easy to put right into FS ozmod_isdn debug code. > The 2nd step is to convert the new file into a .pcap file with adding > TCP/IP dummy packet in front of each TPKT packet. This is done via > test2pcap. This .pcap file is ready to be decoded by wireshark. > > I put both steps into a little shell script and added a 3rd step to > get > those .pcap file emailed from the Server to my Desktop. > > regards > helmut > > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkl0PPgACgkQ4tZeNddg3dwKMACfYYHglospD7FJeY4Ne2Q8qBWJ > pNIAoKtEXQ+RnVg2ahySjd5zKfUcOowQ > =1abg > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/6ddd3eb7/attachment.html From msc at freeswitch.org Mon Jan 19 10:43:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Jan 2009 10:43:53 -0800 Subject: [Freeswitch-users] Interesting Problem with SPA 3000 In-Reply-To: <4974C5FB.3020509@novatex.com.au> References: <4974C5FB.3020509@novatex.com.au> Message-ID: <87f2f3b90901191043u633db77cg19749d023af1d15a@mail.gmail.com> Scott, Is it possible that your SPA is not properly detecting hangups? I'm wondering if there is a setting in the SPA3000 for your country. (Australia, no?) Please check your SPA settings and report back. Thanks, MC On Mon, Jan 19, 2009 at 10:27 AM, Scott Ellis wrote: > I have an interesting problem with a set up using two (or more) SPA 3000's. > > I make a call out on the unit to a PSTN line - great. > > I then try and make another call out on that line - it fails and moves > on to the next one in the bridge call statement. It then goes through on > the second unit. > > Almost always 1:30 later, I get an inbound call from the PSTN showing up > in FreeSwitch, which goes to an extension - and when answered bridges > that extension onto the existing call. > > Now I am sure that this behaviour from the SPA is a little odd, but does > anyone have any tips for dealing with it from the dialplan? Most obvious > being a way to not call the first unit when it already is active... > > I do not have it defined as a gateway - just using > sofia/internal/$1 at 10.0.0.18:5061 so make a call. > > Scott > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon Jan 19 10:46:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Jan 2009 10:46:46 -0800 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> <497485CC.6060906@voiceworks.pl> <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> Message-ID: <87f2f3b90901191046t6b8ac1e9hfa130e48f25a772b@mail.gmail.com> Until Helmut gets his script, etc. all set the only choice you have is to capture the debug output on the command line. -MC On Mon, Jan 19, 2009 at 10:42 AM, Gopalakrishnan A.N wrote: > How can I capture Q931 packets by separating the D channel and B channel? > > On Mon, Jan 19, 2009 at 7:23 PM, Michal Bielicki > wrote: >> >> It is both :) >> >> Helmut Kuper schrieb: >> >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >> Hello Michael, >> >> I'm currently on my way to put all those shell and perl stuff into C >> code. As soon as that works I try to put it in ozmod_isdn. So after that >> FS will be able to enable Q931Pcapfile generation. After that I will >> send the patch to FS. >> >> Hope this is ok for you. >> >> >> What do you mean with "help with jira"? I thought it is just a bug >> tracker... If it's also a feature request tool, I will feed it with >> documentation about what I did. >> >> >> regards >> helmut >> >> Am 17.01.2009 21:44, schrieb Michael S Collins: >> >> >> Guys this is awesome! Helmut, if you need any help with jira just let >> me know. >> -MC >> >> Sent from my iPhone >> >> On Jan 17, 2009, at 9:20 AM, Brian West wrote: >> >> >> >> Maybe open a jira with this info? Maybe it can all be done as a one >> step process in the ozmod_isdn ;) >> >> /b >> >> On Jan 17, 2009, at 10:58 AM, Helmut Kuper wrote: >> >> >> >> Hi Brian, >> >> currently it's a simple perl script which greps all Q931 hexdumps >> from >> FS logfile converting them to a TPKT packet, and writing those to a >> separate local file (wireshark's text2pcap file format). This 1st >> step >> is easy to put right into FS ozmod_isdn debug code. >> The 2nd step is to convert the new file into a .pcap file with adding >> TCP/IP dummy packet in front of each TPKT packet. This is done via >> test2pcap. This .pcap file is ready to be decoded by wireshark. >> >> I put both steps into a little shell script and added a 3rd step to >> get >> those .pcap file emailed from the Server to my Desktop. >> >> regards >> helmut >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -----BEGIN PGP SIGNATURE----- >> Version: GnuPG v1.4.9 (MingW32) >> >> iEYEARECAAYFAkl0PPgACgkQ4tZeNddg3dwKMACfYYHglospD7FJeY4Ne2Q8qBWJ >> pNIAoKtEXQ+RnVg2ahySjd5zKfUcOowQ >> =1abg >> -----END PGP SIGNATURE----- >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Thank you with regards, > Gopal, > PeopleTech Systems Private Limited > www.peopletech.co.in > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Mon Jan 19 10:47:28 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 19 Jan 2009 12:47:28 -0600 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> <497485CC.6060906@voiceworks.pl> <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> Message-ID: Well the D channel is the only channel where any Q931 should be present. If you happen to have it on a B channel something has gone horribly wrong! /b On Jan 19, 2009, at 12:42 PM, Gopalakrishnan A.N wrote: > How can I capture Q931 packets by separating the D channel and B > channel? From Prometheus001 at gmx.net Mon Jan 19 11:09:03 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 19 Jan 2009 20:09:03 +0100 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <87f2f3b90901191021y4e4c93e2vd97d64889711bf77@mail.gmail.com> References: <496E2493.2000207@gmx.net> <87f2f3b90901141307m3d4e0e60hce337ad36b121369@mail.gmail.com> <496EF1C2.8020607@gmx.net> <496F1449.6000407@ewetel.de> <496F18C3.3030807@gmx.net> <496F2C73.9030400@ewetel.de> <496F7413.704@gmx.net> <496F98B9.7040403@gmx.net> <87f2f3b90901191021y4e4c93e2vd97d64889711bf77@mail.gmail.com> Message-ID: <4974CFCF.4060400@gmx.net> Hello Michael, thanks for your response. We have now decided (although this is not the best solution) to put an Asterisk in front of FS who handles the Zap stuff and passes it via SIP to freeswitch. That way we got a stable zap configuration and the benefits of freeswitch, although our IVR application is now respoinding a bit slowly. I also do not have any developer with these skills in sight for fixing these issues. Anyway I hope that we will overcome these zap problems soon so that we can cme back to a pure FS installation. Best regards Peter Michael Collins schrieb: > Peter, > > I believe we are in a bit of a holding pattern right now with OpenZAP > PRI stuff. We have a super user, Stefan, who is working on some Q931 > timers and such but he is working on it in spare time and there's no > hard date. If you know someone with serious PRI and C programming > skillz who can assist then we'd definitely be willing to have some > help. "Patches welcome" as it were. :) > > Thanks, > MC > > On Thu, Jan 15, 2009 at 12:12 PM, Peter P GMX wrote: > >> I did some more tests. When I sequentially setup calls (only one >> simultaneous call at a time), it works for hundreds of calls. >> As soon as I setup 2 calls in parallel ist fails aber a number of calls. >> >> Please find another debug ouput (now with Q.921 debug also). >> The log starts with the latest hangup of a successfull call. After this >> one I receive a >> "2009-01-15 20:26:46 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received >> Release with no matching channel 0" >> and later >> "2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 >> parse error [-3012] [Q931E_INVALID_CRV]" >> >> Is there anyone to fix it? May I donate some money for fixing that? >> >> Best regards >> Peter >> >> >> Debug: >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame >> ----------------- Q.921 Packet [Outgoing] --------------- >> SAPI: 0, TEI: 0, C/R: Command (0) >> >> Type: S Frame, SV: RR (Receive Ready) >> P/F: 0, N(R): 81 [V(A): 80, V(R): 81, V(S): 80] >> >> Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] >> ---------------------------------------------- >> >> 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:813 state_advance() 2:3 STATE >> [TERMINATING] >> 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:1121 state_advance() >> Terminating: Direction Inbound >> 2009-01-15 20:26:44 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() >> got clear channel sig [STOP] >> 2009-01-15 20:26:44 [NOTICE] mod_openzap.c:1437 >> on_clear_channel_signal() Hangup OpenZAP/2:3/21658519 [CS_EXECUTE] >> [NORMAL_CLEARING] >> 2009-01-15 20:26:44 [DEBUG] switch_channel.c:1513 >> switch_channel_perform_hangup() Send signal OpenZAP/2:3/21658519 [KILL] >> 2009-01-15 20:26:44 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal >> OpenZAP/2:3/21658519 [BREAK] >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Receiving message from Layer4 >> (size: 184, type: 77) >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Sending message to Q.921 >> (size: 184) >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Creating Q.931 Message Header: >> ProtDisc 8 (0x8), CRV 126 (0x7e), CRVflag: 1 (0x1), MesType: 77 (0x4d) >> 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 >> -------------------------------------------------------------------------------- >> [08 02 80 7e 4d] >> >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Got frame from Q.931, type: >> 4, tei: 0, size: 9 >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Enqueueing I frame for TEI 0 [0] >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame >> ----------------- Q.921 Packet [Outgoing] --------------- >> SAPI: 0, TEI: 0, C/R: Command (0) >> >> Type: I Frame >> P/F: 0, N(S): 80, N(R): 81 [V(A): 80, V(R): 81, V(S): 80] >> >> Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] >> ---------------------------------------------- >> >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T200 (timeout: 1000 msecs) >> started for TEI 0 >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T203 stopped for TEI 0 >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Q931Rx43 return code: 1 >> 2009-01-15 20:26:44 [DEBUG] mod_event_socket.c:1922 listener_run() >> Session complete, waiting for children >> 2009-01-15 20:26:44 [DEBUG] mod_event_socket.c:1946 listener_run() >> Connection Closed >> 2009-01-15 20:26:44 [DEBUG] switch_ivr_play_say.c:1222 >> switch_ivr_play_file() done playing file >> 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:454 >> switch_core_session_run() (OpenZAP/2:3/21658519) State EXECUTE going to >> sleep >> 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/2:3/21658519) Running State Change >> CS_HANGUP >> 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/2:3/21658519) State HANGUP >> 2009-01-15 20:26:44 [DEBUG] mod_openzap.c:472 channel_on_hangup() >> OpenZAP/2:3/21658519 CHANNEL HANGUP >> 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() OpenZAP/2:3/21658519 Standard HANGUP, >> cause: NORMAL_CLEARING >> 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/2:3/21658519) State HANGUP going to sleep >> 2009-01-15 20:26:44 [DEBUG] switch_core_session.c:939 >> switch_core_session_thread() Session 251 (OpenZAP/2:3/21658519) Locked, >> Waiting on external entities >> 2009-01-15 20:26:44 [NOTICE] switch_core_session.c:957 >> switch_core_session_thread() Session 251 (OpenZAP/2:3/21658519) Ended >> 2009-01-15 20:26:44 [NOTICE] switch_core_session.c:959 >> switch_core_session_thread() Close Channel OpenZAP/2:3/21658519 [CS_HANGUP] >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() New packet received (4 bytes) >> ----------------- Q.921 Packet [Incoming] --------------- >> SAPI: 0, TEI: 0, C/R: Response (0) >> >> Type: S Frame, SV: RR (Receive Ready) >> P/F: 0, N(R): 81 [V(A): 80, V(R): 81, V(S): 81] >> >> Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] >> ---------------------------------------------- >> >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T200 stopped for TEI 0 >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T203 (timeout: 10000 msecs) >> restarted for TEI 0 >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() New packet received (9 bytes) >> ----------------- Q.921 Packet [Incoming] --------------- >> SAPI: 0, TEI: 0, C/R: Command (1) >> >> Type: I Frame >> P/F: 0, N(S): 81, N(R): 81 [V(A): 81, V(R): 81, V(S): 81] >> >> Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] >> ---------------------------------------------- >> >> 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 >> -------------------------------------------------------------------------------- >> [08 02 00 7e 5a] >> >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Received message from Q.921 >> (ind 4, tei 0, size 9) >> MesType: 90, CRVFlag 0 (Originator), CRV 126 (Dialect: 0) >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Sending message to Layer4 >> (size: 103) >> 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got >> an event! Type:[5a] Size:[103] CRV: 126 (0x7e, CTX: Originator) >> 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan >> e0020270 (2:3) source isdn_data->channels_remote_crv[0x7e] >> 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:418 zap_isdn_931_34() Changing >> state on 2:3 from TERMINATING to DOWN >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame >> ----------------- Q.921 Packet [Outgoing] --------------- >> SAPI: 0, TEI: 0, C/R: Response (1) >> >> Type: S Frame, SV: RR (Receive Ready) >> P/F: 0, N(R): 82 [V(A): 81, V(R): 82, V(S): 81] >> >> Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] >> ---------------------------------------------- >> >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T200 stopped for TEI 0 >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T203 (timeout: 10000 msecs) >> restarted for TEI 0 >> 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame >> ----------------- Q.921 Packet [Outgoing] --------------- >> SAPI: 0, TEI: 0, C/R: Command (0) >> >> Type: S Frame, SV: RR (Receive Ready) >> P/F: 0, N(R): 82 [V(A): 81, V(R): 82, V(S): 81] >> >> Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] >> ---------------------------------------------- >> >> 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:813 state_advance() 2:3 STATE >> [DOWN] >> 2009-01-15 20:26:44 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done 2:3 >> 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() New packet received (16 bytes) >> ----------------- Q.921 Packet [Incoming] --------------- >> SAPI: 0, TEI: 0, C/R: Command (1) >> >> Type: I Frame >> P/F: 0, N(S): 82, N(R): 81 [V(A): 81, V(R): 82, V(S): 81] >> >> Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] >> ---------------------------------------------- >> >> 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 12 >> -------------------------------------------------------------------------------- >> [08 02 00 7d 4d 08 05 82 e6 33 30 33] >> >> 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Received message from Q.921 >> (ind 4, tei 0, size 16) >> MesType: 77, CRVFlag 0 (Originator), CRV 125 (Dialect: 0) >> 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Sending message to Layer4 >> (size: 110) >> 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got >> an event! Type:[4d] Size:[110] CRV: 125 (0x7d, CTX: Originator) >> 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 >> (-1:-1) source isdn_data->channels_remote_crv[0x7d] >> 2009-01-15 20:26:46 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received >> Release with no matching channel 0 >> 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Sending message to Q.921 >> (size: 110) >> 2009-01-15 20:26:46 [DEBUG] Span:0 Q.931() Creating Q.931 Message Header: >> ProtDisc 8 (0x8), CRV 125 (0x7d), CRVflag: 1 (0x1), MesType: 90 (0x5a) >> 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse >> error [-3012] [Q931E_INVALID_CRV] >> 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() Sending frame >> ----------------- Q.921 Packet [Outgoing] --------------- >> SAPI: 0, TEI: 0, C/R: Response (1) >> >> Type: S Frame, SV: RR (Receive Ready) >> P/F: 0, N(R): 83 [V(A): 81, V(R): 83, V(S): 81] >> >> Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] >> ---------------------------------------------- >> >> 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() T200 stopped for TEI 0 >> 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() T203 (timeout: 10000 msecs) >> restarted for TEI 0 >> 2009-01-15 20:26:46 [DEBUG] Span:0 Q.921() Sending frame >> ----------------- Q.921 Packet [Outgoing] --------------- >> SAPI: 0, TEI: 0, C/R: Command (0) >> >> Type: S Frame, SV: RR (Receive Ready) >> P/F: 0, N(R): 83 [V(A): 81, V(R): 83, V(S): 81] >> >> Q.921 state: "Multiple Frame Mode Established" (7) [flags: ----] >> ---------------------------------------------- >> >> >> >> >> >> Peter P GMX schrieb: >> >>> Thanks Helmut, >>> >>> I cross-checked with our provider. They use national numbering plan for >>> our lines. So this didn't solve our problem. >>> I also ensured that the local language is DE and ZAP timing is dedicated >>> to span 1. >>> >>> I changed the configs to debug mode for OpenZAP, so I hopefully will get >>> some more info on the next failure. >>> >>> Best regards >>> Peter >>> >>> Helmut Kuper schrieb: >>> >>> >>>> Hi Peter, >>>> >>>> it was simply a change in our TDM Voice Switch. It used a different >>>> numbering plan and we changed it to "national" to get it work with FS >>>> and openzap in Q921/Q931 mode. >>>> >>>> What I still search is a way to configure the numberplan in FS. >>>> >>>> To make it clear: In my case it didn't work from the second FS starts >>>> up. So this differs from your problem. >>>> >>>> >>>> To get an idea what's going on on the TDM link I used a TDM D-Channel >>>> monitoring device and traced the d-channel messages exchanged between FS >>>> and TDM. That should make it easier to see what's wrong when the >>>> problems occur. >>>> But you can also increase FS debug level to debug and trace the Q921 >>>> and Q931 messages in FS console via fs_cli during runtime. You have to >>>> set this in openzap.conf.xml: >>>> >>>> >>>> >>>> >>>> Unfortunately FS doesn't decode the whole Q931 messages, but it shows a >>>> hex representation of the message, so you can manually decode it with >>>> this documents: >>>> >>>> Q.931: http://www.itu.int/rec/T-REC-Q.931-199805-I/en >>>> Q.921: http://www.itu.int/rec/T-REC-Q.921-199709-I/en >>>> >>>> >>>> I think for numberingplan issues you only have to track the Q.931 >>>> messages. >>>> >>>> >>>> The last idea I have to get some light into your problem and to avoid >>>> manually decoding, try to convert FS's q931 hexdump into wiresharks pcap >>>> format. Wireshark should be able to decode it :) >>>> http://wiki.wireshark.org/Q.931 >>>> >>>> Maybe it's a good idea to implement a wireshark export for those >>>> messages in FS. This will make debugging easy and cheap. >>>> >>>> >>>> >>>> Hope it helps a bit. >>>> >>>> >>>> regards >>>> helmut >>>> >>>> Am 15.01.2009 12:06, schrieb Peter P GMX: >>>> >>>> >>>>> Helmut, >>>>> >>>>> can you give me a hint, how you worked around this? >>>>> >>>>> Best regards >>>>> Peter >>>>> >>>>> Helmut Kuper schrieb: >>>>> >>>>> >>>>>> Hi Michael, >>>>>> >>>>>> it must not be the case here, but I had the same error, when incomming >>>>>> calles used a wrong numbering plan resp not the one, FS expected. >>>>>> >>>>>> Just a hint. >>>>>> >>>>>> regards >>>>>> Helmut >>>>>> >>>>>> >>>>>> Am 15.01.2009 09:20, schrieb Peter P GMX: >>>>>> >>>>>> >>>>>>> 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 at lists.freeswitch.org/msg05792.html >>>> >>>> >>>>>>> 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 >>>>>>>>> >>>>>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Mon Jan 19 11:16:33 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 19 Jan 2009 13:16:33 -0600 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <4974CFCF.4060400@gmx.net> References: <496E2493.2000207@gmx.net> <87f2f3b90901141307m3d4e0e60hce337ad36b121369@mail.gmail.com> <496EF1C2.8020607@gmx.net> <496F1449.6000407@ewetel.de> <496F18C3.3030807@gmx.net> <496F2C73.9030400@ewetel.de> <496F7413.704@gmx.net> <496F98B9.7040403@gmx.net> <87f2f3b90901191021y4e4c93e2vd97d64889711bf77@mail.gmail.com> <4974CFCF.4060400@gmx.net> Message-ID: <743E9AD4-F038-4E27-A4FC-B390C7E88975@freeswitch.org> Your company could always sponsor some of the time and effort to improve OpenZAP to meet your needs. Its also a good way to give back to the project. /b On Jan 19, 2009, at 1:09 PM, Peter P GMX wrote: > I also do not have any developer with these skills in sight for fixing > these issues. Anyway I hope that we will overcome these zap problems > soon so that we can cme back to a pure FS installation. > > Best regards > Peter From scott.ellis at novatex.com.au Mon Jan 19 11:21:27 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Tue, 20 Jan 2009 06:21:27 +1100 Subject: [Freeswitch-users] Interesting Problem with SPA 3000 In-Reply-To: <87f2f3b90901191043u633db77cg19749d023af1d15a@mail.gmail.com> References: <4974C5FB.3020509@novatex.com.au> <87f2f3b90901191043u633db77cg19749d023af1d15a@mail.gmail.com> Message-ID: <4974D2B7.5030200@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/242bbbcb/attachment-0001.html From msc at freeswitch.org Mon Jan 19 12:37:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Jan 2009 12:37:26 -0800 Subject: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] In-Reply-To: <743E9AD4-F038-4E27-A4FC-B390C7E88975@freeswitch.org> References: <496E2493.2000207@gmx.net> <496EF1C2.8020607@gmx.net> <496F1449.6000407@ewetel.de> <496F18C3.3030807@gmx.net> <496F2C73.9030400@ewetel.de> <496F7413.704@gmx.net> <496F98B9.7040403@gmx.net> <87f2f3b90901191021y4e4c93e2vd97d64889711bf77@mail.gmail.com> <4974CFCF.4060400@gmx.net> <743E9AD4-F038-4E27-A4FC-B390C7E88975@freeswitch.org> Message-ID: <87f2f3b90901191237i6c0566eek7253222331b5cdaf@mail.gmail.com> On Mon, Jan 19, 2009 at 11:16 AM, Brian West wrote: > Your company could always sponsor some of the time and effort to > improve OpenZAP to meet your needs. Its also a good way to give back > to the project. In this case you'd want to contact the FreeSWITCH team directly at consulting at freeswitch.org so that you could talk specifics. -MC > > /b > > On Jan 19, 2009, at 1:09 PM, Peter P GMX wrote: > >> I also do not have any developer with these skills in sight for fixing >> these issues. Anyway I hope that we will overcome these zap problems >> soon so that we can cme back to a pure FS installation. >> >> Best regards >> Peter > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon Jan 19 12:51:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Jan 2009 12:51:44 -0800 Subject: [Freeswitch-users] Interesting Problem with SPA 3000 In-Reply-To: <4974D2B7.5030200@novatex.com.au> References: <4974C5FB.3020509@novatex.com.au> <87f2f3b90901191043u633db77cg19749d023af1d15a@mail.gmail.com> <4974D2B7.5030200@novatex.com.au> Message-ID: <87f2f3b90901191251x57323db8r4037d5a4e0d859b2@mail.gmail.com> Scott, I think you're at the point where you'll need to collect more info and start a pastebin or two. I'm actually working on the checklist and how-to for reporting bugs and troubleshooting. You can be my first test case!:) Please visit here and start collecting data: http://wiki.freeswitch.org/wiki/Reporting_Bugs Let me know if any of my instructions seem wrong or confusing or just plain don't work. Thanks, MC (mercutioviz) On Mon, Jan 19, 2009 at 11:21 AM, Scott Ellis wrote: > No the hang ups seem to be ok. It seems to be that after FS tries to make a > second call, 1:30 later I get the phantom inbound call. If I do not try and > make that second call everything works fine. I think I can make a dial plan > to manage the outbound calls, but the same thing happens if the first call > is inbound PSTN, then we make outbound. > So I need to have a db variable to indicate "device in use" effectively so > the call attempt is not made. > > Scott > > The second outbound being made > 2009-01-20 06:07:03 [NOTICE] switch_channel.c:565 switch_channel_set_name() > New Channel sofia/internal/43517488 at 10.0.0.18:5061 > [5b9d6822-e65c-11dd-b2e1-993799172013] > 2009-01-20 06:07:03 [NOTICE] sofia.c:3124 sofia_handle_sip_i_state() Hangup > sofia/internal/43517488 at 10.0.0.18:5061 [CS_CONSUME_MEDIA] > [RECOVERY_ON_TIMER_EXPIRE] > 2009-01-20 06:07:04 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 8 > (sofia/internal/43517488 at 10.0.0.18:5061) Ended > 2009-01-20 06:07:04 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel > sofia/internal/43517488 at 10.0.0.18:5061 [CS_HANGUP] > 2009-01-20 06:07:04 [NOTICE] switch_channel.c:565 switch_channel_set_name() > New Channel sofia/internal/43517488 at 10.0.0.17:5061 > [5babe9ba-e65c-11dd-b2e1-993799172013] > > The phantom inbound, goes to 500 is not answered, and then the original call > above is closed. > 2009-01-20 06:07:42 [NOTICE] mod_dptools.c:600 answer_function() Channel > [sofia/internal/Line_8 at 10.0.0.9] has been answered > 2009-01-20 06:07:42 [NOTICE] switch_channel.c:565 switch_channel_set_name() > New Channel sofia/internal/500 [726fc392-e65c-11dd-b2e1-993799172013] > 2009-01-20 06:07:42 [NOTICE] sofia.c:3124 sofia_handle_sip_i_state() Hangup > sofia/internal/500 [CS_CONSUME_MEDIA] [USER_BUSY] > 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 11 (sofia/internal/500) Ended > 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel sofia/internal/500 [CS_HANGUP] > 2009-01-20 06:07:42 [INFO] mod_dptools.c:1984 audio_bridge_function() > Originate Failed. Cause: USER_BUSY > 2009-01-20 06:07:42 [NOTICE] mod_dptools.c:2011 audio_bridge_function() > Hangup sofia/internal/Line_8 at 10.0.0.9 [CS_EXECUTE] [USER_BUSY] > 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 10 (sofia/internal/Line_8 at 10.0.0.9) > Ended > 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel sofia/internal/Line_8 at 10.0.0.9 > [CS_HANGUP] > 2009-01-20 06:11:02 [NOTICE] mod_sofia.c:681 sofia_read_frame() Hangup > sofia/internal/49633633 at 10.0.0.18:5061 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] > 2009-01-20 06:11:02 [NOTICE] switch_ivr_bridge.c:955 > switch_ivr_multi_threaded_bridge() Hangup sofia/internal/402 at 10.0.0.9 > [CS_EXECUTE] [MEDIA_TIMEOUT] > 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 5 (sofia/internal/402 at 10.0.0.9) Ended > 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel sofia/internal/402 at 10.0.0.9 > [CS_HANGUP] > 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 6 > (sofia/internal/49633633 at 10.0.0.18:5061) Ended > 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel > sofia/internal/49633633 at 10.0.0.18:5061 [CS_HANGUP] > > Michael Collins wrote: > > Scott, > > Is it possible that your SPA is not properly detecting hangups? I'm > wondering if there is a setting in the SPA3000 for your country. > (Australia, no?) Please check your SPA settings and report back. > Thanks, > MC > > On Mon, Jan 19, 2009 at 10:27 AM, Scott Ellis > wrote: > > > I have an interesting problem with a set up using two (or more) SPA 3000's. > > I make a call out on the unit to a PSTN line - great. > > I then try and make another call out on that line - it fails and moves > on to the next one in the bridge call statement. It then goes through on > the second unit. > > Almost always 1:30 later, I get an inbound call from the PSTN showing up > in FreeSwitch, which goes to an extension - and when answered bridges > that extension onto the existing call. > > Now I am sure that this behaviour from the SPA is a little odd, but does > anyone have any tips for dealing with it from the dialplan? Most obvious > being a way to not call the first unit when it already is active... > > I do not have it defined as a gateway - just using > sofia/internal/$1 at 10.0.0.18:5061 so make a call. > > Scott > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From scott.ellis at novatex.com.au Mon Jan 19 12:55:09 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Tue, 20 Jan 2009 07:55:09 +1100 Subject: [Freeswitch-users] Is anyone our there using an SPA 3000? (or 3102) Message-ID: <4974E8AD.5070309@novatex.com.au> If so, could you please share your set up? directory files, and dial plan details (gateway details if configured this way)? Scott From pablosaro at gmail.com Mon Jan 19 13:03:20 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Mon, 19 Jan 2009 19:03:20 -0200 Subject: [Freeswitch-users] Conference auto-record not working In-Reply-To: <247f8100901190759n2deab0c6tc7aa07af4654bdb7@mail.gmail.com> References: <247f8100901190649g4086b3d8x7cbedf6a87be3ff2@mail.gmail.com> <247f8100901190716w3c19cdadi63cf9a878fb23e64@mail.gmail.com> <247f8100901190759n2deab0c6tc7aa07af4654bdb7@mail.gmail.com> Message-ID: <247f8100901191303u6ebd4995t573837d8e93e7978@mail.gmail.com> Hi there, I've checked out rev 11279 and it worked out of the box! Thanks Brian for your help. Regards, Pablo On Mon, Jan 19, 2009 at 1:59 PM, Pablo Hernan Saro wrote: > Thanks again. Then, I'll try the latest trunk. > > Pablo > > On Mon, Jan 19, 2009 at 1:22 PM, Brian West wrote: >> >> Please try the latest Trunk... Even 1.0.2 has bugs that are already >> fixed in trunk. Expect releases more often. Btw we never call a >> release stable. >> >> /b >> >> On Jan 19, 2009, at 9:16 AM, Pablo Hernan Saro wrote: >> >> > Thanks Brian! I'll try the latest stable. >> > Have a nice day >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > From ajlong at worldlink.net Mon Jan 19 14:03:27 2009 From: ajlong at worldlink.net (Adam Long) Date: Mon, 19 Jan 2009 17:03:27 -0500 Subject: [Freeswitch-users] Gateways without Registration or User/Pass Authentication Message-ID: <00f601c97a81$c1fc0a20$45f41e60$@net> I am trying to setup a couple of gateways on my external profile that do not user usernames or passwords . How can I setup these gateways to match based on originating IP address? At the moment I have them defined in separate xml files in the sip_profiles/external directory. I am using the default external.xml I would like to be able to dial via this gateway and receive calls from these gateways. For example NYGW1.xml ------- > When I remove the username/password params or set them to "" FreeSWITCH refuses to load the gateway. I tried adding the cidr="" to the gateway but no luck. Really all I want is for calls flowing in from 10.10.10.1 --- > FreeSWITCH External SIP Profile ---- > Match to NYGW1 --- > Dump into a context or extension from where I can route them as I please. Kind of how an Asterisk sip "peer" is matched on IP and routed to a context. Any ideas greatly appreciated! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/c8794c2b/attachment.html From mszlazak at aol.com Mon Jan 19 14:11:41 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 19 Jan 2009 17:11:41 -0500 Subject: [Freeswitch-users] Is anyone our there using an SPA 3000? (or 3102) In-Reply-To: <4974E8AD.5070309@novatex.com.au> References: <4974E8AD.5070309@novatex.com.au> Message-ID: <8CB489A0A60792A-920-471@WEBMAIL-DG04.sim.aol.com> I'm using and SPA3102 at home to connect one of the phones to my PC. Nothing special to FS directory files. I just copied one of the defaults and changed the file name to "line1" since I use "line1" as the ID under "subscriber information" in 3102's configuration. ? ??? ????? ????? ??? ??? ????? ????? ????? ????? ????? ????? ????? ????? ??? ? 3102 has a static IP address. Line 1 has proxy set to FreeSwitch IP. 3102's "line 1" tab has dial plan: (xx.<:@gw0>) PSTN line dial plans are for line 1: ?(<:2007>S0) and the rest are: (xx.) then Line 1 VoIP Caller DP: 2 VoIP Caller Default DP:1 -----Original Message----- From: Scott Ellis To: freeswitch-users at lists.freeswitch.org Sent: Mon, 19 Jan 2009 12:55 pm Subject: [Freeswitch-users] Is anyone our there using an SPA 3000? (or 3102) If so, could you please share your set up? directory files, and dial plan details (gateway details if configured this way)? Scott _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/fe8dee7b/attachment-0001.html From scott.ellis at novatex.com.au Mon Jan 19 15:21:58 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Tue, 20 Jan 2009 10:21:58 +1100 Subject: [Freeswitch-users] Interesting Problem with SPA 3000 In-Reply-To: <87f2f3b90901191251x57323db8r4037d5a4e0d859b2@mail.gmail.com> References: <4974C5FB.3020509@novatex.com.au> <87f2f3b90901191043u633db77cg19749d023af1d15a@mail.gmail.com> <4974D2B7.5030200@novatex.com.au> <87f2f3b90901191251x57323db8r4037d5a4e0d859b2@mail.gmail.com> Message-ID: <49750B16.2040606@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/fc5c0ed3/attachment.html From josephbajin at gmail.com Mon Jan 19 21:32:27 2009 From: josephbajin at gmail.com (Joseph Bajin) Date: Tue, 20 Jan 2009 00:32:27 -0500 Subject: [Freeswitch-users] Gateways without Registration or User/Pass Authentication In-Reply-To: <00f601c97a81$c1fc0a20$45f41e60$@net> References: <00f601c97a81$c1fc0a20$45f41e60$@net> Message-ID: <1dce11f20901192132s77d6cf7ftc912ac078f66662c@mail.gmail.com> You don't need to define a gateway if you are trying to limit incoming calls by a particular IP. You can do it two ways, first by using an ACL to only allow that traffic. Or you can create a dialplan that looks for that particular IP and then sends it to a particular context based on that IP. There are great examples on the wiki. On Mon, Jan 19, 2009 at 5:03 PM, Adam Long wrote: > I am trying to setup a couple of gateways on my external profile that do > not user usernames or passwords . > > How can I setup these gateways to match based on originating IP address? > > At the moment I have them defined in separate xml files in the > sip_profiles/external directory. > > > > I am using the default external.xml > > I would like to be able to dial via this gateway and receive calls from > these gateways. > > > > For example NYGW1.xml ------- > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > When I remove the username/password params or set them to "" FreeSWITCH > refuses to load the gateway. > > I tried adding the cidr="" to the gateway but no luck. > > > > Really all I want is for calls flowing in from 10.10.10.1 --- > FreeSWITCH > External SIP Profile ---- > Match to NYGW1 --- > Dump into a context or > extension from where I can route them as I please. > > > > Kind of how an Asterisk sip "peer" is matched on IP and routed to a > context. > > > > Any ideas greatly appreciated! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/1b913be2/attachment.html From josephbajin at gmail.com Mon Jan 19 21:33:45 2009 From: josephbajin at gmail.com (Joseph Bajin) Date: Tue, 20 Jan 2009 00:33:45 -0500 Subject: [Freeswitch-users] ODBC in place of db/core.db In-Reply-To: References: <008b01c97a5e$97f685c0$c7e39140$@net> <191c3a030901191004g5e63c8fbj60863c592ba515c8@mail.gmail.com> Message-ID: <1dce11f20901192133ka043ebarf84a06811650476d@mail.gmail.com> But that won't accomplish setting up any sort of Failover right? You'll have access to the db, but it would do no good correct... 2009/1/19 Michael Jerris > You can disable the core db using -nosql or if you want to remotely have > access to an odbc db of calls you can make an event socket listener to make > your own db. > > Mike > > > On Jan 19, 2009, at 1:04 PM, "Anthony Minessale" < > anthony.minessale at gmail.com> wrote: > > it's not possible, the core is only sqlite. > > > On Mon, Jan 19, 2009 at 11:51 AM, Adam Long < > ajlong at worldlink.net> wrote: > >> Hi Guys, >> >> >> >> I have read the documentation on the wiki and have successfully compiled >> FreeSwtich with odbc core support. >> >> I am able to get my SIP Profiles and Voicemail databases to load, create, >> and utilize the ODBC database tables successfully. >> >> >> >> However, I cannot seem to figure out how to make FreeSwitch use the ODBC >> database instead of the SQL Lite db/core.db >> >> >> >> I'm sure it's just a matter of properly placing the following line in one >> of the configs? >> >> >> >> >> >> If anyone can point me in the right direction I would greatly appreciate >> it. >> >> >> >> Thank you! >> >> _______________________________________________ >> Freeswitch-users mailing list >> >> Freeswitch-users at 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 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 > iax: > guest at conference.freeswitch.org/888 > googletalk: > conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- --Joe "Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit." --Fred Franzia (The famous wine guy) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/c8a3404b/attachment-0001.html From krice at suspicious.org Mon Jan 19 21:44:49 2009 From: krice at suspicious.org (Ken Rice) Date: Mon, 19 Jan 2009 23:44:49 -0600 Subject: [Freeswitch-users] ODBC in place of db/core.db In-Reply-To: <1dce11f20901192133ka043ebarf84a06811650476d@mail.gmail.com> Message-ID: Even if you could put the core DB into an external RDBMS that still wouldn?t get you failover... The core DB is not really where the call states are tracked that?s just there to make things like getting status outputs easier to keep from having to lock memory everytime someone does a show channels or show calls or whatever From: Joseph Bajin Reply-To: Date: Tue, 20 Jan 2009 00:33:45 -0500 To: Subject: Re: [Freeswitch-users] ODBC in place of db/core.db But that won't accomplish setting up any sort of Failover right? You'll have access to the db, but it would do no good correct... 2009/1/19 Michael Jerris > You can disable the core db using -nosql or if you want to remotely have > access to an odbc db of calls you can make an event socket listener to make > your own db. > > Mike > > > On Jan 19, 2009, at 1:04 PM, "Anthony Minessale" > wrote: > >> it's not possible, the core is only sqlite. >> >> >> On Mon, Jan 19, 2009 at 11:51 AM, Adam Long < >> ajlong at worldlink.net> wrote: >>> Hi Guys, >>> >>> >>> >>> I have read the documentation on the wiki and have successfully compiled >>> FreeSwtich with odbc core support. >>> >>> I am able to get my SIP Profiles and Voicemail databases to load, create, >>> and utilize the ODBC database tables successfully. >>> >>> >>> >>> However, I cannot seem to figure out how to make FreeSwitch use the ODBC >>> database instead of the SQL Lite db/core.db >>> >>> >>> >>> I'm sure it's just a matter of properly placing the following line in one of >>> the configs? >>> >>> >>> >>> >>> >>> If anyone can point me in the right direction I would greatly appreciate it. >>> >>> >>> >>> Thank you! >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> >>> Freeswitch-users at 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 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 >>> iax: >>> guest at conference.freeswitch.org/888 >>> googletalk: >>> conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> --Joe >>>> >>>> "Success is easy if you think of it like Rust: It's inevitable if you >>>> keep at it. Guys claim there are magic moments, but that's just bullshit." >>>> --Fred Franzia (The famous wine guy) >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090119/9c1f341d/attachment.html From juanbackson at gmail.com Mon Jan 19 23:56:56 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 20 Jan 2009 15:56:56 +0800 Subject: [Freeswitch-users] Freeswitch not sending out 183 Message-ID: <27c25bc40901192356j553f8bc3o1ae5295a30c0e4c7@mail.gmail.com> Hi, I am running some continuous testing hitting FS. There are a couple errors that gets popped up occasionally and I am trying to find out why. In one of the trace, I am seeing FS not sending 183. The weird thing is that this problem is not happening everything, but on a very rarely basis. Also, there is no error message being sent. Does anyone know under what situation would freeswitch not sending out 183? recv 771 bytes from udp/[192.168.1.122]:5060 at 14:28:50.460222: ------------------------------------------------------------------------ INVITE sip:0019008 at 192.168.1.116:5070 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bKe6e9.09c88aa1.0 Via: SIP/2.0/UDP 192.168.1.6:7001 From: 19008 ;tag=9 To: 0019008 > Call-ID: 9-10894 at 192.168.1.6 CSeq: 2 INVITE Contact: Max-Forwards: 69 User-Agent: Performance Test Content-Type: application/sdp Content-Length: 272 P-hint: inbound->inbound v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.6 s=- t=0 0 c=IN IP4 192.168.1.6 m=audio 6032 RTP/AVP 0 9 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ send 354 bytes to udp/[192.168.1.122]:5060 at 14:28:50.460805: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bKe6e9.09c88aa1.0 Via: SIP/2.0/UDP 192.168.1.6:7001 Record-Route: From: 19008 ;tag=9 To: 0019008 > Call-ID: 9-10894 at 192.168.1.6 CSeq: 2 INVITE User-Agent: Freeswitch Media Gateway Content-Length: 0 ------------------------------------------------------------------------ 2009-01-20 09:28:50 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/19008 at 192.168.1.122:7001[a7f0594e-e6fe-11dd-9a2c-fbbebdd2887f] 2009-01-20 09:28:50 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 19008->0019008 in context public 2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1653 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 1 execute_extension::a_record XML features 2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1653 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 2 execute_extension::a_stoprecord XML features 2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1653 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 3 execute_extension::a_att_xfer XML features 2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1660 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::b_record XML features 2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1660 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 execute_extension::b_stoprecord XML features 2009-01-20 09:28:50 [INFO] switch_ivr_async.c:1660 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::b_att_xfer XML features 2009-01-20 09:28:50 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/19008 [a816c7fa-e6fe-11dd-9a2c-fbbebdd2887f] send 1229 bytes to udp/[192.168.1.122]:5060 at 14:28:50.715136: ------------------------------------------------------------------------ INVITE sip:19008 at 192.168.1.122:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:5070;rport;branch=z9hG4bKKe15F6Dap2jBa Max-Forwards: 68 From: "19008" ;tag=mB8gaveZvmF8K To: Call-ID: 7f75543a-61a1-122c-8282-001517871e28 CSeq: 110112529 INVITE Contact: User-Agent: Freeswitch Media Gateway Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 335 P-hint: inbound->inbound Remote-Party-ID: "19008" ;screen=yes;privacy=off v=0 o=FreeSWITCH 5551695261821560906 6736752020354058642 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12016 RTP/AVP 0 9 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ recv 354 bytes from udp/[192.168.1.122]:5060 at 14:28:50.745112: ------------------------------------------------------------------------ SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.1.116:5070 ;rport=5070;branch=z9hG4bKKe15F6Dap2jBa From: "19008" ;tag=mB8gaveZvmF8K To: Call-ID: 7f75543a-61a1-122c-8282-001517871e28 CSeq: 110112529 INVITE Server: OpenSIPS (1.4.3-notls (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ recv 438 bytes from udp/[192.168.1.122]:5060 at 14:28:50.748929: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.116:5070 ;received=192.168.1.116;rport=5070;branch=z9hG4bKKe15F6Dap2jBa Record-Route: From: "19008" ;tag=mB8gaveZvmF8K To: ;tag=9 Call-ID: 7f75543a-61a1-122c-8282-001517871e28 CSeq: 110112529 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ recv 740 bytes from udp/[192.168.1.122]:5060 at 14:28:50.749130: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.116:5070 ;received=192.168.1.116;rport=5070;branch=z9hG4bKKe15F6Dap2jBa Record-Route: From: "19008" ;tag=mB8gaveZvmF8K To: ;tag=9 Call-ID: 7f75543a-61a1-122c-8282-001517871e28 CSeq: 110112529 INVITE Contact: Content-Type: application/sdp Content-Length: 272 2009-01-20 09:28:50 [NOTICE] sofia.c:2627 sofia_handle_sip_i_state() Ring-Ready sofia/internal/19008! v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.7 s=- c=IN IP4 192.168.1.7 t=0 0 m=audio 6000 RTP/AVP 0 9 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ send 463 bytes to udp/[192.168.1.122]:5060 at 14:28:50.749566: ------------------------------------------------------------------------ ACK sip:192.168.1.7:7000;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:5070;rport;branch=z9hG4bKmQtyH1yDKB9XN Route: Max-Forwards: 70 From: "19008" ;tag=mB8gaveZvmF8K To: ;tag=9 Call-ID: 7f75543a-61a1-122c-8282-001517871e28 CSeq: 110112529 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2009-01-20 09:28:50 [NOTICE] sofia.c:3065 sofia_handle_sip_i_state() Channel [sofia/internal/19008] has been answered 2009-01-20 09:28:50 [INFO] mod_sofia.c:1294 sofia_receive_message() Asked to send early media by sofia/internal/19008 at 192.168.1.122:7001 2009-01-20 09:28:50 [INFO] mod_sofia.c:1335 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1232449708 1232449709 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-01-20 09:28:50 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Ring-Ready sofia/internal/19008 at 192.168.1.122:7001! 2009-01-20 09:28:50 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Pre-Answer sofia/internal/19008 at 192.168.1.122:7001! 2009-01-20 09:28:50 [NOTICE] switch_ivr_originate.c:1838 switch_ivr_originate() Channel [sofia/internal/19008 at 192.168.1.122:7001] has been answered send 1058 bytes to udp/[192.168.1.122]:5060 at 14:28:50.820186: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bKe6e9.09c88aa1.0 Via: SIP/2.0/UDP 192.168.1.6:7001 Record-Route: From: 19008 ;tag=9 To: 0019008 >;tag=K2er80XUZBSNr Call-ID: 9-10894 at 192.168.1.6 CSeq: 2 INVITE Contact: User-Agent: Freeswitch Media Gateway Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 267 v=0 o=FreeSWITCH 6403206013178204633 6807646593930218658 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ send 1058 bytes to udp/[192.168.1.122]:5060 at 14:28:51.321221: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.122;branch=z9hG4bKe6e9.09c88aa1.0 Via: SIP/2.0/UDP 192.168.1.6:7001 Record-Route: From: 19008 ;tag=9 To: 0019008 >;tag=K2er80XUZBSNr Call-ID: 9-10894 at 192.168.1.6 CSeq: 2 INVITE Contact: User-Agent: Freeswitch Media Gateway Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 267 v=0 o=FreeSWITCH 6403206013178204633 6807646593930218658 IN IP4 192.168.1.116 s=FreeSWITCH c=IN IP4 192.168.1.116 t=0 0 m=audio 12022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/4523c005/attachment-0001.html From cyberalby at gmail.com Tue Jan 20 01:06:48 2009 From: cyberalby at gmail.com (Alberto Ceccarelli) Date: Tue, 20 Jan 2009 10:06:48 +0100 Subject: [Freeswitch-users] Dialing Out Problem via Gateway In-Reply-To: <94c121cc0901191056x6da744f9r60fd5ff21067aa45@mail.gmail.com> References: <94c121cc0901170802i18c883cdv104660fc8f132315@mail.gmail.com> <94c121cc0901191056x6da744f9r60fd5ff21067aa45@mail.gmail.com> Message-ID: <94c121cc0901200106y47b47c2cyb03877718f815016@mail.gmail.com> Sorry, but the problem remains. I can add some details. After response, we can listen the DTMF tones, but not the voice. Now my conf. files: vars.xml: dialplan\default.xml: Ciao. Alby. 2009/1/19 Brian West You modified the Local_Extension in default.xml > > 2009-01-17 16:10:08 [DEBUG] mod_dialplan_xml.c:117 parse_exten() > Regex: [Local_Extension] destination_number(1008) =~ /^+39020000000$/ > 2009-01-17 16:10:08 [ERR] switch_regex.c:94 switch_regex_perform() > COMPILE ERROR: 1 [nothing to repeat][^+39020000000$] > 2009-01-17 16:10:08 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex > mismatch > > > Remove that match for the /^+39020000000$/ and put it back to the > default value... it should work fine. > > /b > > On Jan 17, 2009, at 10:02 AM, Alberto Ceccarelli wrote: > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/2f07bb7e/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: log.zip Type: application/zip Size: 28329 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/2f07bb7e/attachment-0001.zip From krzysiez at go2.pl Tue Jan 20 02:10:57 2009 From: krzysiez at go2.pl (Krzysztof Zimnicki) Date: Tue, 20 Jan 2009 11:10:57 +0100 Subject: [Freeswitch-users] Problem with digium te220p Message-ID: <4c5d42470901200210k735ffc2dh77f5f0b5f196f97d@mail.gmail.com> >Can you post your openzap.conf file? >-MC Sure. [span zt] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 [span zt] name => OpenZAP number => 2 trunk_type => E1 b-channel => 32-46 d-channel => 47 b-channel => 48-62 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/13a88514/attachment.html From brian at freeswitch.org Tue Jan 20 03:34:54 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 20 Jan 2009 05:34:54 -0600 Subject: [Freeswitch-users] Freeswitch not sending out 183 In-Reply-To: <27c25bc40901192356j553f8bc3o1ae5295a30c0e4c7@mail.gmail.com> References: <27c25bc40901192356j553f8bc3o1ae5295a30c0e4c7@mail.gmail.com> Message-ID: I see a 180 ringing. You would see a 183 with SDP if the far end had media to provide inband ringing. /b On Jan 20, 2009, at 1:56 AM, Juan Backson wrote: > I am running some continuous testing hitting FS. There are a couple > errors that gets popped up occasionally and I am trying to find out > why. In one of the trace, I am seeing FS not sending 183. The > weird thing is that this problem is not happening everything, but on > a very rarely basis. Also, there is no error message being sent. > Does anyone know under what situation would freeswitch not sending > out 183? From msc at freeswitch.org Tue Jan 20 06:07:03 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 20 Jan 2009 06:07:03 -0800 Subject: [Freeswitch-users] Problem with digium te220p In-Reply-To: <4c5d42470901200210k735ffc2dh77f5f0b5f196f97d@mail.gmail.com> References: <4c5d42470901200210k735ffc2dh77f5f0b5f196f97d@mail.gmail.com> Message-ID: <9800552B-8A30-47EE-B9EB-837C53E05282@freeswitch.org> Can you join irc later today? I will be on as mercutioviz. I would like to discuss this more. -MC Sent from my iPhone On Jan 20, 2009, at 2:10 AM, "Krzysztof Zimnicki" wrote: > > >Can you post your openzap.conf file? > >-MC > > Sure. > > [span zt] > name => OpenZAP > number => 1 > trunk_type => E1 > b-channel => 1-15 > d-channel => 16 > b-channel => 17-31 > > [span zt] > name => OpenZAP > number => 2 > trunk_type => E1 > b-channel => 32-46 > d-channel => 47 > b-channel => 48-62 > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ravriel_1 at yahoo.com Tue Jan 20 05:52:10 2009 From: ravriel_1 at yahoo.com (Ron Avriel) Date: Tue, 20 Jan 2009 05:52:10 -0800 (PST) Subject: [Freeswitch-users] FreeSWITCH Windows Service does not Start Message-ID: <150416.65876.qm@web45202.mail.sp1.yahoo.com> Hi, If I try starting FreeSWITCH Windows service it immediately fails with a messagebox: "The FreeSWITCH service on local computer started and then stopped. Some services stop automatically...etc." I noticed that the service is installed to log on as "NT AUTHORITY\NetworkService". If I change this to "Local System account" then FreeSWITCH starts and runs OK. This failure occurred on multiple Windows XP servers. Why does it fail and why does it not use the local system account like almost all services? Thanks, Ron From bdeacon at highergear.com Mon Jan 19 16:20:16 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Mon, 19 Jan 2009 16:20:16 -0800 Subject: [Freeswitch-users] Bridge destination returned from a python script? Message-ID: <1232410816.4022.140.camel@dev03.cal.highergear.com> Greetings my soon-to-be-BFF's :) After the last 6 weeks or so thrashing about in the land of Asterisk, I must say I'm quite impressed with what I'm learning about FreeSwitch. So the scenario I'm trying to deal with is bridging a call between two not-internal endpoints where the numbers to connect are determined by database-fed business logic. The recipe here is very close to what I need: http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm Except that I'm a language snob and would greatly prefer python to javascript. :) So my minor question is can this line: Become this line: But javascript isn't a deal-breaker. I have two bridging scenarios that were too tricky for me in asterisk land. (And my apologies, I'm new to telephony, so I don't have the lingo down). Our FS server would be reachable by multiple 800 numbers. Based on the number that was dialed to reach our FS server, we would look up an external phone number + extension to be part 1 of a bridge (which we would use the group_confirm_file script to make sure we had gotten past the IVR). So for that scenario, how would I accomplish something like: Can my foo.bar.baz script make a session.setVariable call that I could drop into data=""? My other scenario is similar, but we need to externally signal the FS server to place a call to phone + extension, confirm it was connected, and then bridge it to a DID. Is XmlRpc the best (only?) way to poke FS into doing something? Thanks muchly! Brian From pauld at versafon.com Tue Jan 20 06:20:20 2009 From: pauld at versafon.com (Paul D.) Date: Tue, 20 Jan 2009 09:20:20 -0500 Subject: [Freeswitch-users] Caller ID presentation Message-ID: <621517400.51232461224223.JavaMail.james@versafon31.versafon.com> Hi, I am fairly new to FS. Is there a way to switch off/on caller ID presentation dynamically, similar to "callingpres" in *? Also, probably a stupid question, but how do I search this mailing list archive? Thx. From brian at freeswitch.org Tue Jan 20 06:58:18 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 20 Jan 2009 08:58:18 -0600 Subject: [Freeswitch-users] Caller ID presentation In-Reply-To: <621517400.51232461224223.JavaMail.james@versafon31.versafon.com> References: <621517400.51232461224223.JavaMail.james@versafon31.versafon.com> Message-ID: Google: site:lists.freeswitch.org search terms /b On Jan 20, 2009, at 8:20 AM, Paul D. wrote: > Also, probably a stupid question, but how do I search this mailing > list > archive? From msc at freeswitch.org Tue Jan 20 07:30:48 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Jan 2009 07:30:48 -0800 Subject: [Freeswitch-users] FreeSWITCH Windows Service does not Start In-Reply-To: <150416.65876.qm@web45202.mail.sp1.yahoo.com> References: <150416.65876.qm@web45202.mail.sp1.yahoo.com> Message-ID: <87f2f3b90901200730h66fbd7c6q1ed74d13a4931e02@mail.gmail.com> Ron, Thanks, I'll ask Mike J to take a quick look. -MC On Tue, Jan 20, 2009 at 5:52 AM, Ron Avriel wrote: > Hi, > > If I try starting FreeSWITCH Windows service it immediately fails with a messagebox: > > "The FreeSWITCH service on local computer started and then stopped. Some services stop automatically...etc." > > I noticed that the service is installed to log on as "NT AUTHORITY\NetworkService". > If I change this to "Local System account" then FreeSWITCH starts and runs OK. > > This failure occurred on multiple Windows XP servers. > > Why does it fail and why does it not use the local system account like almost all services? > > Thanks, > Ron > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Tue Jan 20 07:36:33 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Jan 2009 07:36:33 -0800 Subject: [Freeswitch-users] Bridge destination returned from a python script? In-Reply-To: <1232410816.4022.140.camel@dev03.cal.highergear.com> References: <1232410816.4022.140.camel@dev03.cal.highergear.com> Message-ID: <87f2f3b90901200736pefb8e2fq8b4e3ad7de86ef87@mail.gmail.com> On Mon, Jan 19, 2009 at 4:20 PM, Brian Deacon wrote: > Greetings my soon-to-be-BFF's :) > > After the last 6 weeks or so thrashing about in the land of Asterisk, I > must say I'm quite impressed with what I'm learning about FreeSwitch. > > So the scenario I'm trying to deal with is bridging a call between two > not-internal endpoints where the numbers to connect are determined by > database-fed business logic. > > The recipe here is very close to what I need: > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm > > Except that I'm a language snob and would greatly prefer python to > javascript. :) > > So my minor question is can this line: > > Become this line: > > > But javascript isn't a deal-breaker. I have two bridging scenarios that > were too tricky for me in asterisk land. (And my apologies, I'm new to > telephony, so I don't have the lingo down). Our FS server would be > reachable by multiple 800 numbers. Based on the number that was dialed > to reach our FS server, we would look up an external phone number + > extension to be part 1 of a bridge (which we would use the > group_confirm_file script to make sure we had gotten past the IVR). So > for that scenario, how would I accomplish something like: > > Can my foo.bar.baz script make a session.setVariable call that I could > drop into data=""? > If I understand your scenario correctly then I would say that your best bet would be the recently added mod_easyroute, which was designed specifically for this purpose. Check out this article and the related wiki page: http://www.freeswitch.org/node/158 > My other scenario is similar, but we need to externally signal the FS > server to place a call to phone + extension, confirm it was connected, > and then bridge it to a DID. Is XmlRpc the best (only?) way to poke FS > into doing something? When you say "externally signal" do you mean you have a 3rd party application (or something) that needs to talk to FS? Could you detail the scenario a bit more? I'm positive FS can do what you want but with more detail we can help you find the best way to do it. -MC > > Thanks muchly! > > Brian > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Tue Jan 20 08:08:25 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 20 Jan 2009 11:08:25 -0500 Subject: [Freeswitch-users] FreeSWITCH Windows Service does not Start In-Reply-To: <150416.65876.qm@web45202.mail.sp1.yahoo.com> References: <150416.65876.qm@web45202.mail.sp1.yahoo.com> Message-ID: One issue with the service is we have no console to dump errors too, it sounds like it is failing one of the startup requirements like config files being there. Are you able to start it in non service mode? If so, check permissions on the freeswitch dir that the user running the service has permissions to that dir. Mike On Jan 20, 2009, at 8:52 AM, Ron Avriel wrote: > Hi, > > If I try starting FreeSWITCH Windows service it immediately fails > with a messagebox: > > "The FreeSWITCH service on local computer started and then stopped. > Some services stop automatically...etc." > > I noticed that the service is installed to log on as "NT AUTHORITY > \NetworkService". > If I change this to "Local System account" then FreeSWITCH starts > and runs OK. > > This failure occurred on multiple Windows XP servers. > > Why does it fail and why does it not use the local system account > like almost all services? From mike at jerris.com Tue Jan 20 08:12:29 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 20 Jan 2009 11:12:29 -0500 Subject: [Freeswitch-users] Caller ID presentation In-Reply-To: <621517400.51232461224223.JavaMail.james@versafon31.versafon.com> References: <621517400.51232461224223.JavaMail.james@versafon31.versafon.com> Message-ID: <47870B40-76FD-469C-811F-F13EA07D140E@jerris.com> On Jan 20, 2009, at 9:20 AM, Paul D. wrote: > Hi, > I am fairly new to FS. Is there a way to switch off/on caller ID > presentation dynamically, similar to "callingpres" in *? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_privacy From tomasborrella at gmail.com Tue Jan 20 08:46:56 2009 From: tomasborrella at gmail.com (=?ISO-8859-1?Q?Tom=E1s?=) Date: Tue, 20 Jan 2009 17:46:56 +0100 Subject: [Freeswitch-users] Hang up not received Message-ID: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/e0d18e3a/attachment.html From anthony.minessale at gmail.com Tue Jan 20 08:49:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 20 Jan 2009 10:49:26 -0600 Subject: [Freeswitch-users] Bridge destination returned from a python script? In-Reply-To: <1232410816.4022.140.camel@dev03.cal.highergear.com> References: <1232410816.4022.140.camel@dev03.cal.highergear.com> Message-ID: <191c3a030901200849t612aff33ub7a5427d41749692@mail.gmail.com> regardless of what you execute, if the execute is over and the call has not been hungup by the app it's considered success. So you should be able to do a python script as well just be sure to hangup on anything that does not meet the criteria. On Mon, Jan 19, 2009 at 6:20 PM, Brian Deacon wrote: > Greetings my soon-to-be-BFF's :) > > After the last 6 weeks or so thrashing about in the land of Asterisk, I > must say I'm quite impressed with what I'm learning about FreeSwitch. > > So the scenario I'm trying to deal with is bridging a call between two > not-internal endpoints where the numbers to connect are determined by > database-fed business logic. > > The recipe here is very close to what I need: > > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm > > Except that I'm a language snob and would greatly prefer python to > javascript. :) > > So my minor question is can this line: > > Become this line: > > > But javascript isn't a deal-breaker. I have two bridging scenarios that > were too tricky for me in asterisk land. (And my apologies, I'm new to > telephony, so I don't have the lingo down). Our FS server would be > reachable by multiple 800 numbers. Based on the number that was dialed > to reach our FS server, we would look up an external phone number + > extension to be part 1 of a bridge (which we would use the > group_confirm_file script to make sure we had gotten past the IVR). So > for that scenario, how would I accomplish something like: > > Can my foo.bar.baz script make a session.setVariable call that I could > drop into data=""? > > My other scenario is similar, but we need to externally signal the FS > server to place a call to phone + extension, confirm it was connected, > and then bridge it to a DID. Is XmlRpc the best (only?) way to poke FS > into doing something? > > Thanks muchly! > > Brian > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/ba45af64/attachment.html From kokoska.rokoska at post.cz Tue Jan 20 08:48:59 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 20 Jan 2009 17:48:59 +0100 Subject: [Freeswitch-users] Caller ID presentation In-Reply-To: <47870B40-76FD-469C-811F-F13EA07D140E@jerris.com> References: <621517400.51232461224223.JavaMail.james@versafon31.versafon.com> <47870B40-76FD-469C-811F-F13EA07D140E@jerris.com> Message-ID: <4976007B.5070202@post.cz> Michael Jerris napsal(a): > On Jan 20, 2009, at 9:20 AM, Paul D. wrote: > >> Hi, >> I am fairly new to FS. Is there a way to switch off/on caller ID >> presentation dynamically, similar to "callingpres" in *? > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_privacy > > I'm not sure because I didn't try * callingpres for a long time, but I think it completly hide callerID (CLIP), while FreeSWITCH application "privacy" just "modifies" RPID header. (I am talking about SIP.) If I am not true, please correct me, but if you need real privacy you need: FreeSWITCH alone method: 1. set effective_caller_id_name=announymous 2. set effective_caller_id_number=annonymous 3. set sip_h_Privacy=id 4. set privacy=yes and hope that nobody will look into the packets, because CLIP still remains in the RPID header. FreeSWITCH + Kamailio/OpenSIPS/SER: 1. set sip_h_Privacy=id on FreeSWITCH side 2. if you see Privacy=id header on your proxy, than remove RPID header (better to do it all the time, because RPID is depricated) and rewrite >From header to desired: From: "Anonymous" using uac_replace_from("Anonymous","sip:anonymous at anonymous.invalid"); Hope this helps :-) Best regards, kokoska.rokoska From scott.ellis at novatex.com.au Tue Jan 20 09:03:34 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Wed, 21 Jan 2009 04:03:34 +1100 Subject: [Freeswitch-users] Hang up not received In-Reply-To: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> References: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> Message-ID: <497603E6.3090508@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/fe38d13c/attachment.html From anthony.minessale at gmail.com Tue Jan 20 09:05:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 20 Jan 2009 11:05:19 -0600 Subject: [Freeswitch-users] Hang up not received In-Reply-To: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> References: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> Message-ID: <191c3a030901200905r156a0b79uad5baa726eeeb7c0@mail.gmail.com> This is a common issue with analog phones even traditional answering machines suffer from it. I'm sure you must have had an answering machine at some point that has dial tone as the message it receives. Unless FreeSWITCH has some hint that the call has hungup it will not stop trying to complete the call. If the other side is sending a busy tone to indicate hangup it's possible to use the tone_detect app to pick up on the tones and abort the call. Another thing you could do if you have unlimited inbound is explicitly answer the call in the dialplan before you call your sip phones this will give you a more profound hangup detection but it will make every call count even when nobody answers. On Tue, Jan 20, 2009 at 10:46 AM, Tom?s wrote: > 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. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/72ad3d25/attachment.html From mgg at giagnocavo.net Tue Jan 20 09:05:14 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 20 Jan 2009 12:05:14 -0500 Subject: [Freeswitch-users] FreeSWITCH Windows Service does not Start In-Reply-To: <150416.65876.qm@web45202.mail.sp1.yahoo.com> References: <150416.65876.qm@web45202.mail.sp1.yahoo.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670235F11520@mse17be1.mse17.exchange.ms> You should run things with as little permission as possible. In the case of FreeSWITCH, there's no need for it to run as LocalSystem (which is like a root account). NetworkService is much more restricted, which is good in the case of a security failure, such as a buffer overrun or a misconfigured module/application. Maybe it's not loading because NetworkService doesn't have permissions to the FS directory. -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron Avriel Sent: Tuesday, January 20, 2009 6:52 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeSWITCH Windows Service does not Start Hi, If I try starting FreeSWITCH Windows service it immediately fails with a messagebox: "The FreeSWITCH service on local computer started and then stopped. Some services stop automatically...etc." I noticed that the service is installed to log on as "NT AUTHORITY\NetworkService". If I change this to "Local System account" then FreeSWITCH starts and runs OK. This failure occurred on multiple Windows XP servers. Why does it fail and why does it not use the local system account like almost all services? Thanks, Ron _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From marek at telesave.pl Tue Jan 20 09:09:44 2009 From: marek at telesave.pl (Marek Gorecki) Date: Tue, 20 Jan 2009 18:09:44 +0100 Subject: [Freeswitch-users] what could be efficiency of this hardware ? In-Reply-To: References: Message-ID: <6a7111020901200909o47b9668do9570e1e610fce123@mail.gmail.com> Hi, Merit question: how many calls could be processed by FS on the following hardware: Intel Core 2 Duo 2*2,13 Ghz 3072 MB RAM 400 GB HD 100mbit/s port By processing calls I understand just switching with billing - no processing of any media, no transcoding, just call-through according to dialplan. Question is related to availablility of such server for 90? / month with no charge and no limit for bandwidth ( say: NO LIMIT IN BANDWIDTH ) ( http://www.giga-international.com/detail_server.php?id=87 ) Which parameters should be upgraded for to reach better efficiency ? Regards, /\/\arekg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/09a3c109/attachment.html From scott.ellis at novatex.com.au Tue Jan 20 09:11:58 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Wed, 21 Jan 2009 04:11:58 +1100 Subject: [Freeswitch-users] api_hangup_hook not actually executing the command Message-ID: <497605DE.8010307@novatex.com.au> The on answer is fine, the api_hangup_hook gets to the point where it wants to execute, but then nothing happens. Any thoughts? 2009-01-21 03:08:00 [DEBUG] switch_channel.c:1773 switch_channel_perform_mark_answered() sofia/internal/Line_9 at 10.0.0.9 execute on answer: set_global(10.0.0.19_INCALL=true)2009-01-21 03:08:00 [DEBUG] mod_dptools.c:726 set_global_function() SET GLOBAL [10.0.0.19_INCALL]=[true] 2009-01-21 03:08:09 [DEBUG] switch_core_state_machine.c:416 switch_core_session_run() Hangup Command set_global(10.0.0.19_INCALL=false): It does not execute - the global variable is not set...quite odd. Scott From msc at freeswitch.org Tue Jan 20 09:20:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Jan 2009 09:20:07 -0800 Subject: [Freeswitch-users] api_hangup_hook not actually executing the command In-Reply-To: <497605DE.8010307@novatex.com.au> References: <497605DE.8010307@novatex.com.au> Message-ID: <87f2f3b90901200920paafc4ddw100e4534c78ef075@mail.gmail.com> Can you pastebin the entire call from start to finish? Also, pastebin your dialplan extension. -MC On Tue, Jan 20, 2009 at 9:11 AM, Scott Ellis wrote: > The on answer is fine, the api_hangup_hook gets to the point where it > wants to execute, but then nothing happens. Any thoughts? > > > 2009-01-21 03:08:00 [DEBUG] switch_channel.c:1773 > switch_channel_perform_mark_answered() sofia/internal/Line_9 at 10.0.0.9 > execute on answer: set_global(10.0.0.19_INCALL=true)2009-01-21 03:08:00 > [DEBUG] mod_dptools.c:726 set_global_function() SET GLOBAL > [10.0.0.19_INCALL]=[true] > > 2009-01-21 03:08:09 [DEBUG] switch_core_state_machine.c:416 > switch_core_session_run() Hangup Command set_global(10.0.0.19_INCALL=false): > > It does not execute - the global variable is not set...quite odd. > > Scott > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Jan 20 09:24:28 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 20 Jan 2009 11:24:28 -0600 Subject: [Freeswitch-users] api_hangup_hook not actually executing the command In-Reply-To: <87f2f3b90901200920paafc4ddw100e4534c78ef075@mail.gmail.com> References: <497605DE.8010307@novatex.com.au> <87f2f3b90901200920paafc4ddw100e4534c78ef075@mail.gmail.com> Message-ID: I have just tested this: When I hangup I see: 2009-01-20 11:22:41 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() Hangup Command originate(user/1008 at bkw.org 9999): Works.... So yes we need to see the full log. /b On Jan 20, 2009, at 11:20 AM, Michael Collins wrote: > Can you pastebin the entire call from start to finish? Also, pastebin > your dialplan extension. > -MC From brian at freeswitch.org Tue Jan 20 09:25:17 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 20 Jan 2009 11:25:17 -0600 Subject: [Freeswitch-users] what could be efficiency of this hardware ? In-Reply-To: <6a7111020901200909o47b9668do9570e1e610fce123@mail.gmail.com> References: <6a7111020901200909o47b9668do9570e1e610fce123@mail.gmail.com> Message-ID: <2E4AE714-E392-4152-B313-C78227DDE151@freeswitch.org> We can't tell 100% what you're doing or what load you'll be pulling doing various things on the machine. So the answer can be found if you load test and see what your limits are. /b On Jan 20, 2009, at 11:09 AM, Marek Gorecki wrote: > Hi, > > Merit question: > how many calls could be processed by FS on the following hardware: > > > Intel Core 2 Duo 2*2,13 Ghz > > 3072 MB RAM > > 400 GB HD > > 100mbit/s port > > By processing calls I understand just switching with billing - no > processing of any media, no transcoding, just call-through according > to dialplan. > > Question is related to availablility of such server for 90? / month > with no charge and no limit for bandwidth ( say: NO LIMIT IN > BANDWIDTH ) > ( http://www.giga-international.com/detail_server.php?id=87 ) > > > Which parameters should be upgraded for to reach better efficiency ? > > Regards, > /\/\arekg > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/6e38648f/attachment.html From scott.ellis at novatex.com.au Tue Jan 20 09:35:09 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Wed, 21 Jan 2009 04:35:09 +1100 Subject: [Freeswitch-users] api_hangup_hook not actually executing the command In-Reply-To: References: <497605DE.8010307@novatex.com.au> <87f2f3b90901200920paafc4ddw100e4534c78ef075@mail.gmail.com> Message-ID: <49760B4D.2030609@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/8d635100/attachment.html From anthony.minessale at gmail.com Tue Jan 20 09:47:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 20 Jan 2009 11:47:51 -0600 Subject: [Freeswitch-users] what could be efficiency of this hardware ? In-Reply-To: <2E4AE714-E392-4152-B313-C78227DDE151@freeswitch.org> References: <6a7111020901200909o47b9668do9570e1e610fce123@mail.gmail.com> <2E4AE714-E392-4152-B313-C78227DDE151@freeswitch.org> Message-ID: <191c3a030901200947k787b9724g23eec9cf1e843611@mail.gmail.com> we can tell you that wil 100mb Ethernet, that you are limited to about 1200 calls. On Tue, Jan 20, 2009 at 11:25 AM, Brian West wrote: > We can't tell 100% what you're doing or what load you'll be pulling doing > various things on the machine. So the answer can be found if you load test > and see what your limits are. > /b > > On Jan 20, 2009, at 11:09 AM, Marek Gorecki wrote: > > Hi, > > Merit question: > how many calls could be processed by FS on the following hardware: > > Intel Core 2 Duo 2*2,13 Ghz > 3072 MB RAM > 400 GB HD > 100mbit/s port > By processing calls I understand just switching with billing - no > processing of any media, no transcoding, just call-through according to > dialplan. > > Question is related to availablility of such server for 90? / month with no > charge and no limit for bandwidth ( say: NO LIMIT IN BANDWIDTH ) > ( http://www.giga-international.com/detail_server.php?id=87 ) > > > Which parameters should be upgraded for to reach better efficiency ? > > Regards, > /\/\arekg > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/47bdcfd2/attachment-0001.html From info at freeswitch.es Tue Jan 20 11:45:18 2009 From: info at freeswitch.es (info at freeswitch.es) Date: Tue, 20 Jan 2009 14:45:18 -0500 (COT) Subject: [Freeswitch-users] Stun disabled Message-ID: <1232480718.22206@freeswitch.es> Hi, very basic question: how can I disable stun for internal and external profiles? Thank you in advance Regards From oseslija at gmail.com Tue Jan 20 11:45:56 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Tue, 20 Jan 2009 20:45:56 +0100 Subject: [Freeswitch-users] Hang up not received In-Reply-To: <191c3a030901200905r156a0b79uad5baa726eeeb7c0@mail.gmail.com> References: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> <191c3a030901200905r156a0b79uad5baa726eeeb7c0@mail.gmail.com> Message-ID: <4468a6770901201145m444a2c82gacaffd369579a47c@mail.gmail.com> I tried similar setup with my analog card (X100P) and I'm having same issue. Call is not hungup on the oz side once the caller ends. My telco doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck to detecting busy tone from the telco side. I'll try to modify tones.conf accordingly. Regards, Ognjen (sekil) On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > This is a common issue with analog phones even traditional answering > machines suffer from it. > I'm sure you must have had an answering machine at some point that has dial > tone as the message it receives. > > Unless FreeSWITCH has some hint that the call has hungup it will not stop > trying to complete the call. > > If the other side is sending a busy tone to indicate hangup it's possible > to use the tone_detect app to pick > up on the tones and abort the call. > > Another thing you could do if you have unlimited inbound is explicitly > answer the call in the dialplan before > you call your sip phones this will give you a more profound hangup > detection but it will make every call count > even when nobody answers. > > > > On Tue, Jan 20, 2009 at 10:46 AM, Tom?s wrote: > >> 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. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/b2988822/attachment.html From brian at freeswitch.org Tue Jan 20 12:02:53 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 20 Jan 2009 14:02:53 -0600 Subject: [Freeswitch-users] Stun disabled In-Reply-To: <1232480718.22206@freeswitch.es> References: <1232480718.22206@freeswitch.es> Message-ID: <173802A0-0A41-4BC1-8E5A-651353111EE7@freeswitch.org> Remove the ext-sip-ip and ext-rtp-ip which are set to stun:stun.freeswitch.org that will work. /b On Jan 20, 2009, at 1:45 PM, info at freeswitch.es wrote: > Hi, > > very basic question: > > how can I disable stun for internal and external profiles? > > Thank you in advance > > Regards From oseslija at gmail.com Tue Jan 20 13:43:06 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Tue, 20 Jan 2009 22:43:06 +0100 Subject: [Freeswitch-users] Hang up not received In-Reply-To: <4468a6770901201145m444a2c82gacaffd369579a47c@mail.gmail.com> References: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> <191c3a030901200905r156a0b79uad5baa726eeeb7c0@mail.gmail.com> <4468a6770901201145m444a2c82gacaffd369579a47c@mail.gmail.com> Message-ID: <4468a6770901201343q3a51a493m2b721d4531346a77@mail.gmail.com> Ok, as discussed with Tony on IRC channel I followed his directions which lead to a successfull outcome (like it always does I might add :). One has to use tone_detect app in FreeSWITCH dialplan in order to check for busy tones coming from the PSTN side and if matched fire a hangup application. This is the snippet of my test dp that does the trick (from extension Local_extensions in default.xml): > This means that FS will listen to freq of 425 Hz and wait for 4 positive detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 Hz is the freq telco here uses; for other countries I suggest getting the ITU world tones pdf file and check there): 2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 1/4 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 2/4 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 3/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 4/4 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() TONE busy DETECTED 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING] Regards, Ognjen On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija wrote: > I tried similar setup with my analog card (X100P) and I'm having same > issue. Call is not hungup on the oz side once the caller ends. My telco > doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck > to detecting busy tone from the telco side. I'll try to modify tones.conf > accordingly. > > Regards, > Ognjen > (sekil) > On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> This is a common issue with analog phones even traditional answering >> machines suffer from it. >> I'm sure you must have had an answering machine at some point that has >> dial tone as the message it receives. >> >> Unless FreeSWITCH has some hint that the call has hungup it will not stop >> trying to complete the call. >> >> If the other side is sending a busy tone to indicate hangup it's possible >> to use the tone_detect app to pick >> up on the tones and abort the call. >> >> Another thing you could do if you have unlimited inbound is explicitly >> answer the call in the dialplan before >> you call your sip phones this will give you a more profound hangup >> detection but it will make every call count >> even when nobody answers. >> >> >> >> On Tue, Jan 20, 2009 at 10:46 AM, Tom?s wrote: >> >>> 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. >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 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 >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/e99f8009/attachment-0001.html From jbr at consiglia.dk Tue Jan 20 14:25:24 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Tue, 20 Jan 2009 23:25:24 +0100 Subject: [Freeswitch-users] SBC functions in FS Message-ID: I'm looking into the first design phase of a setup like this: PSTN <-> FS <-> Various SIP PBXs or nothing <-> SBC <-> Internet <-> Customer Where I'm considering using the FS as the SBC and as an alternative to OpenSIPS (OpenSER), where the FS has the advantage of a design having only one type of switch, e.g. less education to the team supporting the system. If this should work, the FS should be able to route SIP messages such as NOTIFY, SUBSCRIBE etc. Is that possible, and what are the measures to implement such a routing function? /Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/931b0ba6/attachment.html From mgg at giagnocavo.net Tue Jan 20 14:35:59 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 20 Jan 2009 17:35:59 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670235F10C2A@mse17be1.mse17.exchange.ms> References: <021001c97768$8c806a60$a5813f20$@net> <6E8D2069C08AA84A83D336E996AE4C670235F10C2A@mse17be1.mse17.exchange.ms> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670235F116EE@mse17be1.mse17.exchange.ms> OK, I just checked in a fix that should make it load, or at least print a helpful exception. However, you will need the freeswitch/mod folder in the LD path, so: export LD_LIBRARY_PATH=/usr/local/freeswitch/mod or edit the ld configs. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Giagnocavo Sent: Thursday, January 15, 2009 4:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 Last time I tried it I actually built from a snapshot, which should be less stable. It was on CentOS 5.2, however. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long Sent: Thursday, January 15, 2009 4:25 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 OK yah i'm experiencing exact same problem here CentOS 5.2 2.6.18-92.el5 #1 SMP Tue Jun 10 18:49:47 EDT 2008 i686 athlon i386 GNU/Linux I too have no problems at all on Windows. I'm going to try a Suse or Ubuntu prebuilt/packaged with Mono 2 I suspect it may be kernel/mono incompatibility. Did you compile mono from tarbal? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim B Sent: Thursday, January 15, 2009 6:18 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 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 at lists.freeswitch.org > Subject: Freeswitch-users Digest, Vol 31, Issue 77 > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 15 Jan 2009 00:20:53 -0800 > > Send Freeswitch-users mailing list submissions to > freeswitch-users at 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 at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at 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 > Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 > To: > Message-ID: > 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 at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig > [root at phone2 mod_managed]# make > [root at phone2 mod_managed]# make install > > 3) Added to modules.conf.xml : > > > 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_012009 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/6a5facdc/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Thu, 15 Jan 2009 00:34:20 -0500 > From: Michael Giagnocavo > Subject: Re: [Freeswitch-users] Using mod_managed Linux/Mono 2.02 > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: > <6E8D2069C08AA84A83D336E996AE4C670235BBB97F at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long > Sent: Wednesday, January 14, 2009 3:45 PM > To: freeswitch-users at 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 at 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 > 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/20090115/73ac27e4/attachment-0001.html > > ------------------------------ > > Message: 3 > Date: Thu, 15 Jan 2009 17:50:30 +1100 > From: Scott Ellis > Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - > Freeswitch equivalent? > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: <496EDCB6.4020802 at 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 > Subject: [Freeswitch-users] Country specific tones - how to > contribute? > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: <496EE2BD.2050102 at 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 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 > Subject: Re: [Freeswitch-users] Country specific tones - how to > contribute? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <20090115072405.GA15789 at jdc.jasonjgw.net> > Content-Type: text/plain; charset=us-ascii > > Scott Ellis 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 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" > Subject: [Freeswitch-users] Changes in PlayAndGetDigits > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <27c25bc40901142343l34a3e99ftecf0df971e8e32f6 at 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 > Subject: Re: [Freeswitch-users] OpenZAP parse error [-3012] > [Q931E_INVALID_CRV] > To: freeswitch-users at lists.freeswitch.org > Message-ID: <496EF1C2.8020607 at gmx.net> > Content-Type: text/plain; charset=ISO-8859-1 > > Hello Michael, > > how much $$ are we talking about? I need this issue to be solved quickly > and it's worth to spend some money. > > I've read the following post: > > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05792.html > and have the same symptom with "after hundreds of calls I start to get b > channels that are stuck in states like "TERMINATING" or "HANGUP"" > > Best regards > Peter > > Michael Collins schrieb: > > I believe these are all symptoms of something that Stefan is working > > on: better Q931 timers. It's been on the todo list for some time but > > we've had absolutely NOBODY willing to pony up serious $$ 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 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 at 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 at 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 at 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(tm): Keep your life in sync. Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/33b333a3/attachment-0001.html From brian at freeswitch.org Tue Jan 20 14:38:01 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 20 Jan 2009 16:38:01 -0600 Subject: [Freeswitch-users] SBC functions in FS In-Reply-To: References: Message-ID: <17F1D20D-4052-48B0-A234-B3E7BC517D2F@freeswitch.org> This isn't possible currently without some code changes. /b On Jan 20, 2009, at 4:25 PM, Jon Bruel wrote: > > If this should work, the FS should be able to route SIP messages > such as NOTIFY, SUBSCRIBE etc. Is that possible, and what are the > measures to implement such a routing function? /Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/a9cfb253/attachment.html From pauld at versafon.com Tue Jan 20 16:23:33 2009 From: pauld at versafon.com (Paul D.) Date: Tue, 20 Jan 2009 19:23:33 -0500 Subject: [Freeswitch-users] Caller ID presentation In-Reply-To: <4976007B.5070202@post.cz> References: <621517400.51232461224223.JavaMail.james@versafon31.versafon.com> <47870B40-76FD-469C-811F-F13EA07D140E@jerris.com> <4976007B.5070202@post.cz> Message-ID: <1891770839.61232497410547.JavaMail.james@versafon31.versafon.com> kokoska rokoska wrote: > > Michael Jerris napsal(a): > >> On Jan 20, 2009, at 9:20 AM, Paul D. wrote: >> >> >>> Hi, >>> I am fairly new to FS. Is there a way to switch off/on caller ID >>> presentation dynamically, similar to "callingpres" in *? >>> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_privacy >> >> >> > > I'm not sure because I didn't try * callingpres for a long time, but I > think it completly hide callerID (CLIP), while FreeSWITCH application > "privacy" just "modifies" RPID header. (I am talking about SIP.) > > If I am not true, please correct me, but if you need real privacy you need: > > FreeSWITCH alone method: > 1. set effective_caller_id_name=announymous > 2. set effective_caller_id_number=annonymous > 3. set sip_h_Privacy=id > 4. set privacy=yes > and hope that nobody will look into the packets, because CLIP still > remains in the RPID header. > > FreeSWITCH + Kamailio/OpenSIPS/SER: > 1. set sip_h_Privacy=id on FreeSWITCH side > 2. if you see Privacy=id header on your proxy, than remove RPID header > (better to do it all the time, because RPID is depricated) and rewrite > >From header to desired: > From: "Anonymous" > using uac_replace_from("Anonymous","sip:anonymous at anonymous.invalid"); > > Hope this helps :-) > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Thanks guys, appreciate your help, especially "kokoska.rokoska" for such detailed answer. From bdeacon at highergear.com Tue Jan 20 17:28:16 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Tue, 20 Jan 2009 17:28:16 -0800 Subject: [Freeswitch-users] Bridge destination returned from a python script? In-Reply-To: <87f2f3b90901200736pefb8e2fq8b4e3ad7de86ef87@mail.gmail.com> References: <1232410816.4022.140.camel@dev03.cal.highergear.com> <87f2f3b90901200736pefb8e2fq8b4e3ad7de86ef87@mail.gmail.com> Message-ID: <1232501296.4022.151.camel@dev03.cal.highergear.com> Thanks for the quick turnaround, Michael, I'll check out mod_easyroute right away. As to your other need-for-clarification: > My other scenario is similar, but we need to externally signal the FS > > server to place a call to phone + extension, confirm it was connected, > > and then bridge it to a DID. Is XmlRpc the best (only?) way to poke FS > > into doing something? > > When you say "externally signal" do you mean you have a 3rd party > application (or something) that needs to talk to FS? Could you detail > the scenario a bit more? I'm positive FS can do what you want but with > more detail we can help you find the best way to do it. > -MC > So we'll have a browser-based app making REST calls to our java-based web server (tomcat). The tomcat server makes a few decisions about what to do and then chooses among several actions which involve triggering some activity from the FS server on another physical machine (which will be behind the same firewall if that matters). For instance, a user would click a link on a webpage that would cause freeswitch to call the phone at their desk, have them press # or something to indicate they are a human and not an IVR, then dial another number to bridge that user to. I'm including my original question below in case you need to refer to it. Thanks again, Brian On Tue, 2009-01-20 at 07:36 -0800, Michael Collins wrote: > On Mon, Jan 19, 2009 at 4:20 PM, Brian Deacon wrote: > > Greetings my soon-to-be-BFF's :) > > > > After the last 6 weeks or so thrashing about in the land of Asterisk, I > > must say I'm quite impressed with what I'm learning about FreeSwitch. > > > > So the scenario I'm trying to deal with is bridging a call between two > > not-internal endpoints where the numbers to connect are determined by > > database-fed business logic. > > > > The recipe here is very close to what I need: > > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm > > > > Except that I'm a language snob and would greatly prefer python to > > javascript. :) > > > > So my minor question is can this line: > > > > Become this line: > > > > > > But javascript isn't a deal-breaker. I have two bridging scenarios that > > were too tricky for me in asterisk land. (And my apologies, I'm new to > > telephony, so I don't have the lingo down). Our FS server would be > > reachable by multiple 800 numbers. Based on the number that was dialed > > to reach our FS server, we would look up an external phone number + > > extension to be part 1 of a bridge (which we would use the > > group_confirm_file script to make sure we had gotten past the IVR). So > > for that scenario, how would I accomplish something like: > > > > Can my foo.bar.baz script make a session.setVariable call that I could > > drop into data=""? > > > > If I understand your scenario correctly then I would say that your > best bet would be the recently added mod_easyroute, which was designed > specifically for this purpose. Check out this article and the related > wiki page: > http://www.freeswitch.org/node/158 > > > My other scenario is similar, but we need to externally signal the FS > > server to place a call to phone + extension, confirm it was connected, > > and then bridge it to a DID. Is XmlRpc the best (only?) way to poke FS > > into doing something? > > When you say "externally signal" do you mean you have a 3rd party > application (or something) that needs to talk to FS? Could you detail > the scenario a bit more? I'm positive FS can do what you want but with > more detail we can help you find the best way to do it. > -MC > > > > > Thanks muchly! > > > > Brian > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From stevecrozz at gmail.com Tue Jan 20 17:36:26 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Tue, 20 Jan 2009 17:36:26 -0800 Subject: [Freeswitch-users] Bridge destination returned from a python script? In-Reply-To: <1232501296.4022.151.camel@dev03.cal.highergear.com> References: <1232410816.4022.140.camel@dev03.cal.highergear.com> <87f2f3b90901200736pefb8e2fq8b4e3ad7de86ef87@mail.gmail.com> <1232501296.4022.151.camel@dev03.cal.highergear.com> Message-ID: <11990ade0901201736o5cdf554fi607c8755b304db6a@mail.gmail.com> mod_socket On Tue, Jan 20, 2009 at 5:28 PM, Brian Deacon wrote: > Thanks for the quick turnaround, Michael, > > I'll check out mod_easyroute right away. As to your other > need-for-clarification: > >> My other scenario is similar, but we need to externally signal the FS >> > server to place a call to phone + extension, confirm it was connected, >> > and then bridge it to a DID. Is XmlRpc the best (only?) way to poke FS >> > into doing something? >> >> When you say "externally signal" do you mean you have a 3rd party >> application (or something) that needs to talk to FS? Could you detail >> the scenario a bit more? I'm positive FS can do what you want but with >> more detail we can help you find the best way to do it. >> -MC >> > > So we'll have a browser-based app making REST calls to our java-based > web server (tomcat). The tomcat server makes a few decisions about what > to do and then chooses among several actions which involve triggering > some activity from the FS server on another physical machine (which will > be behind the same firewall if that matters). > > For instance, a user would click a link on a webpage that would cause > freeswitch to call the phone at their desk, have them press # or > something to indicate they are a human and not an IVR, then dial another > number to bridge that user to. > > I'm including my original question below in case you need to refer to > it. > > Thanks again, > Brian > > > > On Tue, 2009-01-20 at 07:36 -0800, Michael Collins wrote: >> On Mon, Jan 19, 2009 at 4:20 PM, Brian Deacon wrote: >> > Greetings my soon-to-be-BFF's :) >> > >> > After the last 6 weeks or so thrashing about in the land of Asterisk, I >> > must say I'm quite impressed with what I'm learning about FreeSwitch. >> > >> > So the scenario I'm trying to deal with is bridging a call between two >> > not-internal endpoints where the numbers to connect are determined by >> > database-fed business logic. >> > >> > The recipe here is very close to what I need: >> > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm >> > >> > Except that I'm a language snob and would greatly prefer python to >> > javascript. :) >> > >> > So my minor question is can this line: >> > >> > Become this line: >> > >> > >> > But javascript isn't a deal-breaker. I have two bridging scenarios that >> > were too tricky for me in asterisk land. (And my apologies, I'm new to >> > telephony, so I don't have the lingo down). Our FS server would be >> > reachable by multiple 800 numbers. Based on the number that was dialed >> > to reach our FS server, we would look up an external phone number + >> > extension to be part 1 of a bridge (which we would use the >> > group_confirm_file script to make sure we had gotten past the IVR). So >> > for that scenario, how would I accomplish something like: >> > >> > Can my foo.bar.baz script make a session.setVariable call that I could >> > drop into data=""? >> > >> >> If I understand your scenario correctly then I would say that your >> best bet would be the recently added mod_easyroute, which was designed >> specifically for this purpose. Check out this article and the related >> wiki page: >> http://www.freeswitch.org/node/158 >> >> > My other scenario is similar, but we need to externally signal the FS >> > server to place a call to phone + extension, confirm it was connected, >> > and then bridge it to a DID. Is XmlRpc the best (only?) way to poke FS >> > into doing something? >> >> When you say "externally signal" do you mean you have a 3rd party >> application (or something) that needs to talk to FS? Could you detail >> the scenario a bit more? I'm positive FS can do what you want but with >> more detail we can help you find the best way to do it. >> -MC >> >> > >> > Thanks muchly! >> > >> > Brian >> > >> > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Laurent.Fabre at kirranet.com Tue Jan 20 19:57:25 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Wed, 21 Jan 2009 04:57:25 +0100 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed Message-ID: Hi, I'm experiencing a few ? random ? segfaults ever since I enabled core ODBC support. They usually happen during startup or quickly after...or not at all as long as I do not restart freeswitch. The backend is a postgresql 7.4, every module has a distinct database (LATIN9, with OIDS). I'm having a few issues with presence & call routing which tend to break from time to time. The freeswitch box is hosted and I've got several phones with STUN enabled behind a NAT firewall (namely pfSense). I'm not sure how those things might relate but maybe it's worth mentionning. I've decided to update a few hours ago but still crashing. I was about to test mod_easyroute when I decided to run gdb and look into it : http://pastebin.freeswitch.org/6830 Could someone please help me pointing the issue? :) Thanks in advance, -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61<+33170247461> laurent.fabre at kirranet.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/8d582c80/attachment.html From brian at freeswitch.org Tue Jan 20 19:56:27 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 20 Jan 2009 21:56:27 -0600 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed In-Reply-To: References: Message-ID: Please open a jira http://jira.freeswitch.org include all the requested data in detail. /b On Jan 20, 2009, at 9:57 PM, Laurent Fabre wrote: > Hi, > > I?m experiencing a few ? random ? segfaults ever since I enabled > core ODBC support. They usually happen during startup or quickly > after?or not at all as long as I do not restart freeswitch. > > The backend is a postgresql 7.4, every module has a distinct > database (LATIN9, with OIDS). > > I?m having a few issues with presence & call routing which tend to > break from time to time. The freeswitch box is hosted and I?ve got > several phones with STUN enabled behind a NAT firewall (namely > pfSense). I?m not sure how those things might relate but maybe it?s > worth mentionning. > > I?ve decided to update a few hours ago but still crashing. > > I was about to test mod_easyroute when I decided to run gdb and look > into it : http://pastebin.freeswitch.org/6830 > > Could someone please help me pointing the issue? J > > Thanks in advance, > > -- Laurent FABRE > Directeur g?n?ral > 10, rue d'Aumale > 75009 Paris > Tel: +33.(0)1.42.81.28.20 > Mob: +33.(0)6.75.75.02.96 > Fax: +33.(0)1.70.24.74.61 > laurent.fabre at kirranet.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090120/55572bc2/attachment-0001.html From gkuri at ieee.org Tue Jan 20 20:30:51 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 20 Jan 2009 22:30:51 -0600 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed In-Reply-To: References: Message-ID: <4976A4FB.6030807@ieee.org> I've been experiencing the same since I upgraded to the latest rev of trunk from rev 11113, but I've been waiting to collect more data before opening a jira and try to at least narrow down which revision broke things. Laurent - If you open a jira, I'll add the output of the core dumps, debugging info, and any other data I have up to this point to the jira. Gabe Brian West wrote: > Please open a jira http://jira.freeswitch.org include all the requested > data in detail. > > /b > > On Jan 20, 2009, at 9:57 PM, Laurent Fabre wrote: > >> Hi, >> >> I?m experiencing a few ? random ? segfaults ever since I enabled core >> ODBC support. They usually happen during startup or quickly after?or >> not at all as long as I do not restart freeswitch. >> >> The backend is a postgresql 7.4, every module has a distinct database >> (LATIN9, with OIDS). >> >> I?m having a few issues with presence & call routing which tend to >> break from time to time. The freeswitch box is hosted and I?ve got >> several phones with STUN enabled behind a NAT firewall (namely >> pfSense). I?m not sure how those things might relate but maybe it?s >> worth mentionning. >> >> I?ve decided to update a few hours ago but still crashing. >> >> I was about to test mod_easyroute when I decided to run gdb and look >> into it : http://pastebin.freeswitch.org/6830 >> >> Could someone please help me pointing the issue? J >> >> Thanks in advance, >> >> -- Laurent FABRE >> Directeur g?n?ral >> 10, rue d'Aumale >> 75009 Paris >> Tel: +33.(0)1.42.81.28.20 >> Mob: +33.(0)6.75.75.02.96 >> Fax: +33.(0)1.70.24.74.61 <+33170247461> >> laurent.fabre at kirranet.com >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Laurent.Fabre at kirranet.com Tue Jan 20 20:45:50 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Wed, 21 Jan 2009 05:45:50 +0100 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed In-Reply-To: References: Message-ID: http://jira.freeswitch.org/browse/FSCORE-276 Done. -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61<+33170247461> laurent.fabre at kirranet.com De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Brian West Envoy? : mercredi 21 janvier 2009 04:56 ? : freeswitch-users at lists.freeswitch.org Objet : Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed Please open a jira http://jira.freeswitch.org include all the requested data in detail. /b On Jan 20, 2009, at 9:57 PM, Laurent Fabre wrote: Hi, I'm experiencing a few ? random ? segfaults ever since I enabled core ODBC support. They usually happen during startup or quickly after...or not at all as long as I do not restart freeswitch. The backend is a postgresql 7.4, every module has a distinct database (LATIN9, with OIDS). I'm having a few issues with presence & call routing which tend to break from time to time. The freeswitch box is hosted and I've got several phones with STUN enabled behind a NAT firewall (namely pfSense). I'm not sure how those things might relate but maybe it's worth mentionning. I've decided to update a few hours ago but still crashing. I was about to test mod_easyroute when I decided to run gdb and look into it : http://pastebin.freeswitch.org/6830 Could someone please help me pointing the issue? :) Thanks in advance, -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61<+33170247461> laurent.fabre at kirranet.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/d888ec15/attachment.html From msc at freeswitch.org Tue Jan 20 21:11:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Jan 2009 21:11:43 -0800 Subject: [Freeswitch-users] New wiki page needs your attention! Message-ID: <87f2f3b90901202111x3811ba36ve2ec3ecda31c6855@mail.gmail.com> Hello FreeSWITCHers! I'm putting the finishing touches on a wiki page that we hope will make it easier for users to request help and for the dev team and power users to digest and process those requests. What I need first and foremost is for everyone to please read this page: http://wiki.freeswitch.org/wiki/Reporting_Bugs Please give me feedback. Put yourself in the shoes of a relative newbie. Is the information easy to follow? On the flip side, if you wanted to help someone, ask yourself, if they follow the steps on this page will that suffice? Are there places that need improvement? Can you think of anything else that can be added? NOTE: I'm still working on the TDM/OpenZAP section as well as the sections on scripting, event socket, elements of a jira ticket, etc. If you have suggestions for content on those pages please email me off list or hop and and fill in some of the blanks. The core development team really appreciates all of your help. Now that FS is growing like mad we are at the point where it is imperative that we have reliable documentation for new ones so that the developers and other experts can focus on advancing the project even further. Let's all lend a hand by improving the documentation. Thanks again! -MC (mercutioviz) From msc at freeswitch.org Tue Jan 20 21:12:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Jan 2009 21:12:30 -0800 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed In-Reply-To: References: Message-ID: <87f2f3b90901202112n66793061q8b0724a3aaf8619d@mail.gmail.com> Merci beaucoup! The devs will keep an eye on this and will hopefully have a resolution soon. -MC On Tue, Jan 20, 2009 at 8:45 PM, Laurent Fabre wrote: > http://jira.freeswitch.org/browse/FSCORE-276 > > > > Done. > > > > -- Laurent FABRE > Directeur g?n?ral > 10, rue d'Aumale > 75009 Paris > Tel: +33.(0)1.42.81.28.20 > Mob: +33.(0)6.75.75.02.96 > Fax: +33.(0)1.70.24.74.61 > laurent.fabre at kirranet.com > > > > De : freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Brian > West > Envoy? : mercredi 21 janvier 2009 04:56 > ? : freeswitch-users at lists.freeswitch.org > Objet : Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > > > Please open a jira http://jira.freeswitch.org include all the requested data > in detail. > > > > /b > > > > On Jan 20, 2009, at 9:57 PM, Laurent Fabre wrote: > > Hi, > > > > I'm experiencing a few ? random ? segfaults ever since I enabled core ODBC > support. They usually happen during startup or quickly after?or not at all > as long as I do not restart freeswitch. > > > > The backend is a postgresql 7.4, every module has a distinct database > (LATIN9, with OIDS). > > > > I'm having a few issues with presence & call routing which tend to break > from time to time. The freeswitch box is hosted and I've got several phones > with STUN enabled behind a NAT firewall (namely pfSense). I'm not sure how > those things might relate but maybe it's worth mentionning. > > > > I've decided to update a few hours ago but still crashing. > > > > I was about to test mod_easyroute when I decided to run gdb and look into > it : http://pastebin.freeswitch.org/6830 > > > > Could someone please help me pointing the issue? J > > > > Thanks in advance, > > > > -- Laurent FABRE > Directeur g?n?ral > 10, rue d'Aumale > 75009 Paris > Tel: +33.(0)1.42.81.28.20 > Mob: +33.(0)6.75.75.02.96 > Fax: +33.(0)1.70.24.74.61 > laurent.fabre at kirranet.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From krice at freeswitch.org Tue Jan 20 21:58:44 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 20 Jan 2009 23:58:44 -0600 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed In-Reply-To: <87f2f3b90901202112n66793061q8b0724a3aaf8619d@mail.gmail.com> Message-ID: I think I found the problem here... Not 100%... But I did find an issue in switch_odbc.c... I'll have to talk to Tony about this one... There is a condition where we can return an error and we shouldn't > From: Michael Collins > Reply-To: > Date: Tue, 20 Jan 2009 21:12:30 -0800 > To: > Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > Merci beaucoup! The devs will keep an eye on this and will hopefully > have a resolution soon. > -MC > > On Tue, Jan 20, 2009 at 8:45 PM, Laurent Fabre > wrote: >> http://jira.freeswitch.org/browse/FSCORE-276 >> >> >> >> Done. >> >> >> >> -- Laurent FABRE >> Directeur g?n?ral >> 10, rue d'Aumale >> 75009 Paris >> Tel: +33.(0)1.42.81.28.20 >> Mob: +33.(0)6.75.75.02.96 >> Fax: +33.(0)1.70.24.74.61 >> laurent.fabre at kirranet.com >> >> >> >> De : freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Brian >> West >> Envoy? : mercredi 21 janvier 2009 04:56 >> ? : freeswitch-users at lists.freeswitch.org >> Objet : Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed >> >> >> >> Please open a jira http://jira.freeswitch.org include all the requested data >> in detail. >> >> >> >> /b >> >> >> >> On Jan 20, 2009, at 9:57 PM, Laurent Fabre wrote: >> >> Hi, >> >> >> >> I'm experiencing a few ? random ? segfaults ever since I enabled core ODBC >> support. They usually happen during startup or quickly after?or not at all >> as long as I do not restart freeswitch. >> >> >> >> The backend is a postgresql 7.4, every module has a distinct database >> (LATIN9, with OIDS). >> >> >> >> I'm having a few issues with presence & call routing which tend to break >> from time to time. The freeswitch box is hosted and I've got several phones >> with STUN enabled behind a NAT firewall (namely pfSense). I'm not sure how >> those things might relate but maybe it's worth mentionning. >> >> >> >> I've decided to update a few hours ago but still crashing. >> >> >> >> I was about to test mod_easyroute when I decided to run gdb and look into >> it : http://pastebin.freeswitch.org/6830 >> >> >> >> Could someone please help me pointing the issue? J >> >> >> >> Thanks in advance, >> >> >> >> -- Laurent FABRE >> Directeur g?n?ral >> 10, rue d'Aumale >> 75009 Paris >> Tel: +33.(0)1.42.81.28.20 >> Mob: +33.(0)6.75.75.02.96 >> Fax: +33.(0)1.70.24.74.61 >> laurent.fabre at kirranet.com >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ravriel_1 at yahoo.com Tue Jan 20 22:21:41 2009 From: ravriel_1 at yahoo.com (Ron Avriel) Date: Tue, 20 Jan 2009 22:21:41 -0800 (PST) Subject: [Freeswitch-users] FreeSWITCH Windows Service does not Start References: <150416.65876.qm@web45202.mail.sp1.yahoo.com> Message-ID: <118825.1733.qm@web45216.mail.sp1.yahoo.com> Thanks for answer. The problem is missing permission for the network service account to FreeSWITCH directory. Once permission is granted FreeSWITCH service starts OK. However, I think that unless there's a way to grant that permission inside the service installation code ("freeswitch -install") then the service should be configured to run as local system account. Otherwise, I think it will cause a lot of trouble to all. Ron ----- Original Message ---- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, January 20, 2009 6:08:25 PM > Subject: Re: [Freeswitch-users] FreeSWITCH Windows Service does not Start > > One issue with the service is we have no console to dump errors too, > it sounds like it is failing one of the startup requirements like > config files being there. Are you able to start it in non service > mode? If so, check permissions on the freeswitch dir that the user > running the service has permissions to that dir. > > Mike > > On Jan 20, 2009, at 8:52 AM, Ron Avriel wrote: > > > Hi, > > > > If I try starting FreeSWITCH Windows service it immediately fails > > with a messagebox: > > > > "The FreeSWITCH service on local computer started and then stopped. > > Some services stop automatically...etc." > > > > I noticed that the service is installed to log on as "NT AUTHORITY > > \NetworkService". > > If I change this to "Local System account" then FreeSWITCH starts > > and runs OK. > > > > This failure occurred on multiple Windows XP servers. > > > > Why does it fail and why does it not use the local system account > > like almost all services? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Laurent.Fabre at kirranet.com Tue Jan 20 22:56:32 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Wed, 21 Jan 2009 07:56:32 +0100 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed In-Reply-To: References: <87f2f3b90901202112n66793061q8b0724a3aaf8619d@mail.gmail.com> Message-ID: Hi Ken, Thanks for looking into it. It just crashed again on the voicemails table after running like a charm for 1 hour: http://pastebin.freeswitch.org/6832 If I can be of any assistance, please let me know. Regards, -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61 laurent.fabre at kirranet.com -----Message d'origine----- De?: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Ken Rice Envoy??: mercredi 21 janvier 2009 06:59 ??: freeswitch-users at lists.freeswitch.org Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed I think I found the problem here... Not 100%... But I did find an issue in switch_odbc.c... I'll have to talk to Tony about this one... There is a condition where we can return an error and we shouldn't > From: Michael Collins > Reply-To: > Date: Tue, 20 Jan 2009 21:12:30 -0800 > To: > Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > Merci beaucoup! The devs will keep an eye on this and will hopefully > have a resolution soon. > -MC > > On Tue, Jan 20, 2009 at 8:45 PM, Laurent Fabre > wrote: >> http://jira.freeswitch.org/browse/FSCORE-276 >> >> >> >> Done. >> >> >> >> -- Laurent FABRE >> Directeur g?n?ral >> 10, rue d'Aumale >> 75009 Paris >> Tel: +33.(0)1.42.81.28.20 >> Mob: +33.(0)6.75.75.02.96 >> Fax: +33.(0)1.70.24.74.61 >> laurent.fabre at kirranet.com >> >> >> >> De : freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Brian >> West >> Envoy? : mercredi 21 janvier 2009 04:56 >> ? : freeswitch-users at lists.freeswitch.org >> Objet : Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed >> >> >> >> Please open a jira http://jira.freeswitch.org include all the requested data >> in detail. >> >> >> >> /b >> >> >> >> On Jan 20, 2009, at 9:57 PM, Laurent Fabre wrote: >> >> Hi, >> >> >> >> I'm experiencing a few ? random ? segfaults ever since I enabled core ODBC >> support. They usually happen during startup or quickly after?or not at all >> as long as I do not restart freeswitch. >> >> >> >> The backend is a postgresql 7.4, every module has a distinct database >> (LATIN9, with OIDS). >> >> >> >> I'm having a few issues with presence & call routing which tend to break >> from time to time. The freeswitch box is hosted and I've got several phones >> with STUN enabled behind a NAT firewall (namely pfSense). I'm not sure how >> those things might relate but maybe it's worth mentionning. >> >> >> >> I've decided to update a few hours ago but still crashing. >> >> >> >> I was about to test mod_easyroute when I decided to run gdb and look into >> it : http://pastebin.freeswitch.org/6830 >> >> >> >> Could someone please help me pointing the issue? J >> >> >> >> Thanks in advance, >> >> >> >> -- Laurent FABRE >> Directeur g?n?ral >> 10, rue d'Aumale >> 75009 Paris >> Tel: +33.(0)1.42.81.28.20 >> Mob: +33.(0)6.75.75.02.96 >> Fax: +33.(0)1.70.24.74.61 >> laurent.fabre at kirranet.com >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From krice at suspicious.org Tue Jan 20 22:50:28 2009 From: krice at suspicious.org (Ken Rice) Date: Wed, 21 Jan 2009 00:50:28 -0600 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed In-Reply-To: Message-ID: Can you try that attached patch? K > From: Laurent Fabre > Reply-To: > Date: Wed, 21 Jan 2009 07:56:32 +0100 > To: "freeswitch-users at lists.freeswitch.org" > > Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > Hi Ken, > > Thanks for looking into it. > > It just crashed again on the voicemails table after running like a charm for 1 > hour: http://pastebin.freeswitch.org/6832 > > If I can be of any assistance, please let me know. > > Regards, > > -- Laurent FABRE > Directeur g?n?ral > 10, rue d'Aumale > 75009 Paris > Tel: +33.(0)1.42.81.28.20 > Mob: +33.(0)6.75.75.02.96 > Fax: +33.(0)1.70.24.74.61 > laurent.fabre at kirranet.com > > -----Message d'origine----- > De?: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Ken Rice > Envoy??: mercredi 21 janvier 2009 06:59 > ??: freeswitch-users at lists.freeswitch.org > Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > I think I found the problem here... Not 100%... But I did find an issue in > switch_odbc.c... > > I'll have to talk to Tony about this one... There is a condition where we > can return an error and we shouldn't > > >> From: Michael Collins >> Reply-To: >> Date: Tue, 20 Jan 2009 21:12:30 -0800 >> To: >> Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed >> >> Merci beaucoup! The devs will keep an eye on this and will hopefully >> have a resolution soon. >> -MC >> >> On Tue, Jan 20, 2009 at 8:45 PM, Laurent Fabre >> wrote: >>> http://jira.freeswitch.org/browse/FSCORE-276 >>> >>> >>> >>> Done. >>> >>> >>> >>> -- Laurent FABRE >>> Directeur g?n?ral >>> 10, rue d'Aumale >>> 75009 Paris >>> Tel: +33.(0)1.42.81.28.20 >>> Mob: +33.(0)6.75.75.02.96 >>> Fax: +33.(0)1.70.24.74.61 >>> laurent.fabre at kirranet.com >>> >>> >>> >>> De : freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Brian >>> West >>> Envoy? : mercredi 21 janvier 2009 04:56 >>> ? : freeswitch-users at lists.freeswitch.org >>> Objet : Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed >>> >>> >>> >>> Please open a jira http://jira.freeswitch.org include all the requested data >>> in detail. >>> >>> >>> >>> /b >>> >>> >>> >>> On Jan 20, 2009, at 9:57 PM, Laurent Fabre wrote: >>> >>> Hi, >>> >>> >>> >>> I'm experiencing a few ? random ? segfaults ever since I enabled core ODBC >>> support. They usually happen during startup or quickly after?or not at all >>> as long as I do not restart freeswitch. >>> >>> >>> >>> The backend is a postgresql 7.4, every module has a distinct database >>> (LATIN9, with OIDS). >>> >>> >>> >>> I'm having a few issues with presence & call routing which tend to break >>> from time to time. The freeswitch box is hosted and I've got several phones >>> with STUN enabled behind a NAT firewall (namely pfSense). I'm not sure how >>> those things might relate but maybe it's worth mentionning. >>> >>> >>> >>> I've decided to update a few hours ago but still crashing. >>> >>> >>> >>> I was about to test mod_easyroute when I decided to run gdb and look into >>> it : http://pastebin.freeswitch.org/6830 >>> >>> >>> >>> Could someone please help me pointing the issue? J >>> >>> >>> >>> Thanks in advance, >>> >>> >>> >>> -- Laurent FABRE >>> Directeur g?n?ral >>> 10, rue d'Aumale >>> 75009 Paris >>> Tel: +33.(0)1.42.81.28.20 >>> Mob: +33.(0)6.75.75.02.96 >>> Fax: +33.(0)1.70.24.74.61 >>> laurent.fabre at kirranet.com >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: switch_odbc.patch Type: application/octet-stream Size: 626 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/6d11b2d7/attachment.obj From stevecrozz at gmail.com Tue Jan 20 23:04:13 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Tue, 20 Jan 2009 23:04:13 -0800 Subject: [Freeswitch-users] firing events from javascript - working example needed Message-ID: <11990ade0901202304k1d3b6bf6h75d4cf677eaf29ee@mail.gmail.com> 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 From pmhshz at gmail.com Tue Jan 20 23:10:38 2009 From: pmhshz at gmail.com (shehzad p) Date: Tue, 20 Jan 2009 23:10:38 -0800 (PST) Subject: [Freeswitch-users] Method getVariable cause error on FS 1.0.2 in javascript Message-ID: <21578116.post@talk.nabble.com> Hi all, I am using the javascript for call termination, and after call complete, I use getVariable mehtod to get some values like PDD. Below is my javascript that perform this: ============================================================ dialstr="sofia/outbound/1111 at xxx.xxx.xx.xx"; newsession.originate(session, dialstr); if(session.ready() && newsession.ready()) bridge(session, newsession); if(newsession.getVariable("progressmsec") != false) //in FS 1.0.2 and SVN version SCRIPT TERMINATES on above line pdd = newsession.getVariable("progressmsec"); else pdd = 0; ................. ========================================================== Above script works fine in FS 1.0.1. Now I am testing FS 1.0.2, It also works fine when call is being ANSWERED, PDD is correctly collected, BUT when Call is not routed successfully due to some network problem, or User No-Answer / Busy, then below error occurs: ------------------------------------------------------------------------------------------------------------------------------------------------------- 2009-01-19 19:40:43 [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! -------------------------------------------------------------------------------------------------------------------------------------------------------- And after this error, script DOES NOT continue execution and TERMINATES. I also tested SVN version (revision 11266), but same error occurs there too. What should be the cause here, so this behavior of FS is changed from 1.0.1 to 1.0.2. Thanks. msp -- View this message in context: http://www.nabble.com/Method-getVariable-cause-error-on-FS-1.0.2-in-javascript-tp21578116p21578116.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Laurent.Fabre at kirranet.com Tue Jan 20 23:27:08 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Wed, 21 Jan 2009 08:27:08 +0100 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed In-Reply-To: References: Message-ID: Applied the patch, compiled, restarted. So far so good. Everything looks OK. Registrations, subscriptions, calls are working as expected. I have an issue with the presence on one of the softphone (presence.wminfo?) but it's probably unrelated and just a glitch in Bria for Outlook. Keep you posted. -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61 laurent.fabre at kirranet.com -----Message d'origine----- De?: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Ken Rice Envoy??: mercredi 21 janvier 2009 07:50 ??: freeswitch-users at lists.freeswitch.org Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed Can you try that attached patch? K > From: Laurent Fabre > Reply-To: > Date: Wed, 21 Jan 2009 07:56:32 +0100 > To: "freeswitch-users at lists.freeswitch.org" > > Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC > crashed > > Hi Ken, > > Thanks for looking into it. > > It just crashed again on the voicemails table after running like a > charm for 1 > hour: http://pastebin.freeswitch.org/6832 > > If I can be of any assistance, please let me know. > > Regards, > > -- Laurent FABRE > Directeur g?n?ral > 10, rue d'Aumale > 75009 Paris > Tel: +33.(0)1.42.81.28.20 > Mob: +33.(0)6.75.75.02.96 > Fax: +33.(0)1.70.24.74.61 > laurent.fabre at kirranet.com > > -----Message d'origine----- > De?: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de > Ken Rice Envoy??: mercredi 21 janvier 2009 06:59 ??: > freeswitch-users at lists.freeswitch.org > Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > I think I found the problem here... Not 100%... But I did find an > issue in switch_odbc.c... > > I'll have to talk to Tony about this one... There is a condition where > we can return an error and we shouldn't > > >> From: Michael Collins >> Reply-To: >> Date: Tue, 20 Jan 2009 21:12:30 -0800 >> To: >> Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC >> crashed >> >> Merci beaucoup! The devs will keep an eye on this and will hopefully >> have a resolution soon. >> -MC >> >> On Tue, Jan 20, 2009 at 8:45 PM, Laurent Fabre >> wrote: >>> http://jira.freeswitch.org/browse/FSCORE-276 >>> >>> >>> >>> Done. >>> >>> >>> >>> -- Laurent FABRE >>> Directeur g?n?ral >>> 10, rue d'Aumale >>> 75009 Paris >>> Tel: +33.(0)1.42.81.28.20 >>> Mob: +33.(0)6.75.75.02.96 >>> Fax: +33.(0)1.70.24.74.61 >>> laurent.fabre at kirranet.com >>> >>> >>> >>> De : freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de >>> Brian West Envoy? : mercredi 21 janvier 2009 04:56 ? : >>> freeswitch-users at lists.freeswitch.org >>> Objet : Re: [Freeswitch-users] Freeswitch latest revision ODBC >>> crashed >>> >>> >>> >>> Please open a jira http://jira.freeswitch.org include all the >>> requested data in detail. >>> >>> >>> >>> /b >>> >>> >>> >>> On Jan 20, 2009, at 9:57 PM, Laurent Fabre wrote: >>> >>> Hi, >>> >>> >>> >>> I'm experiencing a few ? random ? segfaults ever since I enabled >>> core ODBC support. They usually happen during startup or quickly >>> after?or not at all as long as I do not restart freeswitch. >>> >>> >>> >>> The backend is a postgresql 7.4, every module has a distinct >>> database (LATIN9, with OIDS). >>> >>> >>> >>> I'm having a few issues with presence & call routing which tend to >>> break from time to time. The freeswitch box is hosted and I've got >>> several phones with STUN enabled behind a NAT firewall (namely >>> pfSense). I'm not sure how those things might relate but maybe it's worth mentionning. >>> >>> >>> >>> I've decided to update a few hours ago but still crashing. >>> >>> >>> >>> I was about to test mod_easyroute when I decided to run gdb and look >>> into it : http://pastebin.freeswitch.org/6830 >>> >>> >>> >>> Could someone please help me pointing the issue? J >>> >>> >>> >>> Thanks in advance, >>> >>> >>> >>> -- Laurent FABRE >>> Directeur g?n?ral >>> 10, rue d'Aumale >>> 75009 Paris >>> Tel: +33.(0)1.42.81.28.20 >>> Mob: +33.(0)6.75.75.02.96 >>> Fax: +33.(0)1.70.24.74.61 >>> laurent.fabre at kirranet.com >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>> sers >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>> sers >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org From krice at suspicious.org Tue Jan 20 23:22:04 2009 From: krice at suspicious.org (Ken Rice) Date: Wed, 21 Jan 2009 01:22:04 -0600 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed In-Reply-To: Message-ID: I'm not sure that patch will work... It works for me on an unrelated thing (mod_easyroute) so hopefully it will work for you here K > From: Laurent Fabre > Reply-To: > Date: Wed, 21 Jan 2009 08:27:08 +0100 > To: "freeswitch-users at lists.freeswitch.org" > > Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > Applied the patch, compiled, restarted. > > So far so good. Everything looks OK. Registrations, subscriptions, calls are > working as expected. > > I have an issue with the presence on one of the softphone (presence.wminfo?) > but it's probably unrelated and just a glitch in Bria for Outlook. > > Keep you posted. > > -- Laurent FABRE > Directeur g?n?ral > 10, rue d'Aumale > 75009 Paris > Tel: +33.(0)1.42.81.28.20 > Mob: +33.(0)6.75.75.02.96 > Fax: +33.(0)1.70.24.74.61 > laurent.fabre at kirranet.com > > > -----Message d'origine----- > De?: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Ken Rice > Envoy??: mercredi 21 janvier 2009 07:50 > ??: freeswitch-users at lists.freeswitch.org > Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > Can you try that attached patch? > > K > > >> From: Laurent Fabre >> Reply-To: >> Date: Wed, 21 Jan 2009 07:56:32 +0100 >> To: "freeswitch-users at lists.freeswitch.org" >> >> Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC >> crashed >> >> Hi Ken, >> >> Thanks for looking into it. >> >> It just crashed again on the voicemails table after running like a >> charm for 1 >> hour: http://pastebin.freeswitch.org/6832 >> >> If I can be of any assistance, please let me know. >> >> Regards, >> >> -- Laurent FABRE >> Directeur g?n?ral >> 10, rue d'Aumale >> 75009 Paris >> Tel: +33.(0)1.42.81.28.20 >> Mob: +33.(0)6.75.75.02.96 >> Fax: +33.(0)1.70.24.74.61 >> laurent.fabre at kirranet.com >> >> -----Message d'origine----- >> De?: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de >> Ken Rice Envoy??: mercredi 21 janvier 2009 06:59 ??: >> freeswitch-users at lists.freeswitch.org >> Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed >> >> I think I found the problem here... Not 100%... But I did find an >> issue in switch_odbc.c... >> >> I'll have to talk to Tony about this one... There is a condition where >> we can return an error and we shouldn't >> >> >>> From: Michael Collins >>> Reply-To: >>> Date: Tue, 20 Jan 2009 21:12:30 -0800 >>> To: >>> Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC >>> crashed >>> >>> Merci beaucoup! The devs will keep an eye on this and will hopefully >>> have a resolution soon. >>> -MC >>> >>> On Tue, Jan 20, 2009 at 8:45 PM, Laurent Fabre >>> wrote: >>>> http://jira.freeswitch.org/browse/FSCORE-276 >>>> >>>> >>>> >>>> Done. >>>> >>>> >>>> >>>> -- Laurent FABRE >>>> Directeur g?n?ral >>>> 10, rue d'Aumale >>>> 75009 Paris >>>> Tel: +33.(0)1.42.81.28.20 >>>> Mob: +33.(0)6.75.75.02.96 >>>> Fax: +33.(0)1.70.24.74.61 >>>> laurent.fabre at kirranet.com >>>> >>>> >>>> >>>> De : freeswitch-users-bounces at lists.freeswitch.org >>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de >>>> Brian West Envoy? : mercredi 21 janvier 2009 04:56 ? : >>>> freeswitch-users at lists.freeswitch.org >>>> Objet : Re: [Freeswitch-users] Freeswitch latest revision ODBC >>>> crashed >>>> >>>> >>>> >>>> Please open a jira http://jira.freeswitch.org include all the >>>> requested data in detail. >>>> >>>> >>>> >>>> /b >>>> >>>> >>>> >>>> On Jan 20, 2009, at 9:57 PM, Laurent Fabre wrote: >>>> >>>> Hi, >>>> >>>> >>>> >>>> I'm experiencing a few ? random ? segfaults ever since I enabled >>>> core ODBC support. They usually happen during startup or quickly >>>> after?or not at all as long as I do not restart freeswitch. >>>> >>>> >>>> >>>> The backend is a postgresql 7.4, every module has a distinct >>>> database (LATIN9, with OIDS). >>>> >>>> >>>> >>>> I'm having a few issues with presence & call routing which tend to >>>> break from time to time. The freeswitch box is hosted and I've got >>>> several phones with STUN enabled behind a NAT firewall (namely >>>> pfSense). I'm not sure how those things might relate but maybe it's worth >>>> mentionning. >>>> >>>> >>>> >>>> I've decided to update a few hours ago but still crashing. >>>> >>>> >>>> >>>> I was about to test mod_easyroute when I decided to run gdb and look >>>> into it : http://pastebin.freeswitch.org/6830 >>>> >>>> >>>> >>>> Could someone please help me pointing the issue? J >>>> >>>> >>>> >>>> Thanks in advance, >>>> >>>> >>>> >>>> -- Laurent FABRE >>>> Directeur g?n?ral >>>> 10, rue d'Aumale >>>> 75009 Paris >>>> Tel: +33.(0)1.42.81.28.20 >>>> Mob: +33.(0)6.75.75.02.96 >>>> Fax: +33.(0)1.70.24.74.61 >>>> laurent.fabre at kirranet.com >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>>> sers >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>>> sers >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> ers >>> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> _______________________________________________ > Freeswitch-users mailing >> list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman >> /listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >> ions/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org From krzysiez at go2.pl Wed Jan 21 00:55:08 2009 From: krzysiez at go2.pl (=?UTF-8?Q?Krzysztof_Zimnicki?=) Date: Wed, 21 Jan 2009 09:55:08 +0100 Subject: [Freeswitch-users] =?utf-8?q?Problem_with_digium_te220p?= Message-ID: <43dfd602.2741ece3.4976e2ec.94b1f@o2.pl> >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. From tomasborrella at gmail.com Wed Jan 21 01:15:56 2009 From: tomasborrella at gmail.com (=?ISO-8859-1?Q?Tom=E1s?=) Date: Wed, 21 Jan 2009 10:15:56 +0100 Subject: [Freeswitch-users] Hang up not received In-Reply-To: <4468a6770901201343q3a51a493m2b721d4531346a77@mail.gmail.com> References: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> <191c3a030901200905r156a0b79uad5baa726eeeb7c0@mail.gmail.com> <4468a6770901201145m444a2c82gacaffd369579a47c@mail.gmail.com> <4468a6770901201343q3a51a493m2b721d4531346a77@mail.gmail.com> Message-ID: <46336fde0901210115o7645c261o732fe939cfc2380e@mail.gmail.com> Scott, I imagined that it could be an OpenZap problem, but I didn't find an OpenZap mailing list, so I sent the email to FS list. Do you know where can I find more information about OpenZap hardware support and developement status (I have special interest in Loop Start)?? Anthony and Ognjen, I've tried tone detection and thanks to that FS is detecting hung up, but I faced the problem that tone detector answer the call... That's my dialplan: When I receive a call from PSTN, tone detection answer the call (the caller hears only one first tone and then hears "nothing" until I pick up the call on softphone). So, I think that tone detection solution does not resolve my problem... Is there any other possibility to detect hang up without answering the call (using Loop Start signaling) or have we to wait until OpenZap is completely developed? Thanks in advance. On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija wrote: > Ok, as discussed with Tony on IRC channel I followed his directions which > lead to a successfull outcome (like it always does I might add :). > > One has to use tone_detect app in FreeSWITCH dialplan in order to check for > busy tones coming from the PSTN side and if matched fire a hangup > application. This is the snippet of my test dp that does the trick (from > extension Local_extensions in default.xml): > > > > > > This means that FS will listen to freq of 425 Hz and wait for 4 positive > detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 > Hz is the freq telco here uses; for other countries I suggest getting the > ITU world tones pdf file and check there): > > 2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() > TONE busy HIT 1/4 > 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() > TONE busy HIT 2/4 > 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() > TONE busy HIT 3/4 > 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() > TONE busy HIT 4/4 > 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() > TONE busy DETECTED > 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup > OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING] > > Regards, > Ognjen > > On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija wrote: > >> I tried similar setup with my analog card (X100P) and I'm having same >> issue. Call is not hungup on the oz side once the caller ends. My telco >> doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck >> to detecting busy tone from the telco side. I'll try to modify tones.conf >> accordingly. >> >> Regards, >> Ognjen >> (sekil) >> On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> This is a common issue with analog phones even traditional answering >>> machines suffer from it. >>> I'm sure you must have had an answering machine at some point that has >>> dial tone as the message it receives. >>> >>> Unless FreeSWITCH has some hint that the call has hungup it will not stop >>> trying to complete the call. >>> >>> If the other side is sending a busy tone to indicate hangup it's possible >>> to use the tone_detect app to pick >>> up on the tones and abort the call. >>> >>> Another thing you could do if you have unlimited inbound is explicitly >>> answer the call in the dialplan before >>> you call your sip phones this will give you a more profound hangup >>> detection but it will make every call count >>> even when nobody answers. >>> >>> >>> >>> On Tue, Jan 20, 2009 at 10:46 AM, Tom?s wrote: >>> >>>> 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. >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 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 >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/2a62a3ad/attachment-0001.html From tomasborrella at gmail.com Wed Jan 21 01:23:31 2009 From: tomasborrella at gmail.com (=?ISO-8859-1?Q?Tom=E1s?=) Date: Wed, 21 Jan 2009 10:23:31 +0100 Subject: [Freeswitch-users] New wiki page needs your attention! In-Reply-To: <87f2f3b90901202111x3811ba36ve2ec3ecda31c6855@mail.gmail.com> References: <87f2f3b90901202111x3811ba36ve2ec3ecda31c6855@mail.gmail.com> Message-ID: <46336fde0901210123i79743875xf7ca530104154f9c@mail.gmail.com> Thanks Michael, I think that's very useful information for newbies like me... On the other hand, I'm looking forward to the OpenZap section because I'm having some problems with my card, and maybe in the new section I could find useful information... And If you need a tester with a X101P card, you can count on me to do that. On Wed, Jan 21, 2009 at 6:11 AM, Michael Collins wrote: > Hello FreeSWITCHers! > > I'm putting the finishing touches on a wiki page that we hope will > make it easier for users to request help and for the dev team and > power users to digest and process those requests. What I need first > and foremost is for everyone to please read this page: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Please give me feedback. Put yourself in the shoes of a relative > newbie. Is the information easy to follow? On the flip side, if you > wanted to help someone, ask yourself, if they follow the steps on this > page will that suffice? Are there places that need improvement? Can > you think of anything else that can be added? > > NOTE: I'm still working on the TDM/OpenZAP section as well as the > sections on scripting, event socket, elements of a jira ticket, etc. > If you have suggestions for content on those pages please email me off > list or hop and and fill in some of the blanks. > > The core development team really appreciates all of your help. Now > that FS is growing like mad we are at the point where it is imperative > that we have reliable documentation for new ones so that the > developers and other experts can focus on advancing the project even > further. Let's all lend a hand by improving the documentation. > > Thanks again! > -MC (mercutioviz) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/cdd0e8f4/attachment.html From scott.ellis at novatex.com.au Wed Jan 21 01:36:11 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Wed, 21 Jan 2009 20:36:11 +1100 Subject: [Freeswitch-users] Hang up not received In-Reply-To: <46336fde0901210115o7645c261o732fe939cfc2380e@mail.gmail.com> References: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> <191c3a030901200905r156a0b79uad5baa726eeeb7c0@mail.gmail.com> <4468a6770901201145m444a2c82gacaffd369579a47c@mail.gmail.com> <4468a6770901201343q3a51a493m2b721d4531346a77@mail.gmail.com> <46336fde0901210115o7645c261o732fe939cfc2380e@mail.gmail.com> Message-ID: <4976EC8B.3080606@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/88f7f25c/attachment.html From oseslija at gmail.com Wed Jan 21 01:49:19 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Wed, 21 Jan 2009 10:49:19 +0100 Subject: [Freeswitch-users] Hang up not received In-Reply-To: <4976EC8B.3080606@novatex.com.au> References: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> <191c3a030901200905r156a0b79uad5baa726eeeb7c0@mail.gmail.com> <4468a6770901201145m444a2c82gacaffd369579a47c@mail.gmail.com> <4468a6770901201343q3a51a493m2b721d4531346a77@mail.gmail.com> <46336fde0901210115o7645c261o732fe939cfc2380e@mail.gmail.com> <4976EC8B.3080606@novatex.com.au> Message-ID: <4468a6770901210149l3afb036eo7e7d1151baecc37e@mail.gmail.com> When call comes in from Openzap, tone_detect app does pre_answer of a call cause it's need media to start detecting tones in the first place. This behaviour is something that I see on calls inside my telco when calling from analogue lines. I don't think this is a big of deal because ringback provided by FS will make caller understand that the call is still in progress. One can make its own ringback to sound exactly the same as telco's. I don't think that we'll ever make POTS behave like digital protocols do. Regards, Ognjen On Wed, Jan 21, 2009 at 10:36 AM, Scott Ellis wrote: > I had a similar problem, you can use > (I added an "au" > ring definition to my vars.xml file) > > To get what you want. > > I also had a problem that you get two rings, then an answer then to the > system generated ring tone, which was confusing some of our (not to bright) > callers. > > As we don't use callerID I turned that flag off in the openzap.conf.xml > file - I thought that this would do what I wanted (answer the instant the > call is detected), but the change in the config file does not make it all > the way down to the point where it takes action. At this point I hacked the > code to get what I wanted. I have to create a JIRA entry with the details > yet. > > As far as I understand, this is the right place for OpenZap, as it is a > product of the FS project. > > Scott > > Tom?s wrote: > > Scott, I imagined that it could be an OpenZap problem, but I didn't find an > OpenZap mailing list, so I sent the email to FS list. Do you know where can > I find more information about OpenZap hardware support and developement > status (I have special interest in Loop Start)?? > > Anthony and Ognjen, I've tried tone detection and thanks to that FS is > detecting hung up, but I faced the problem that tone detector answer the > call... > > That's my dialplan: > > > > > data="sofia/internal/1003%${server-domain-name}, > sofia/internal/1004%${server-domain-name}"/> > > > > When I receive a call from PSTN, tone detection answer the call (the caller > hears only one first tone and then hears "nothing" until I pick up the call > on softphone). > > So, I think that tone detection solution does not resolve my problem... Is > there any other possibility to detect hang up without answering the call > (using Loop Start signaling) or have we to wait until OpenZap is completely > developed? > > Thanks in advance. > > On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija wrote: > >> Ok, as discussed with Tony on IRC channel I followed his directions which >> lead to a successfull outcome (like it always does I might add :). >> >> One has to use tone_detect app in FreeSWITCH dialplan in order to check >> for busy tones coming from the PSTN side and if matched fire a hangup >> application. This is the snippet of my test dp that does the trick (from >> extension Local_extensions in default.xml): >> >> >> >> > >> This means that FS will listen to freq of 425 Hz and wait for 4 positive >> detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 >> Hz is the freq telco here uses; for other countries I suggest getting the >> ITU world tones pdf file and check there): >> >> 2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 1/4 >> 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 2/4 >> 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 3/4 >> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 4/4 >> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() >> TONE busy DETECTED >> 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup >> OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING] >> >> Regards, >> Ognjen >> >> On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija wrote: >> >>> I tried similar setup with my analog card (X100P) and I'm having same >>> issue. Call is not hungup on the oz side once the caller ends. My telco >>> doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck >>> to detecting busy tone from the telco side. I'll try to modify tones.conf >>> accordingly. >>> >>> Regards, >>> Ognjen >>> (sekil) >>> On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> This is a common issue with analog phones even traditional answering >>>> machines suffer from it. >>>> I'm sure you must have had an answering machine at some point that has >>>> dial tone as the message it receives. >>>> >>>> Unless FreeSWITCH has some hint that the call has hungup it will not >>>> stop trying to complete the call. >>>> >>>> If the other side is sending a busy tone to indicate hangup it's >>>> possible to use the tone_detect app to pick >>>> up on the tones and abort the call. >>>> >>>> Another thing you could do if you have unlimited inbound is explicitly >>>> answer the call in the dialplan before >>>> you call your sip phones this will give you a more profound hangup >>>> detection but it will make every call count >>>> even when nobody answers. >>>> >>>> >>>> >>>> On Tue, Jan 20, 2009 at 10:46 AM, Tom?s wrote: >>>> >>>>> 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. >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 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 >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/0b19e789/attachment-0001.html From helmut.kuper at ewetel.de Wed Jan 21 03:58:45 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 21 Jan 2009 12:58:45 +0100 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <87f2f3b90901191046t6b8ac1e9hfa130e48f25a772b@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> <497485CC.6060906@voiceworks.pl> <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> <87f2f3b90901191046t6b8ac1e9hfa130e48f25a772b@mail.gmail.com> Message-ID: <49770DF5.6070905@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, just an update about my progress in this. Currently I have working C code which creates a pcap file containing all needed protocoll headers in front of the Q931 dump. I use libpcap to create the pcap file. The protocol addresses of each protocol header structure are dynamically set by the code to reflect direction, sequence and timeline of each Q931 packet in whireshark. Wireshark can read and decode the current packets generated by my C code correctly. regards helmut Am 19.01.2009 19:46, schrieb Michael Collins: > Until Helmut gets his script, etc. all set the only choice you have is > to capture the debug output on the command line. > -MC -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl3DfUACgkQ4tZeNddg3dz5QACePWZ6l5SEWcirI8gE4ad4N4tF rP0AoIerlF8nk/IB//eEHqSP8j5C8B5Z =rmAo -----END PGP SIGNATURE----- From mgg at giagnocavo.net Wed Jan 21 04:38:15 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 21 Jan 2009 07:38:15 -0500 Subject: [Freeswitch-users] FreeSWITCH Windows Service does not Start In-Reply-To: <118825.1733.qm@web45216.mail.sp1.yahoo.com> References: <150416.65876.qm@web45202.mail.sp1.yahoo.com> <118825.1733.qm@web45216.mail.sp1.yahoo.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670235F118DE@mse17be1.mse17.exchange.ms> You're right, there should be a full installer system that'll ask what account you want to use, check permissions, etc. -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron Avriel Sent: Tuesday, January 20, 2009 11:22 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSWITCH Windows Service does not Start Thanks for answer. The problem is missing permission for the network service account to FreeSWITCH directory. Once permission is granted FreeSWITCH service starts OK. However, I think that unless there's a way to grant that permission inside the service installation code ("freeswitch -install") then the service should be configured to run as local system account. Otherwise, I think it will cause a lot of trouble to all. Ron ----- Original Message ---- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, January 20, 2009 6:08:25 PM > Subject: Re: [Freeswitch-users] FreeSWITCH Windows Service does not Start > > One issue with the service is we have no console to dump errors too, > it sounds like it is failing one of the startup requirements like > config files being there. Are you able to start it in non service > mode? If so, check permissions on the freeswitch dir that the user > running the service has permissions to that dir. > > Mike > > On Jan 20, 2009, at 8:52 AM, Ron Avriel wrote: > > > Hi, > > > > If I try starting FreeSWITCH Windows service it immediately fails > > with a messagebox: > > > > "The FreeSWITCH service on local computer started and then stopped. > > Some services stop automatically...etc." > > > > I noticed that the service is installed to log on as "NT AUTHORITY > > \NetworkService". > > If I change this to "Local System account" then FreeSWITCH starts > > and runs OK. > > > > This failure occurred on multiple Windows XP servers. > > > > Why does it fail and why does it not use the local system account > > like almost all services? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pmhshz at gmail.com Wed Jan 21 05:40:14 2009 From: pmhshz at gmail.com (shehzad p) Date: Wed, 21 Jan 2009 05:40:14 -0800 (PST) Subject: [Freeswitch-users] How to bridge without Answer? Message-ID: <21583334.post@talk.nabble.com> 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. From msc at freeswitch.org Wed Jan 21 05:48:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 05:48:23 -0800 Subject: [Freeswitch-users] Problem with digium te220p In-Reply-To: <43dfd602.2741ece3.4976e2ec.94b1f@o2.pl> References: <43dfd602.2741ece3.4976e2ec.94b1f@o2.pl> Message-ID: <87f2f3b90901210548q62c880dao12fb57f3ba1ae5f3@mail.gmail.com> can you post your openzap.conf file? -MC On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki wrote: >>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. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Wed Jan 21 05:50:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 05:50:54 -0800 Subject: [Freeswitch-users] New wiki page needs your attention! In-Reply-To: <46336fde0901210123i79743875xf7ca530104154f9c@mail.gmail.com> References: <87f2f3b90901202111x3811ba36ve2ec3ecda31c6855@mail.gmail.com> <46336fde0901210123i79743875xf7ca530104154f9c@mail.gmail.com> Message-ID: <87f2f3b90901210550j153dad55w25aadd72ea5e2710@mail.gmail.com> On Wed, Jan 21, 2009 at 1:23 AM, Tom?s wrote: > Thanks Michael, I think that's very useful information for newbies like > me... > > On the other hand, I'm looking forward to the OpenZap section because I'm > having some problems with my card, and maybe in the new section I could find > useful information... And If you need a tester with a X101P card, you can > count on me to do that. Excellent! Yes, that will be helpful. I will keep plugging away at the reporting bugs page and hopefully soon we'll have the openzap and other sections all done. Also, once you get your X101P all set up and working please be ready to share your setup info on the wiki. -MC > > On Wed, Jan 21, 2009 at 6:11 AM, Michael Collins wrote: >> >> Hello FreeSWITCHers! >> >> I'm putting the finishing touches on a wiki page that we hope will >> make it easier for users to request help and for the dev team and >> power users to digest and process those requests. What I need first >> and foremost is for everyone to please read this page: >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> Please give me feedback. Put yourself in the shoes of a relative >> newbie. Is the information easy to follow? On the flip side, if you >> wanted to help someone, ask yourself, if they follow the steps on this >> page will that suffice? Are there places that need improvement? Can >> you think of anything else that can be added? >> >> NOTE: I'm still working on the TDM/OpenZAP section as well as the >> sections on scripting, event socket, elements of a jira ticket, etc. >> If you have suggestions for content on those pages please email me off >> list or hop and and fill in some of the blanks. >> >> The core development team really appreciates all of your help. Now >> that FS is growing like mad we are at the point where it is imperative >> that we have reliable documentation for new ones so that the >> developers and other experts can focus on advancing the project even >> further. Let's all lend a hand by improving the documentation. >> >> Thanks again! >> -MC (mercutioviz) >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Wed Jan 21 05:52:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 05:52:47 -0800 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <49770DF5.6070905@ewetel.de> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> <497485CC.6060906@voiceworks.pl> <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> <87f2f3b90901191046t6b8ac1e9hfa130e48f25a772b@mail.gmail.com> <49770DF5.6070905@ewetel.de> Message-ID: <87f2f3b90901210552r466ca850xd2961b895d905f78@mail.gmail.com> On Wed, Jan 21, 2009 at 3:58 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > just an update about my progress in this. > > Currently I have working C code which creates a pcap file containing all > needed protocoll headers in front of the Q931 dump. I use libpcap to > create the pcap file. The protocol addresses of each protocol header > structure are dynamically set by the code to reflect direction, sequence > and timeline of each Q931 packet in whireshark. Wireshark can read and > decode the current packets generated by my C code correctly. This is fantastic! How close are you to being ready for others to try it out? Tony and Mike are very interested in seeing this move forward because it will make their job go a lot faster when doing PRI/BRI work. Thanks! -MC > > regards > helmut > > > > Am 19.01.2009 19:46, schrieb Michael Collins: >> Until Helmut gets his script, etc. all set the only choice you have is >> to capture the debug output on the command line. >> -MC > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkl3DfUACgkQ4tZeNddg3dz5QACePWZ6l5SEWcirI8gE4ad4N4tF > rP0AoIerlF8nk/IB//eEHqSP8j5C8B5Z > =rmAo > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Wed Jan 21 05:53:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 21 Jan 2009 07:53:02 -0600 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <49770DF5.6070905@ewetel.de> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> <497485CC.6060906@voiceworks.pl> <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> <87f2f3b90901191046t6b8ac1e9hfa130e48f25a772b@mail.gmail.com> <49770DF5.6070905@ewetel.de> Message-ID: <191c3a030901210553o21ec41bbq9a6d4e3bc49e8b34@mail.gmail.com> Do you have it integrated into openzap or just standalone? We can give you commit access to openzap if you want to maintain the patch in tree if you'd like. On Wed, Jan 21, 2009 at 5:58 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > just an update about my progress in this. > > Currently I have working C code which creates a pcap file containing all > needed protocoll headers in front of the Q931 dump. I use libpcap to > create the pcap file. The protocol addresses of each protocol header > structure are dynamically set by the code to reflect direction, sequence > and timeline of each Q931 packet in whireshark. Wireshark can read and > decode the current packets generated by my C code correctly. > > regards > helmut > > > > Am 19.01.2009 19:46, schrieb Michael Collins: > > Until Helmut gets his script, etc. all set the only choice you have is > > to capture the debug output on the command line. > > -MC > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkl3DfUACgkQ4tZeNddg3dz5QACePWZ6l5SEWcirI8gE4ad4N4tF > rP0AoIerlF8nk/IB//eEHqSP8j5C8B5Z > =rmAo > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/6ca18907/attachment.html From msc at freeswitch.org Wed Jan 21 05:54:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 05:54:12 -0800 Subject: [Freeswitch-users] FreeSWITCH Windows Service does not Start In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670235F118DE@mse17be1.mse17.exchange.ms> References: <150416.65876.qm@web45202.mail.sp1.yahoo.com> <118825.1733.qm@web45216.mail.sp1.yahoo.com> <6E8D2069C08AA84A83D336E996AE4C670235F118DE@mse17be1.mse17.exchange.ms> Message-ID: <87f2f3b90901210554vdb132fs18d439103ad1ab42@mail.gmail.com> On Wed, Jan 21, 2009 at 4:38 AM, Michael Giagnocavo wrote: > You're right, there should be a full installer system that'll ask what account you want to use, check permissions, etc. > -Michael Has somone opened a feature request issue on jira yet? If not, I highly recommend that you do so -MC > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron Avriel > Sent: Tuesday, January 20, 2009 11:22 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSWITCH Windows Service does not Start > > Thanks for answer. > The problem is missing permission for the network service account to FreeSWITCH directory. > Once permission is granted FreeSWITCH service starts OK. > > However, I think that unless there's a way to grant that permission inside the service > installation code ("freeswitch -install") then the service should be configured to run as local system account. > Otherwise, I think it will cause a lot of trouble to all. > > Ron > > > > > ----- Original Message ---- >> From: Michael Jerris >> To: freeswitch-users at lists.freeswitch.org >> Sent: Tuesday, January 20, 2009 6:08:25 PM >> Subject: Re: [Freeswitch-users] FreeSWITCH Windows Service does not Start >> >> One issue with the service is we have no console to dump errors too, >> it sounds like it is failing one of the startup requirements like >> config files being there. Are you able to start it in non service >> mode? If so, check permissions on the freeswitch dir that the user >> running the service has permissions to that dir. >> >> Mike >> >> On Jan 20, 2009, at 8:52 AM, Ron Avriel wrote: >> >> > Hi, >> > >> > If I try starting FreeSWITCH Windows service it immediately fails >> > with a messagebox: >> > >> > "The FreeSWITCH service on local computer started and then stopped. >> > Some services stop automatically...etc." >> > >> > I noticed that the service is installed to log on as "NT AUTHORITY >> > \NetworkService". >> > If I change this to "Local System account" then FreeSWITCH starts >> > and runs OK. >> > >> > This failure occurred on multiple Windows XP servers. >> > >> > Why does it fail and why does it not use the local system account >> > like almost all services? >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Wed Jan 21 05:56:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 21 Jan 2009 07:56:51 -0600 Subject: [Freeswitch-users] How to bridge without Answer? In-Reply-To: <21583334.post@talk.nabble.com> References: <21583334.post@talk.nabble.com> Message-ID: <191c3a030901210556n2d443179n17d8bbb9ed24b8ab@mail.gmail.com> 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 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/32390e36/attachment.html From msc at freeswitch.org Wed Jan 21 06:03:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 06:03:21 -0800 Subject: [Freeswitch-users] firing events from javascript - working example needed In-Reply-To: <11990ade0901202304k1d3b6bf6h75d4cf677eaf29ee@mail.gmail.com> References: <11990ade0901202304k1d3b6bf6h75d4cf677eaf29ee@mail.gmail.com> Message-ID: <87f2f3b90901210603i39db5167rbc255cc78880a121@mail.gmail.com> can you create a pastebin with the two scripts in question? We'll take a look and see if we can figure out what's going on. Thanks, MC On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby wrote: > 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 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From krzysiez at go2.pl Wed Jan 21 06:33:13 2009 From: krzysiez at go2.pl (Krzysztof Zimnicki) Date: Wed, 21 Jan 2009 15:33:13 +0100 Subject: [Freeswitch-users] Problem with digium te220p In-Reply-To: <87f2f3b90901210548q62c880dao12fb57f3ba1ae5f3@mail.gmail.com> References: <43dfd602.2741ece3.4976e2ec.94b1f@o2.pl> <87f2f3b90901210548q62c880dao12fb57f3ba1ae5f3@mail.gmail.com> Message-ID: <4c5d42470901210633h5f9abca0u6eb097c52c82987d@mail.gmail.com> conf/openzap.conf [span zt] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 [span zt] name => OpenZAP number => 2 trunk_type => E1 b-channel => 32-46 d-channel => 47 b-channel => 48-62 On Wed, Jan 21, 2009 at 2:48 PM, Michael Collins wrote: > can you post your openzap.conf file? > -MC > > On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki > wrote: > >>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. > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/2a159f00/attachment.html From msc at freeswitch.org Wed Jan 21 06:45:06 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 06:45:06 -0800 Subject: [Freeswitch-users] Problem with digium te220p In-Reply-To: <4c5d42470901210633h5f9abca0u6eb097c52c82987d@mail.gmail.com> References: <43dfd602.2741ece3.4976e2ec.94b1f@o2.pl> <87f2f3b90901210548q62c880dao12fb57f3ba1ae5f3@mail.gmail.com> <4c5d42470901210633h5f9abca0u6eb097c52c82987d@mail.gmail.com> Message-ID: <87f2f3b90901210645q773c8a82p4897843fb3c05699@mail.gmail.com> Okay, try the changes I note below -MC On Wed, Jan 21, 2009 at 6:33 AM, Krzysztof Zimnicki wrote: > conf/openzap.conf > > [span zt] [span zt PRI_1] > name => OpenZAP > number => 1 > trunk_type => E1 > b-channel => 1-15 > d-channel => 16 > b-channel => 17-31 > [span zt] [span zt PRI_2] > name => OpenZAP > number => 2 > > trunk_type => E1 > b-channel => 32-46 > d-channel => 47 > b-channel => 48-62 > > > On Wed, Jan 21, 2009 at 2:48 PM, Michael Collins wrote: >> >> can you post your openzap.conf file? >> -MC >> >> On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki >> wrote: >> >>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. >> > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Wed Jan 21 06:52:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 06:52:57 -0800 Subject: [Freeswitch-users] Hang up not received In-Reply-To: <4976EC8B.3080606@novatex.com.au> References: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> <191c3a030901200905r156a0b79uad5baa726eeeb7c0@mail.gmail.com> <4468a6770901201145m444a2c82gacaffd369579a47c@mail.gmail.com> <4468a6770901201343q3a51a493m2b721d4531346a77@mail.gmail.com> <46336fde0901210115o7645c261o732fe939cfc2380e@mail.gmail.com> <4976EC8B.3080606@novatex.com.au> Message-ID: <87f2f3b90901210652m14abed25l5749e9792d0501f1@mail.gmail.com> On Wed, Jan 21, 2009 at 1:36 AM, Scott Ellis wrote: > I had a similar problem, you can use > (I added an "au" > ring definition to my vars.xml file) > > To get what you want. > > I also had a problem that you get two rings, then an answer then to the > system generated ring tone, which was confusing some of our (not to bright) > callers. > > As we don't use callerID I turned that flag off in the openzap.conf.xml file > - I thought that this would do what I wanted (answer the instant the call is > detected), but the change in the config file does not make it all the way > down to the point where it takes action. At this point I hacked the code to > get what I wanted. I have to create a JIRA entry with the details yet. > > As far as I understand, this is the right place for OpenZap, as it is a > product of the FS project. At this point there is not a separate mailing list for OpenZAP stuff so here is as good a place as any to ask OZ questions. :) -MC > > Scott > > Tom?s wrote: > > Scott, I imagined that it could be an OpenZap problem, but I didn't find an > OpenZap mailing list, so I sent the email to FS list. Do you know where can > I find more information about OpenZap hardware support and developement > status (I have special interest in Loop Start)?? > > Anthony and Ognjen, I've tried tone detection and thanks to that FS is > detecting hung up, but I faced the problem that tone detector answer the > call... > > That's my dialplan: > > > > > data="sofia/internal/1003%${server-domain-name}, > sofia/internal/1004%${server-domain-name}"/> > > > > When I receive a call from PSTN, tone detection answer the call (the caller > hears only one first tone and then hears "nothing" until I pick up the call > on softphone). > > So, I think that tone detection solution does not resolve my problem... Is > there any other possibility to detect hang up without answering the call > (using Loop Start signaling) or have we to wait until OpenZap is completely > developed? > > Thanks in advance. > > On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija wrote: >> >> Ok, as discussed with Tony on IRC channel I followed his directions which >> lead to a successfull outcome (like it always does I might add :). >> >> One has to use tone_detect app in FreeSWITCH dialplan in order to check >> for busy tones coming from the PSTN side and if matched fire a hangup >> application. This is the snippet of my test dp that does the trick (from >> extension Local_extensions in default.xml): >> >> >> > data="user/${dialed_extension}@${domain_name}"/> >> This means that FS will listen to freq of 425 Hz and wait for 4 positive >> detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 >> Hz is the freq telco here uses; for other countries I suggest getting the >> ITU world tones pdf file and check there): >> >> 2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 1/4 >> 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 2/4 >> 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 3/4 >> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 4/4 >> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() >> TONE busy DETECTED >> 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup >> OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING] >> >> Regards, >> Ognjen >> >> On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija >> wrote: >>> >>> I tried similar setup with my analog card (X100P) and I'm having same >>> issue. Call is not hungup on the oz side once the caller ends. My telco >>> doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck >>> to detecting busy tone from the telco side. I'll try to modify tones.conf >>> accordingly. >>> >>> Regards, >>> Ognjen >>> (sekil) >>> On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale >>> wrote: >>>> >>>> This is a common issue with analog phones even traditional answering >>>> machines suffer from it. >>>> I'm sure you must have had an answering machine at some point that has >>>> dial tone as the message it receives. >>>> >>>> Unless FreeSWITCH has some hint that the call has hungup it will not >>>> stop trying to complete the call. >>>> >>>> If the other side is sending a busy tone to indicate hangup it's >>>> possible to use the tone_detect app to pick >>>> up on the tones and abort the call. >>>> >>>> Another thing you could do if you have unlimited inbound is explicitly >>>> answer the call in the dialplan before >>>> you call your sip phones this will give you a more profound hangup >>>> detection but it will make every call count >>>> even when nobody answers. >>>> >>>> >>>> >>>> On Tue, Jan 20, 2009 at 10:46 AM, Tom?s wrote: >>>>> >>>>> 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. >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 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 >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jaugenstine at gmail.com Wed Jan 21 06:53:41 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Wed, 21 Jan 2009 06:53:41 -0800 Subject: [Freeswitch-users] ATA-answering machine question/recommendation Message-ID: <207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45@mail.gmail.com> I have an application that requires answering machine detection. I have not been able to locate any documentation indicating that there is explicit support for answering machine detection. I have received recommendations on call flows that would include DTMF entry by the called party to detect by implication answering machines, however, I need an explicit methodology. My question is, does anyone have any experience with ATAs that might have this capability. I am interested in any solution that might even include Avaya, Cisco, or other hardware device interfaced with Freeswitch that would provide an explicit answering machine detection capability. Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/36129ec2/attachment-0001.html From msc at freeswitch.org Wed Jan 21 06:54:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 06:54:20 -0800 Subject: [Freeswitch-users] Hang up not received In-Reply-To: <4468a6770901210149l3afb036eo7e7d1151baecc37e@mail.gmail.com> References: <46336fde0901200846n5549d4eatfe9d2259f96548e1@mail.gmail.com> <191c3a030901200905r156a0b79uad5baa726eeeb7c0@mail.gmail.com> <4468a6770901201145m444a2c82gacaffd369579a47c@mail.gmail.com> <4468a6770901201343q3a51a493m2b721d4531346a77@mail.gmail.com> <46336fde0901210115o7645c261o732fe939cfc2380e@mail.gmail.com> <4976EC8B.3080606@novatex.com.au> <4468a6770901210149l3afb036eo7e7d1151baecc37e@mail.gmail.com> Message-ID: <87f2f3b90901210654j2fb7bc8dn46053f64da66674a@mail.gmail.com> On Wed, Jan 21, 2009 at 1:49 AM, Ognjen Seslija wrote: > When call comes in from Openzap, tone_detect app does pre_answer of a call > cause it's need media to start detecting tones in the first place. This > behaviour is something that I see on calls inside my telco when calling from > analogue lines. I don't think this is a big of deal because ringback > provided by FS will make caller understand that the call is still in > progress. One can make its own ringback to sound exactly the same as > telco's. > > I don't think that we'll ever make POTS behave like digital protocols do. So true! -MC > > Regards, > Ognjen > > On Wed, Jan 21, 2009 at 10:36 AM, Scott Ellis > wrote: >> >> I had a similar problem, you can use >> (I added an "au" >> ring definition to my vars.xml file) >> >> To get what you want. >> >> I also had a problem that you get two rings, then an answer then to the >> system generated ring tone, which was confusing some of our (not to bright) >> callers. >> >> As we don't use callerID I turned that flag off in the openzap.conf.xml >> file - I thought that this would do what I wanted (answer the instant the >> call is detected), but the change in the config file does not make it all >> the way down to the point where it takes action. At this point I hacked the >> code to get what I wanted. I have to create a JIRA entry with the details >> yet. >> >> As far as I understand, this is the right place for OpenZap, as it is a >> product of the FS project. >> >> Scott >> >> Tom?s wrote: >> >> Scott, I imagined that it could be an OpenZap problem, but I didn't find >> an OpenZap mailing list, so I sent the email to FS list. Do you know where >> can I find more information about OpenZap hardware support and developement >> status (I have special interest in Loop Start)?? >> >> Anthony and Ognjen, I've tried tone detection and thanks to that FS is >> detecting hung up, but I faced the problem that tone detector answer the >> call... >> >> That's my dialplan: >> >> >> >> >> > data="sofia/internal/1003%${server-domain-name}, >> sofia/internal/1004%${server-domain-name}"/> >> >> >> >> When I receive a call from PSTN, tone detection answer the call (the >> caller hears only one first tone and then hears "nothing" until I pick up >> the call on softphone). >> >> So, I think that tone detection solution does not resolve my problem... Is >> there any other possibility to detect hang up without answering the call >> (using Loop Start signaling) or have we to wait until OpenZap is completely >> developed? >> >> Thanks in advance. >> >> On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija >> wrote: >>> >>> Ok, as discussed with Tony on IRC channel I followed his directions which >>> lead to a successfull outcome (like it always does I might add :). >>> >>> One has to use tone_detect app in FreeSWITCH dialplan in order to check >>> for busy tones coming from the PSTN side and if matched fire a hangup >>> application. This is the snippet of my test dp that does the trick (from >>> extension Local_extensions in default.xml): >>> >>> >>> >> data="user/${dialed_extension}@${domain_name}"/> >>> This means that FS will listen to freq of 425 Hz and wait for 4 positive >>> detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 >>> Hz is the freq telco here uses; for other countries I suggest getting the >>> ITU world tones pdf file and check there): >>> >>> 2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 >>> tone_detect_callback() TONE busy HIT 1/4 >>> 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 >>> tone_detect_callback() TONE busy HIT 2/4 >>> 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 >>> tone_detect_callback() TONE busy HIT 3/4 >>> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 >>> tone_detect_callback() TONE busy HIT 4/4 >>> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 >>> tone_detect_callback() TONE busy DETECTED >>> 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup >>> OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING] >>> >>> Regards, >>> Ognjen >>> >>> On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija >>> wrote: >>>> >>>> I tried similar setup with my analog card (X100P) and I'm having same >>>> issue. Call is not hungup on the oz side once the caller ends. My telco >>>> doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck >>>> to detecting busy tone from the telco side. I'll try to modify tones.conf >>>> accordingly. >>>> >>>> Regards, >>>> Ognjen >>>> (sekil) >>>> On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale >>>> wrote: >>>>> >>>>> This is a common issue with analog phones even traditional answering >>>>> machines suffer from it. >>>>> I'm sure you must have had an answering machine at some point that has >>>>> dial tone as the message it receives. >>>>> >>>>> Unless FreeSWITCH has some hint that the call has hungup it will not >>>>> stop trying to complete the call. >>>>> >>>>> If the other side is sending a busy tone to indicate hangup it's >>>>> possible to use the tone_detect app to pick >>>>> up on the tones and abort the call. >>>>> >>>>> Another thing you could do if you have unlimited inbound is explicitly >>>>> answer the call in the dialplan before >>>>> you call your sip phones this will give you a more profound hangup >>>>> detection but it will make every call count >>>>> even when nobody answers. >>>>> >>>>> >>>>> >>>>> On Tue, Jan 20, 2009 at 10:46 AM, Tom?s >>>>> wrote: >>>>>> >>>>>> 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. >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 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 >>>>> iax:guest at conference.freeswitch.org/888 >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> pstn:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From helmut.kuper at ewetel.de Wed Jan 21 07:01:52 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 21 Jan 2009 16:01:52 +0100 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <191c3a030901210553o21ec41bbq9a6d4e3bc49e8b34@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> <497485CC.6060906@voiceworks.pl> <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> <87f2f3b90901191046t6b8ac1e9hfa130e48f25a772b@mail.gmail.com> <49770DF5.6070905@ewetel.de> <191c3a030901210553o21ec41bbq9a6d4e3bc49e8b34@mail.gmail.com> Message-ID: <497738E0.3050502@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, It's a standalone program to profe concept. So I think I will get it working this week as standalone. Next week I plan to start to integrate the code into openzap/ozmod_isdn.c Biggest problem currently (in my head) is the file handling of the pcap file. Currently I have only one file which is opened at startup time and closed during shutdown. In between there is no checking for a certain file size to make a file cycle decision or stopping and closing the file. I guess it is better to have a FS console command to start stop the trace on demand instead of having a config parameter which enables the tracing during FS lifetime. Further problem is that I have no way to test the code on a windows machine. So I have no idea whether it compiles on windows or not. Thanks for your support so far. regards Helmut Am 21.01.2009 14:53, schrieb Anthony Minessale: > Do you have it integrated into openzap or just standalone? > We can give you commit access to openzap if you want to maintain the patch > in tree if you'd like. > > > On Wed, Jan 21, 2009 at 5:58 AM, Helmut Kuper wrote: > > Hello, > > just an update about my progress in this. > > Currently I have working C code which creates a pcap file containing all > needed protocoll headers in front of the Q931 dump. I use libpcap to > create the pcap file. The protocol addresses of each protocol header > structure are dynamically set by the code to reflect direction, sequence > and timeline of each Q931 packet in whireshark. Wireshark can read and > decode the current packets generated by my C code correctly. > > regards > helmut > > > > Am 19.01.2009 19:46, schrieb Michael Collins: >>>> Until Helmut gets his script, etc. all set the only choice you have is >>>> to capture the debug output on the command line. >>>> -MC >> _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl3OOAACgkQ4tZeNddg3dyorQCfReKgMKGtBl+k7gIVwkoqXHRP gc0An0iY8sevWPttx3c4oEWZd9tepGxP =gN/s -----END PGP SIGNATURE----- From Laurent.Fabre at kirranet.com Wed Jan 21 08:07:35 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Wed, 21 Jan 2009 17:07:35 +0100 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed In-Reply-To: References: Message-ID: Been running all day without crashing. Will see on the long term. -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61 laurent.fabre at kirranet.com -----Message d'origine----- De?: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Ken Rice Envoy??: mercredi 21 janvier 2009 08:22 ??: freeswitch-users at lists.freeswitch.org Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed I'm not sure that patch will work... It works for me on an unrelated thing (mod_easyroute) so hopefully it will work for you here K > From: Laurent Fabre > Reply-To: > Date: Wed, 21 Jan 2009 08:27:08 +0100 > To: "freeswitch-users at lists.freeswitch.org" > > Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > Applied the patch, compiled, restarted. > > So far so good. Everything looks OK. Registrations, subscriptions, calls are > working as expected. > > I have an issue with the presence on one of the softphone (presence.wminfo?) > but it's probably unrelated and just a glitch in Bria for Outlook. > > Keep you posted. > > -- Laurent FABRE > Directeur g?n?ral > 10, rue d'Aumale > 75009 Paris > Tel: +33.(0)1.42.81.28.20 > Mob: +33.(0)6.75.75.02.96 > Fax: +33.(0)1.70.24.74.61 > laurent.fabre at kirranet.com > > > -----Message d'origine----- > De?: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Ken Rice > Envoy??: mercredi 21 janvier 2009 07:50 > ??: freeswitch-users at lists.freeswitch.org > Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > Can you try that attached patch? > > K > > >> From: Laurent Fabre >> Reply-To: >> Date: Wed, 21 Jan 2009 07:56:32 +0100 >> To: "freeswitch-users at lists.freeswitch.org" >> >> Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC >> crashed >> >> Hi Ken, >> >> Thanks for looking into it. >> >> It just crashed again on the voicemails table after running like a >> charm for 1 >> hour: http://pastebin.freeswitch.org/6832 >> >> If I can be of any assistance, please let me know. >> >> Regards, >> >> -- Laurent FABRE >> Directeur g?n?ral >> 10, rue d'Aumale >> 75009 Paris >> Tel: +33.(0)1.42.81.28.20 >> Mob: +33.(0)6.75.75.02.96 >> Fax: +33.(0)1.70.24.74.61 >> laurent.fabre at kirranet.com >> >> -----Message d'origine----- >> De?: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de >> Ken Rice Envoy??: mercredi 21 janvier 2009 06:59 ??: >> freeswitch-users at lists.freeswitch.org >> Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed >> >> I think I found the problem here... Not 100%... But I did find an >> issue in switch_odbc.c... >> >> I'll have to talk to Tony about this one... There is a condition where >> we can return an error and we shouldn't >> >> >>> From: Michael Collins >>> Reply-To: >>> Date: Tue, 20 Jan 2009 21:12:30 -0800 >>> To: >>> Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC >>> crashed >>> >>> Merci beaucoup! The devs will keep an eye on this and will hopefully >>> have a resolution soon. >>> -MC >>> >>> On Tue, Jan 20, 2009 at 8:45 PM, Laurent Fabre >>> wrote: >>>> http://jira.freeswitch.org/browse/FSCORE-276 >>>> >>>> >>>> >>>> Done. >>>> >>>> >>>> >>>> -- Laurent FABRE >>>> Directeur g?n?ral >>>> 10, rue d'Aumale >>>> 75009 Paris >>>> Tel: +33.(0)1.42.81.28.20 >>>> Mob: +33.(0)6.75.75.02.96 >>>> Fax: +33.(0)1.70.24.74.61 >>>> laurent.fabre at kirranet.com >>>> >>>> >>>> >>>> De : freeswitch-users-bounces at lists.freeswitch.org >>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de >>>> Brian West Envoy? : mercredi 21 janvier 2009 04:56 ? : >>>> freeswitch-users at lists.freeswitch.org >>>> Objet : Re: [Freeswitch-users] Freeswitch latest revision ODBC >>>> crashed >>>> >>>> >>>> >>>> Please open a jira http://jira.freeswitch.org include all the >>>> requested data in detail. >>>> >>>> >>>> >>>> /b >>>> >>>> >>>> >>>> On Jan 20, 2009, at 9:57 PM, Laurent Fabre wrote: >>>> >>>> Hi, >>>> >>>> >>>> >>>> I'm experiencing a few ? random ? segfaults ever since I enabled >>>> core ODBC support. They usually happen during startup or quickly >>>> after?or not at all as long as I do not restart freeswitch. >>>> >>>> >>>> >>>> The backend is a postgresql 7.4, every module has a distinct >>>> database (LATIN9, with OIDS). >>>> >>>> >>>> >>>> I'm having a few issues with presence & call routing which tend to >>>> break from time to time. The freeswitch box is hosted and I've got >>>> several phones with STUN enabled behind a NAT firewall (namely >>>> pfSense). I'm not sure how those things might relate but maybe it's worth >>>> mentionning. >>>> >>>> >>>> >>>> I've decided to update a few hours ago but still crashing. >>>> >>>> >>>> >>>> I was about to test mod_easyroute when I decided to run gdb and look >>>> into it : http://pastebin.freeswitch.org/6830 >>>> >>>> >>>> >>>> Could someone please help me pointing the issue? J >>>> >>>> >>>> >>>> Thanks in advance, >>>> >>>> >>>> >>>> -- Laurent FABRE >>>> Directeur g?n?ral >>>> 10, rue d'Aumale >>>> 75009 Paris >>>> Tel: +33.(0)1.42.81.28.20 >>>> Mob: +33.(0)6.75.75.02.96 >>>> Fax: +33.(0)1.70.24.74.61 >>>> laurent.fabre at kirranet.com >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>>> sers >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>>> sers >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> ers >>> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> _______________________________________________ > Freeswitch-users mailing >> list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman >> /listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >> ions/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mike at jerris.com Wed Jan 21 08:07:31 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 21 Jan 2009 11:07:31 -0500 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <497738E0.3050502@ewetel.de> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> <497485CC.6060906@voiceworks.pl> <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> <87f2f3b90901191046t6b8ac1e9hfa130e48f25a772b@mail.gmail.com> <49770DF5.6070905@ewetel.de> <191c3a030901210553o21ec41bbq9a6d4e3bc49e8b34@mail.gmail.com> <497738E0.3050502@ewetel.de> Message-ID: <7F92BE9F-1C25-4A14-A2DF-958DEDEEFC76@jerris.com> If you can get it to work in openzap, it will be easy enough for me to port when we do the windows driver integration for openzap. Mike On Jan 21, 2009, at 10:01 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > It's a standalone program to profe concept. So I think I will get it > working this week as standalone. > > Next week I plan to start to integrate the code into openzap/ > ozmod_isdn.c > > Biggest problem currently (in my head) is the file handling of the > pcap > file. Currently I have only one file which is opened at startup time > and > closed during shutdown. In between there is no checking for a certain > file size to make a file cycle decision or stopping and closing the > file. I guess it is better to have a FS console command to start stop > the trace on demand instead of having a config parameter which enables > the tracing during FS lifetime. > > Further problem is that I have no way to test the code on a windows > machine. So I have no idea whether it compiles on windows or not. > > Thanks for your support so far. > > regards > Helmut > > Am 21.01.2009 14:53, schrieb Anthony Minessale: >> Do you have it integrated into openzap or just standalone? >> We can give you commit access to openzap if you want to maintain >> the patch >> in tree if you'd like. >> >> >> On Wed, Jan 21, 2009 at 5:58 AM, Helmut Kuper >> wrote: >> >> Hello, >> >> just an update about my progress in this. >> >> Currently I have working C code which creates a pcap file >> containing all >> needed protocoll headers in front of the Q931 dump. I use libpcap to >> create the pcap file. The protocol addresses of each protocol header >> structure are dynamically set by the code to reflect direction, >> sequence >> and timeline of each Q931 packet in whireshark. Wireshark can read >> and >> decode the current packets generated by my C code correctly. >> >> regards >> helmut >> >> >> >> Am 19.01.2009 19:46, schrieb Michael Collins: >>>>> Until Helmut gets his script, etc. all set the only choice you >>>>> have is >>>>> to capture the debug output on the command line. >>>>> -MC >>> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkl3OOAACgkQ4tZeNddg3dyorQCfReKgMKGtBl+k7gIVwkoqXHRP > gc0An0iY8sevWPttx3c4oEWZd9tepGxP > =gN/s > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Kareem.Hamdy at trustvesta.com Wed Jan 21 08:14:54 2009 From: Kareem.Hamdy at trustvesta.com (Kareem Hamdy) Date: Wed, 21 Jan 2009 08:14:54 -0800 Subject: [Freeswitch-users] How to bridge without Answer? (Anthony Minessale) In-Reply-To: References: Message-ID: <1134625859513549B3B943E0133490E202681C2EB9@TDCP-EXSTORE-01.ad.trustvesta.com> 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: Wednesday, January 21, 2009 6:54 AM To: freeswitch-users at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 31, Issue 125 Send Freeswitch-users mailing list submissions to freeswitch-users at 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 at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at 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 Subject: Re: [Freeswitch-users] How to bridge without Answer? To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030901210556n2d443179n17d8bbb9ed24b8ab at 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 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/32390e36/attachment-0001.html ------------------------------ Message: 2 Date: Wed, 21 Jan 2009 06:03:21 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] firing events from javascript - working example needed To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90901210603i39db5167rbc255cc78880a121 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 can you create a pastebin with the two scripts in question? We'll take a look and see if we can figure out what's going on. Thanks, MC On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby wrote: > 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 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ------------------------------ Message: 3 Date: Wed, 21 Jan 2009 15:33:13 +0100 From: Krzysztof Zimnicki Subject: Re: [Freeswitch-users] Problem with digium te220p To: freeswitch-users at lists.freeswitch.org Message-ID: <4c5d42470901210633h5f9abca0u6eb097c52c82987d at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" conf/openzap.conf [span zt] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 [span zt] name => OpenZAP number => 2 trunk_type => E1 b-channel => 32-46 d-channel => 47 b-channel => 48-62 On Wed, Jan 21, 2009 at 2:48 PM, Michael Collins wrote: > can you post your openzap.conf file? > -MC > > On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki > wrote: > >>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. > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/2a159f00/attachment-0001.html ------------------------------ Message: 4 Date: Wed, 21 Jan 2009 06:45:06 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] Problem with digium te220p To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90901210645q773c8a82p4897843fb3c05699 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Okay, try the changes I note below -MC On Wed, Jan 21, 2009 at 6:33 AM, Krzysztof Zimnicki wrote: > conf/openzap.conf > > [span zt] [span zt PRI_1] > name => OpenZAP > number => 1 > trunk_type => E1 > b-channel => 1-15 > d-channel => 16 > b-channel => 17-31 > [span zt] [span zt PRI_2] > name => OpenZAP > number => 2 > > trunk_type => E1 > b-channel => 32-46 > d-channel => 47 > b-channel => 48-62 > > > On Wed, Jan 21, 2009 at 2:48 PM, Michael Collins wrote: >> >> can you post your openzap.conf file? >> -MC >> >> On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki >> wrote: >> >>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. >> > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ Message: 5 Date: Wed, 21 Jan 2009 06:52:57 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] Hang up not received To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90901210652m14abed25l5749e9792d0501f1 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Wed, Jan 21, 2009 at 1:36 AM, Scott Ellis wrote: > I had a similar problem, you can use > (I added an "au" > ring definition to my vars.xml file) > > To get what you want. > > I also had a problem that you get two rings, then an answer then to the > system generated ring tone, which was confusing some of our (not to bright) > callers. > > As we don't use callerID I turned that flag off in the openzap.conf.xml file > - I thought that this would do what I wanted (answer the instant the call is > detected), but the change in the config file does not make it all the way > down to the point where it takes action. At this point I hacked the code to > get what I wanted. I have to create a JIRA entry with the details yet. > > As far as I understand, this is the right place for OpenZap, as it is a > product of the FS project. At this point there is not a separate mailing list for OpenZAP stuff so here is as good a place as any to ask OZ questions. :) -MC > > Scott > > Tom?s wrote: > > Scott, I imagined that it could be an OpenZap problem, but I didn't find an > OpenZap mailing list, so I sent the email to FS list. Do you know where can > I find more information about OpenZap hardware support and developement > status (I have special interest in Loop Start)?? > > Anthony and Ognjen, I've tried tone detection and thanks to that FS is > detecting hung up, but I faced the problem that tone detector answer the > call... > > That's my dialplan: > > > > > data="sofia/internal/1003%${server-domain-name}, > sofia/internal/1004%${server-domain-name}"/> > > > > When I receive a call from PSTN, tone detection answer the call (the caller > hears only one first tone and then hears "nothing" until I pick up the call > on softphone). > > So, I think that tone detection solution does not resolve my problem... Is > there any other possibility to detect hang up without answering the call > (using Loop Start signaling) or have we to wait until OpenZap is completely > developed? > > Thanks in advance. > > On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija wrote: >> >> Ok, as discussed with Tony on IRC channel I followed his directions which >> lead to a successfull outcome (like it always does I might add :). >> >> One has to use tone_detect app in FreeSWITCH dialplan in order to check >> for busy tones coming from the PSTN side and if matched fire a hangup >> application. This is the snippet of my test dp that does the trick (from >> extension Local_extensions in default.xml): >> >> >> > data="user/${dialed_extension}@${domain_name}"/> >> This means that FS will listen to freq of 425 Hz and wait for 4 positive >> detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 >> Hz is the freq telco here uses; for other countries I suggest getting the >> ITU world tones pdf file and check there): >> >> 2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 1/4 >> 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 2/4 >> 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 3/4 >> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 4/4 >> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() >> TONE busy DETECTED >> 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup >> OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING] >> >> Regards, >> Ognjen >> >> On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija >> wrote: >>> >>> I tried similar setup with my analog card (X100P) and I'm having same >>> issue. Call is not hungup on the oz side once the caller ends. My telco >>> doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck >>> to detecting busy tone from the telco side. I'll try to modify tones.conf >>> accordingly. >>> >>> Regards, >>> Ognjen >>> (sekil) >>> On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale >>> wrote: >>>> >>>> This is a common issue with analog phones even traditional answering >>>> machines suffer from it. >>>> I'm sure you must have had an answering machine at some point that has >>>> dial tone as the message it receives. >>>> >>>> Unless FreeSWITCH has some hint that the call has hungup it will not >>>> stop trying to complete the call. >>>> >>>> If the other side is sending a busy tone to indicate hangup it's >>>> possible to use the tone_detect app to pick >>>> up on the tones and abort the call. >>>> >>>> Another thing you could do if you have unlimited inbound is explicitly >>>> answer the call in the dialplan before >>>> you call your sip phones this will give you a more profound hangup >>>> detection but it will make every call count >>>> even when nobody answers. >>>> >>>> >>>> >>>> On Tue, Jan 20, 2009 at 10:46 AM, Tom?s wrote: >>>>> >>>>> 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. >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 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 >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ Message: 6 Date: Wed, 21 Jan 2009 06:53:41 -0800 From: jonathan augenstine Subject: [Freeswitch-users] ATA-answering machine question/recommendation To: freeswitch-users at lists.freeswitch.org Message-ID: <207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" I have an application that requires answering machine detection. I have not been able to locate any documentation indicating that there is explicit support for answering machine detection. I have received recommendations on call flows that would include DTMF entry by the called party to detect by implication answering machines, however, I need an explicit methodology. My question is, does anyone have any experience with ATAs that might have this capability. I am interested in any solution that might even include Avaya, Cisco, or other hardware device interfaced with Freeswitch that would provide an explicit answering machine detection capability. Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/36129ec2/attachment.html ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 125 ************************************************* From brian at freeswitch.org Wed Jan 21 08:34:26 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 21 Jan 2009 10:34:26 -0600 Subject: [Freeswitch-users] How to bridge without Answer? (Anthony Minessale) In-Reply-To: <1134625859513549B3B943E0133490E202681C2EB9@TDCP-EXSTORE-01.ad.trustvesta.com> References: <1134625859513549B3B943E0133490E202681C2EB9@TDCP-EXSTORE-01.ad.trustvesta.com> Message-ID: You can already do this ... its how a phone call already works. CALL A -> FS -> CALL B Call A will answer when Call B is picked up passing the answer over to Call A. /b On Jan 21, 2009, at 10:14 AM, Kareem Hamdy wrote: > 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 From ajlong at worldlink.net Wed Jan 21 08:44:33 2009 From: ajlong at worldlink.net (Adam Long) Date: Wed, 21 Jan 2009 11:44:33 -0500 Subject: [Freeswitch-users] How to bridge without Answer? (Anthony Minessale) In-Reply-To: <1134625859513549B3B943E0133490E202681C2EB9@TDCP-EXSTORE-01.ad.trustvesta.com> References: <1134625859513549B3B943E0133490E202681C2EB9@TDCP-EXSTORE-01.ad.trustvesta.com> Message-ID: <021701c97be7$8a622740$9f2675c0$@net> Something like the following perhaps??? Is this possible? This would be a bridge without answer would it not? http://www.worldlink.net/sip_signals_b2bua.gif http://www.worldlink.net/sip_signals_b2bua.gif -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kareem Hamdy Sent: Wednesday, January 21, 2009 11:15 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How to bridge without Answer? (Anthony Minessale) 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: Wednesday, January 21, 2009 6:54 AM To: freeswitch-users at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 31, Issue 125 Send Freeswitch-users mailing list submissions to freeswitch-users at 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 at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at 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 Subject: Re: [Freeswitch-users] How to bridge without Answer? To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030901210556n2d443179n17d8bbb9ed24b8ab at 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 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/ 32390e36/attachment-0001.html ------------------------------ Message: 2 Date: Wed, 21 Jan 2009 06:03:21 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] firing events from javascript - working example needed To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90901210603i39db5167rbc255cc78880a121 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 can you create a pastebin with the two scripts in question? We'll take a look and see if we can figure out what's going on. Thanks, MC On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby wrote: > 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 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ------------------------------ Message: 3 Date: Wed, 21 Jan 2009 15:33:13 +0100 From: Krzysztof Zimnicki Subject: Re: [Freeswitch-users] Problem with digium te220p To: freeswitch-users at lists.freeswitch.org Message-ID: <4c5d42470901210633h5f9abca0u6eb097c52c82987d at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" conf/openzap.conf [span zt] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 [span zt] name => OpenZAP number => 2 trunk_type => E1 b-channel => 32-46 d-channel => 47 b-channel => 48-62 On Wed, Jan 21, 2009 at 2:48 PM, Michael Collins wrote: > can you post your openzap.conf file? > -MC > > On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki > wrote: > >>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. > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/ 2a159f00/attachment-0001.html ------------------------------ Message: 4 Date: Wed, 21 Jan 2009 06:45:06 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] Problem with digium te220p To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90901210645q773c8a82p4897843fb3c05699 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Okay, try the changes I note below -MC On Wed, Jan 21, 2009 at 6:33 AM, Krzysztof Zimnicki wrote: > conf/openzap.conf > > [span zt] [span zt PRI_1] > name => OpenZAP > number => 1 > trunk_type => E1 > b-channel => 1-15 > d-channel => 16 > b-channel => 17-31 > [span zt] [span zt PRI_2] > name => OpenZAP > number => 2 > > trunk_type => E1 > b-channel => 32-46 > d-channel => 47 > b-channel => 48-62 > > > On Wed, Jan 21, 2009 at 2:48 PM, Michael Collins wrote: >> >> can you post your openzap.conf file? >> -MC >> >> On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki >> wrote: >> >>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. >> > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ Message: 5 Date: Wed, 21 Jan 2009 06:52:57 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] Hang up not received To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90901210652m14abed25l5749e9792d0501f1 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Wed, Jan 21, 2009 at 1:36 AM, Scott Ellis wrote: > I had a similar problem, you can use > (I added an "au" > ring definition to my vars.xml file) > > To get what you want. > > I also had a problem that you get two rings, then an answer then to the > system generated ring tone, which was confusing some of our (not to bright) > callers. > > As we don't use callerID I turned that flag off in the openzap.conf.xml file > - I thought that this would do what I wanted (answer the instant the call is > detected), but the change in the config file does not make it all the way > down to the point where it takes action. At this point I hacked the code to > get what I wanted. I have to create a JIRA entry with the details yet. > > As far as I understand, this is the right place for OpenZap, as it is a > product of the FS project. At this point there is not a separate mailing list for OpenZAP stuff so here is as good a place as any to ask OZ questions. :) -MC > > Scott > > Tom?s wrote: > > Scott, I imagined that it could be an OpenZap problem, but I didn't find an > OpenZap mailing list, so I sent the email to FS list. Do you know where can > I find more information about OpenZap hardware support and developement > status (I have special interest in Loop Start)?? > > Anthony and Ognjen, I've tried tone detection and thanks to that FS is > detecting hung up, but I faced the problem that tone detector answer the > call... > > That's my dialplan: > > > > > data="sofia/internal/1003%${server-domain-name}, > sofia/internal/1004%${server-domain-name}"/> > > > > When I receive a call from PSTN, tone detection answer the call (the caller > hears only one first tone and then hears "nothing" until I pick up the call > on softphone). > > So, I think that tone detection solution does not resolve my problem... Is > there any other possibility to detect hang up without answering the call > (using Loop Start signaling) or have we to wait until OpenZap is completely > developed? > > Thanks in advance. > > On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija wrote: >> >> Ok, as discussed with Tony on IRC channel I followed his directions which >> lead to a successfull outcome (like it always does I might add :). >> >> One has to use tone_detect app in FreeSWITCH dialplan in order to check >> for busy tones coming from the PSTN side and if matched fire a hangup >> application. This is the snippet of my test dp that does the trick (from >> extension Local_extensions in default.xml): >> >> >> > data="user/${dialed_extension}@${domain_name}"/> >> This means that FS will listen to freq of 425 Hz and wait for 4 positive >> detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 >> Hz is the freq telco here uses; for other countries I suggest getting the >> ITU world tones pdf file and check there): >> >> 2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 1/4 >> 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 2/4 >> 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 3/4 >> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() >> TONE busy HIT 4/4 >> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() >> TONE busy DETECTED >> 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup >> OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING] >> >> Regards, >> Ognjen >> >> On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija >> wrote: >>> >>> I tried similar setup with my analog card (X100P) and I'm having same >>> issue. Call is not hungup on the oz side once the caller ends. My telco >>> doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck >>> to detecting busy tone from the telco side. I'll try to modify tones.conf >>> accordingly. >>> >>> Regards, >>> Ognjen >>> (sekil) >>> On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale >>> wrote: >>>> >>>> This is a common issue with analog phones even traditional answering >>>> machines suffer from it. >>>> I'm sure you must have had an answering machine at some point that has >>>> dial tone as the message it receives. >>>> >>>> Unless FreeSWITCH has some hint that the call has hungup it will not >>>> stop trying to complete the call. >>>> >>>> If the other side is sending a busy tone to indicate hangup it's >>>> possible to use the tone_detect app to pick >>>> up on the tones and abort the call. >>>> >>>> Another thing you could do if you have unlimited inbound is explicitly >>>> answer the call in the dialplan before >>>> you call your sip phones this will give you a more profound hangup >>>> detection but it will make every call count >>>> even when nobody answers. >>>> >>>> >>>> >>>> On Tue, Jan 20, 2009 at 10:46 AM, Tom?s wrote: >>>>> >>>>> 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. >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 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 >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ Message: 6 Date: Wed, 21 Jan 2009 06:53:41 -0800 From: jonathan augenstine Subject: [Freeswitch-users] ATA-answering machine question/recommendation To: freeswitch-users at lists.freeswitch.org Message-ID: <207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" I have an application that requires answering machine detection. I have not been able to locate any documentation indicating that there is explicit support for answering machine detection. I have received recommendations on call flows that would include DTMF entry by the called party to detect by implication answering machines, however, I need an explicit methodology. My question is, does anyone have any experience with ATAs that might have this capability. I am interested in any solution that might even include Avaya, Cisco, or other hardware device interfaced with Freeswitch that would provide an explicit answering machine detection capability. Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/ 36129ec2/attachment.html ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 125 ************************************************* _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/5ba39d2a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 24408 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/5ba39d2a/attachment-0001.gif From stevecrozz at gmail.com Wed Jan 21 08:53:14 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 21 Jan 2009 08:53:14 -0800 Subject: [Freeswitch-users] firing events from javascript - working example needed In-Reply-To: <87f2f3b90901210603i39db5167rbc255cc78880a121@mail.gmail.com> References: <11990ade0901202304k1d3b6bf6h75d4cf677eaf29ee@mail.gmail.com> <87f2f3b90901210603i39db5167rbc255cc78880a121@mail.gmail.com> Message-ID: <11990ade0901210853i44deba10oc856ab44334ac2a1@mail.gmail.com> Today I was able to see the event on the listener by subscribing to all events. But I'd like to only subscribe to a subset if possible. I thought that it would pop up when subscribing to CUSTOM events. I've put it all together very neatly here: http://pastebin.com/m6f8f7b43 --Stephen On Wed, Jan 21, 2009 at 6:03 AM, Michael Collins wrote: > can you create a pastebin with the two scripts in question? We'll take > a look and see if we can figure out what's going on. > Thanks, > MC > > On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby wrote: >> 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 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Wed Jan 21 08:57:11 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 21 Jan 2009 11:57:11 -0500 Subject: [Freeswitch-users] How to bridge without Answer? (Anthony Minessale) In-Reply-To: <021701c97be7$8a622740$9f2675c0$@net> References: <1134625859513549B3B943E0133490E202681C2EB9@TDCP-EXSTORE-01.ad.trustvesta.com> <021701c97be7$8a622740$9f2675c0$@net> Message-ID: <5D7D92B4-88B3-477E-9F7B-36D174B9DD07@jerris.com> a normal call should flow like that, with the possible exception of the ack handling, we don't wait for the a leg ack before we ack the b leg and the same for the 200ok to bye going the other way. Mike On Jan 21, 2009, at 11:44 AM, Adam Long wrote: > Something like the following perhaps??? Is this possible? > This would be a bridge without answer would it not? > http://www.worldlink.net/sip_signals_b2bua.gif > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Kareem Hamdy > Sent: Wednesday, January 21, 2009 11:15 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] How to bridge without Answer? > (Anthony Minessale) > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/11f478a3/attachment.html From freeswitch-users at digitaldan.com Wed Jan 21 08:48:23 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Wed, 21 Jan 2009 09:48:23 -0700 (MST) Subject: [Freeswitch-users] Recording ULAW files In-Reply-To: <3281576.161232556220781.JavaMail.root@zimbra> Message-ID: <10376173.181232556503428.JavaMail.root@zimbra> 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! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/17aeda6f/attachment.html From anthony.minessale at gmail.com Wed Jan 21 09:56:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 21 Jan 2009 11:56:45 -0600 Subject: [Freeswitch-users] Recording ULAW files In-Reply-To: <10376173.181232556503428.JavaMail.root@zimbra> References: <3281576.161232556220781.JavaMail.root@zimbra> <10376173.181232556503428.JavaMail.root@zimbra> Message-ID: <191c3a030901210956k14601950wa32ce1232313a941@mail.gmail.com> 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, 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/cd57ba70/attachment.html From freeswitch-users at digitaldan.com Wed Jan 21 11:05:51 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Wed, 21 Jan 2009 12:05:51 -0700 (MST) Subject: [Freeswitch-users] Recording ULAW files In-Reply-To: <15131694.251232564345254.JavaMail.root@zimbra> Message-ID: <7009019.301232564751905.JavaMail.root@zimbra> 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" To: freeswitch-users at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/39718b6a/attachment-0001.html From anthony.minessale at gmail.com Wed Jan 21 11:28:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 21 Jan 2009 13:28:19 -0600 Subject: [Freeswitch-users] Recording ULAW files In-Reply-To: <7009019.301232564751905.JavaMail.root@zimbra> References: <15131694.251232564345254.JavaMail.root@zimbra> <7009019.301232564751905.JavaMail.root@zimbra> Message-ID: <191c3a030901211128hc5b4101vd5103a1b649b0a64@mail.gmail.com> 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, 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" > To: freeswitch-users at 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, 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 at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/70e53d6e/attachment.html From lucas at johnnyvoip.com Wed Jan 21 11:32:50 2009 From: lucas at johnnyvoip.com (lucas at johnnyvoip.com) Date: Wed, 21 Jan 2009 19:32:50 +0000 Subject: [Freeswitch-users] ATA-answering machine question/recommendation In-Reply-To: <207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45@mail.gmail.com> References: <207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45@mail.gmail.com> Message-ID: <1461348542-1232566409-cardhu_decombobulator_blackberry.rim.net-569820458-@bxe161.bisx.prod.on.blackberry> Hi Jonathan, Mod_vmd (voicemail detection) should do the trick. Just search the wiki for mod_vmd, there are a number of ways of using it. Sent from my BlackBerry device on the Rogers Wireless Network -----Original Message----- From: jonathan augenstine Date: Wed, 21 Jan 2009 06:53:41 To: Subject: [Freeswitch-users] ATA-answering machine question/recommendation _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Wed Jan 21 12:00:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 12:00:55 -0800 Subject: [Freeswitch-users] firing events from javascript - working example needed In-Reply-To: <11990ade0901210853i44deba10oc856ab44334ac2a1@mail.gmail.com> References: <11990ade0901202304k1d3b6bf6h75d4cf677eaf29ee@mail.gmail.com> <87f2f3b90901210603i39db5167rbc255cc78880a121@mail.gmail.com> <11990ade0901210853i44deba10oc856ab44334ac2a1@mail.gmail.com> Message-ID: <87f2f3b90901211200v547ffecbt57edf25edbae8874@mail.gmail.com> thanks we'll have a look. Also, please use pastebin.freeswitch.org in the future because it makes it easier for us to find things. :) -MC On Wed, Jan 21, 2009 at 8:53 AM, Stephen Crosby wrote: > Today I was able to see the event on the listener by subscribing to > all events. But I'd like to only subscribe to a subset if possible. I > thought that it would pop up when subscribing to CUSTOM events. I've > put it all together very neatly here: > http://pastebin.com/m6f8f7b43 > > --Stephen > > On Wed, Jan 21, 2009 at 6:03 AM, Michael Collins wrote: >> can you create a pastebin with the two scripts in question? We'll take >> a look and see if we can figure out what's going on. >> Thanks, >> MC >> >> On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby wrote: >>> 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 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Wed Jan 21 12:10:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 21 Jan 2009 14:10:21 -0600 Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent? In-Reply-To: <496F0D38.5030904@novatex.com.au> References: <496EBFDE.2070305@drlake.com.au> <496EDCB6.4020802@novatex.com.au> <496F0D38.5030904@novatex.com.au> Message-ID: <191c3a030901211210p361c11d7q3ff45a8e73c02a95@mail.gmail.com> did you answer the call in your dialplan? do you have a full debug log of a call with that parameter enabled on the analog span in question? On Thu, Jan 15, 2009 at 4:17 AM, Scott Ellis wrote: > After poking around in the code, it looks like if I set name="enable-callerid" value="false"/> in openzap.conf.xml, it should > skip the GET_CALLERID state, and I should get the call answered straight > away. > > mod_openzap.c > > } else if (!strcasecmp(var, "enable-callerid")) { > enable_callerid = val; > > > if (zap_configure_span("analog", span, on_analog_signal, > "tonemap", tonegroup, > "digit_timeout", &to, > "max_dialstr", &max, > "hotline", hotline, > "enable_callerid", enable_callerid, > TAG_END) != ZAP_SUCCESS) { > zap_log(ZAP_LOG_ERROR, "Error starting OpenZAP span > %d\n", span_id); > continue; > } > > ozmod_analog.c > > else if (!strcasecmp(var, "enable_callerid")) { > if (!(val = va_arg(ap, char *))) { > break; > } > if (zap_true(val)) { > flags |= ZAP_ANALOG_CALLERID; > } else { > flags &= ~ZAP_ANALOG_CALLERID; > } > > and > > case ZAP_OOB_RING_START: > { > if (event->channel->type != ZAP_CHAN_TYPE_FXO) { > zap_log(ZAP_LOG_ERROR, "Cannot get a RING_START event on > a non-fxo channel, please check your config.\n"); > zap_set_state_locked(event->channel, > ZAP_CHANNEL_STATE_DOWN); > goto end; > } > if (!event->channel->ring_count && (event->channel->state == > ZAP_CHANNEL_STATE_DOWN && !zap_test_flag(event->channel, > ZAP_CHANNEL_INTHREAD))) { > if (zap_test_flag(analog_data, ZAP_ANALOG_CALLERID)) { > zap_set_state_locked(event->channel, > ZAP_CHANNEL_STATE_GET_CALLERID); > } else { > zap_set_state_locked(event->channel, > ZAP_CHANNEL_STATE_IDLE); > } > event->channel->ring_count = 1; > zap_mutex_unlock(event->channel->mutex); > locked = 0; > zap_thread_create_detached(zap_analog_channel_run, > event->channel); > } else { > event->channel->ring_count++; > } > } > break; > > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [DOWN] > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:760 process_event() Changing > state on 1:1 from DOWN to GET_CALLERID > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() > ANALOG CHANNEL thread starting. > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 1:1 for GET_CALLERID > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:45 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:48 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [RING_START][1:1] STATE [GET_CALLERID] > 2009-01-15 20:19:49 [DEBUG] ozmod_analog.c:292 zap_analog_channel_run() > Changing state on 1:1 from GET_CALLERID to IDLE > 2009-01-15 20:19:49 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 1:1 for IDLE > 2009-01-15 20:19:49 [DEBUG] mod_openzap.c:1165 on_fxo_signal() got FXO > sig 1:1 [START] > > The code all looks right, but I am not getting what I think should > happen. Anyone with any ideas? > > Scott > > Scott Ellis wrote: > > 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 > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/b88c39fa/attachment-0001.html From nyeastufo at gmail.com Wed Jan 21 12:21:10 2009 From: nyeastufo at gmail.com (Dave) Date: Wed, 21 Jan 2009 15:21:10 -0500 Subject: [Freeswitch-users] sample code of how to use the C level API instead of dialplan Message-ID: 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. From freeswitch-users at digitaldan.com Wed Jan 21 12:24:13 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Wed, 21 Jan 2009 13:24:13 -0700 (MST) Subject: [Freeswitch-users] Recording ULAW files In-Reply-To: <12215202.521232569450054.JavaMail.root@zimbra> Message-ID: <25700470.541232569453024.JavaMail.root@zimbra> Thanks, that's what I'm currently doing now. ----- Original Message ----- From: "Anthony Minessale" To: freeswitch-users at 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 at 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 at gmail.com > To: freeswitch-users at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/c73e91a3/attachment.html From msc at freeswitch.org Wed Jan 21 12:26:48 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 12:26:48 -0800 Subject: [Freeswitch-users] sample code of how to use the C level API instead of dialplan In-Reply-To: References: Message-ID: <87f2f3b90901211226r380dbdf8r2ab65f25d5482d5e@mail.gmail.com> On Wed, Jan 21, 2009 at 12:21 PM, Dave wrote: > 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. It might be helpful to know what you hope to accomplish. FreeSWITCH has a lot of hooks and C level programming may or may not be best for your particular application. Have you checked out, for example, the event socket? http://wiki.freeswitch.org/wiki/Event_Socket That might be a happy medium for you between the upper level of the dialplan and the nitty-gritty of C programming. -MC > > > > > > ___________________________________________________________ > Sent by ePrompter, the premier email notification software. > Free download at http://www.ePrompter.com. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Jan 21 12:28:04 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 21 Jan 2009 14:28:04 -0600 Subject: [Freeswitch-users] Recording ULAW files In-Reply-To: <7009019.301232564751905.JavaMail.root@zimbra> References: <7009019.301232564751905.JavaMail.root@zimbra> Message-ID: mod_sndfile already registered .ul and .al ;) /b On Jan 21, 2009, at 1:05 PM, freeswitch-users at 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/be88befa/attachment.html From msc at freeswitch.org Wed Jan 21 13:43:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 13:43:18 -0800 Subject: [Freeswitch-users] firing events from javascript - working example needed In-Reply-To: <87f2f3b90901211200v547ffecbt57edf25edbae8874@mail.gmail.com> References: <11990ade0901202304k1d3b6bf6h75d4cf677eaf29ee@mail.gmail.com> <87f2f3b90901210603i39db5167rbc255cc78880a121@mail.gmail.com> <11990ade0901210853i44deba10oc856ab44334ac2a1@mail.gmail.com> <87f2f3b90901211200v547ffecbt57edf25edbae8874@mail.gmail.com> Message-ID: <87f2f3b90901211343n24f1abd5y6947ff98062017aa@mail.gmail.com> Stephen, I've been able to duplicate this behavior on my Mac with r11333. It seems to work with Lua but not with Javascript. I am going to discuss it with the devs and possibly open a jira issue. In the meantime would you be willing to try it with Lua, even just for testing? This worked for me: -- Test sending custom events in Lua local event = freeswitch.Event("custom"); event:addHeader("Sample Custom Event", "no"); event:fire(); I saved the above as /usr/local/freeswitch/scripts/event1.lua I then opened two terminal windows, one to freeswitch CLI and the other a telnet into the event socket On the event socket I logged in and listened for custom events: telnet localhost 8021 auth ClueCon events plain custom On FS CLI I typed: lua event1.lua On the event socket I immediately see this: Sample Custom Event: no Event-Name: CUSTOM Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc FreeSWITCH-Hostname: michael-collinss-macbook-pro.local FreeSWITCH-IPv4: 192.168.1.5 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-01-21%2013%3A14%3A35 Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A14%3A35%20GMT Event-Date-Timestamp: 1232572475813346 Event-Calling-File: switch_cpp.cpp Event-Calling-Function: fire Event-Calling-Line-Number: 295 However, when I do the same kind of thing with js it doesn't work: // Sample event sent from JavaScript console_log("INFO","Starting event1.js sample event sender...\n"); var msg = "Hello, welcome to the FreeSWITCH demo application please enter some text into the chat box"; e = new Event("custom", "message"); e.addBody(msg); e.fire(); I saved the above as /usr/local/freeswitch/scripts/event1.js I run it from the FS CLI: jsrun event1.js And I see my console message pop up but I don't see anything on the event socket However, if I do this at the event socket: events plain all And then do jsrun event1.js from FS CLI then I do see my event on the event socket like this: Content-Length: 559 Content-Type: text/event-plain Event-Subclass: message Event-Name: CUSTOM Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc FreeSWITCH-Hostname: michael-collinss-macbook-pro.local FreeSWITCH-IPv4: 192.168.1.5 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-01-21%2013%3A07%3A48 Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A07%3A48%20GMT Event-Date-Timestamp: 1232572068370864 Event-Calling-File: mod_spidermonkey.c Event-Calling-Function: event_fire Event-Calling-Line-Number: 671 Content-Length: 90 Hello, welcome to the FreeSWITCH demo application please enter some text into the chat box So, there's definitely something going on, we just need to find out what for sure. I'll be in touch. -MC (mercutioviz) On Wed, Jan 21, 2009 at 12:00 PM, Michael Collins wrote: > thanks we'll have a look. Also, please use pastebin.freeswitch.org in > the future because it makes it easier for us to find things. :) > -MC > > On Wed, Jan 21, 2009 at 8:53 AM, Stephen Crosby wrote: >> Today I was able to see the event on the listener by subscribing to >> all events. But I'd like to only subscribe to a subset if possible. I >> thought that it would pop up when subscribing to CUSTOM events. I've >> put it all together very neatly here: >> http://pastebin.com/m6f8f7b43 >> >> --Stephen >> >> On Wed, Jan 21, 2009 at 6:03 AM, Michael Collins wrote: >>> can you create a pastebin with the two scripts in question? We'll take >>> a look and see if we can figure out what's going on. >>> Thanks, >>> MC >>> >>> On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby wrote: >>>> 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 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From stevecrozz at gmail.com Wed Jan 21 13:49:29 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 21 Jan 2009 13:49:29 -0800 Subject: [Freeswitch-users] firing events from javascript - working example needed In-Reply-To: <87f2f3b90901211343n24f1abd5y6947ff98062017aa@mail.gmail.com> References: <11990ade0901202304k1d3b6bf6h75d4cf677eaf29ee@mail.gmail.com> <87f2f3b90901210603i39db5167rbc255cc78880a121@mail.gmail.com> <11990ade0901210853i44deba10oc856ab44334ac2a1@mail.gmail.com> <87f2f3b90901211200v547ffecbt57edf25edbae8874@mail.gmail.com> <87f2f3b90901211343n24f1abd5y6947ff98062017aa@mail.gmail.com> Message-ID: <11990ade0901211349gf14f301jd01e8a79e043d730@mail.gmail.com> Sure, I'll give it a try when I get home. On Wed, Jan 21, 2009 at 1:43 PM, Michael Collins wrote: > Stephen, > > I've been able to duplicate this behavior on my Mac with r11333. It > seems to work with Lua but not with Javascript. I am going to discuss > it with the devs and possibly open a jira issue. In the meantime would > you be willing to try it with Lua, even just for testing? This worked > for me: > > -- Test sending custom events in Lua > local event = freeswitch.Event("custom"); > event:addHeader("Sample Custom Event", "no"); > event:fire(); > > I saved the above as /usr/local/freeswitch/scripts/event1.lua > > I then opened two terminal windows, one to freeswitch CLI and the > other a telnet into the event socket > On the event socket I logged in and listened for custom events: > telnet localhost 8021 > auth ClueCon > events plain custom > > On FS CLI I typed: > lua event1.lua > > On the event socket I immediately see this: > Sample Custom Event: no > Event-Name: CUSTOM > Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc > FreeSWITCH-Hostname: michael-collinss-macbook-pro.local > FreeSWITCH-IPv4: 192.168.1.5 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-01-21%2013%3A14%3A35 > Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A14%3A35%20GMT > Event-Date-Timestamp: 1232572475813346 > Event-Calling-File: switch_cpp.cpp > Event-Calling-Function: fire > Event-Calling-Line-Number: 295 > > However, when I do the same kind of thing with js it doesn't work: > > // Sample event sent from JavaScript > console_log("INFO","Starting event1.js sample event sender...\n"); > var msg = "Hello, welcome to the FreeSWITCH demo application > please enter some text into the chat box"; > e = new Event("custom", "message"); > e.addBody(msg); > e.fire(); > > > I saved the above as /usr/local/freeswitch/scripts/event1.js > I run it from the FS CLI: > jsrun event1.js > And I see my console message pop up but I don't see anything on the event socket > However, if I do this at the event socket: > events plain all > > And then do jsrun event1.js from FS CLI then I do see my event on the > event socket like this: > Content-Length: 559 > Content-Type: text/event-plain > > Event-Subclass: message > Event-Name: CUSTOM > Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc > FreeSWITCH-Hostname: michael-collinss-macbook-pro.local > FreeSWITCH-IPv4: 192.168.1.5 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-01-21%2013%3A07%3A48 > Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A07%3A48%20GMT > Event-Date-Timestamp: 1232572068370864 > Event-Calling-File: mod_spidermonkey.c > Event-Calling-Function: event_fire > Event-Calling-Line-Number: 671 > Content-Length: 90 > > Hello, welcome to the FreeSWITCH demo application please enter some > text into the chat box > > So, there's definitely something going on, we just need to find out > what for sure. I'll be in touch. > -MC (mercutioviz) > > On Wed, Jan 21, 2009 at 12:00 PM, Michael Collins wrote: >> thanks we'll have a look. Also, please use pastebin.freeswitch.org in >> the future because it makes it easier for us to find things. :) >> -MC >> >> On Wed, Jan 21, 2009 at 8:53 AM, Stephen Crosby wrote: >>> Today I was able to see the event on the listener by subscribing to >>> all events. But I'd like to only subscribe to a subset if possible. I >>> thought that it would pop up when subscribing to CUSTOM events. I've >>> put it all together very neatly here: >>> http://pastebin.com/m6f8f7b43 >>> >>> --Stephen >>> >>> On Wed, Jan 21, 2009 at 6:03 AM, Michael Collins wrote: >>>> can you create a pastebin with the two scripts in question? We'll take >>>> a look and see if we can figure out what's going on. >>>> Thanks, >>>> MC >>>> >>>> On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby wrote: >>>>> 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 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Wed Jan 21 13:59:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 13:59:44 -0800 Subject: [Freeswitch-users] firing events from javascript - working example needed In-Reply-To: <11990ade0901211349gf14f301jd01e8a79e043d730@mail.gmail.com> References: <11990ade0901202304k1d3b6bf6h75d4cf677eaf29ee@mail.gmail.com> <87f2f3b90901210603i39db5167rbc255cc78880a121@mail.gmail.com> <11990ade0901210853i44deba10oc856ab44334ac2a1@mail.gmail.com> <87f2f3b90901211200v547ffecbt57edf25edbae8874@mail.gmail.com> <87f2f3b90901211343n24f1abd5y6947ff98062017aa@mail.gmail.com> <11990ade0901211349gf14f301jd01e8a79e043d730@mail.gmail.com> Message-ID: <87f2f3b90901211359s4e344e2ftee747267c209c5f2@mail.gmail.com> RESOLVED! Stephen, when using event sub-classes you need to specify the subclass when subscribing to listen to the events. I added a small entry to the wiki page that I hope makes it clear: http://wiki.freeswitch.org/wiki/Javascript_Event#Subscribing_To_Custom_Events Let me know if you have any more questions. -MC P.S. - I will follow up with Lua and Perl and make sure that the wiki is clear on how to subscribe to events. it looks like the default for Lua is not to have an event subclass but the default for JavaScript does have an event subclass... :) On Wed, Jan 21, 2009 at 1:49 PM, Stephen Crosby wrote: > Sure, I'll give it a try when I get home. > > On Wed, Jan 21, 2009 at 1:43 PM, Michael Collins wrote: >> Stephen, >> >> I've been able to duplicate this behavior on my Mac with r11333. It >> seems to work with Lua but not with Javascript. I am going to discuss >> it with the devs and possibly open a jira issue. In the meantime would >> you be willing to try it with Lua, even just for testing? This worked >> for me: >> >> -- Test sending custom events in Lua >> local event = freeswitch.Event("custom"); >> event:addHeader("Sample Custom Event", "no"); >> event:fire(); >> >> I saved the above as /usr/local/freeswitch/scripts/event1.lua >> >> I then opened two terminal windows, one to freeswitch CLI and the >> other a telnet into the event socket >> On the event socket I logged in and listened for custom events: >> telnet localhost 8021 >> auth ClueCon >> events plain custom >> >> On FS CLI I typed: >> lua event1.lua >> >> On the event socket I immediately see this: >> Sample Custom Event: no >> Event-Name: CUSTOM >> Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc >> FreeSWITCH-Hostname: michael-collinss-macbook-pro.local >> FreeSWITCH-IPv4: 192.168.1.5 >> FreeSWITCH-IPv6: %3A%3A1 >> Event-Date-Local: 2009-01-21%2013%3A14%3A35 >> Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A14%3A35%20GMT >> Event-Date-Timestamp: 1232572475813346 >> Event-Calling-File: switch_cpp.cpp >> Event-Calling-Function: fire >> Event-Calling-Line-Number: 295 >> >> However, when I do the same kind of thing with js it doesn't work: >> >> // Sample event sent from JavaScript >> console_log("INFO","Starting event1.js sample event sender...\n"); >> var msg = "Hello, welcome to the FreeSWITCH demo application >> please enter some text into the chat box"; >> e = new Event("custom", "message"); >> e.addBody(msg); >> e.fire(); >> >> >> I saved the above as /usr/local/freeswitch/scripts/event1.js >> I run it from the FS CLI: >> jsrun event1.js >> And I see my console message pop up but I don't see anything on the event socket >> However, if I do this at the event socket: >> events plain all >> >> And then do jsrun event1.js from FS CLI then I do see my event on the >> event socket like this: >> Content-Length: 559 >> Content-Type: text/event-plain >> >> Event-Subclass: message >> Event-Name: CUSTOM >> Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc >> FreeSWITCH-Hostname: michael-collinss-macbook-pro.local >> FreeSWITCH-IPv4: 192.168.1.5 >> FreeSWITCH-IPv6: %3A%3A1 >> Event-Date-Local: 2009-01-21%2013%3A07%3A48 >> Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A07%3A48%20GMT >> Event-Date-Timestamp: 1232572068370864 >> Event-Calling-File: mod_spidermonkey.c >> Event-Calling-Function: event_fire >> Event-Calling-Line-Number: 671 >> Content-Length: 90 >> >> Hello, welcome to the FreeSWITCH demo application please enter some >> text into the chat box >> >> So, there's definitely something going on, we just need to find out >> what for sure. I'll be in touch. >> -MC (mercutioviz) >> >> On Wed, Jan 21, 2009 at 12:00 PM, Michael Collins wrote: >>> thanks we'll have a look. Also, please use pastebin.freeswitch.org in >>> the future because it makes it easier for us to find things. :) >>> -MC >>> >>> On Wed, Jan 21, 2009 at 8:53 AM, Stephen Crosby wrote: >>>> Today I was able to see the event on the listener by subscribing to >>>> all events. But I'd like to only subscribe to a subset if possible. I >>>> thought that it would pop up when subscribing to CUSTOM events. I've >>>> put it all together very neatly here: >>>> http://pastebin.com/m6f8f7b43 >>>> >>>> --Stephen >>>> >>>> On Wed, Jan 21, 2009 at 6:03 AM, Michael Collins wrote: >>>>> can you create a pastebin with the two scripts in question? We'll take >>>>> a look and see if we can figure out what's going on. >>>>> Thanks, >>>>> MC >>>>> >>>>> On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby wrote: >>>>>> 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 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Wed Jan 21 14:09:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 21 Jan 2009 16:09:52 -0600 Subject: [Freeswitch-users] firing events from javascript - working example needed In-Reply-To: <87f2f3b90901211343n24f1abd5y6947ff98062017aa@mail.gmail.com> References: <11990ade0901202304k1d3b6bf6h75d4cf677eaf29ee@mail.gmail.com> <87f2f3b90901210603i39db5167rbc255cc78880a121@mail.gmail.com> <11990ade0901210853i44deba10oc856ab44334ac2a1@mail.gmail.com> <87f2f3b90901211200v547ffecbt57edf25edbae8874@mail.gmail.com> <87f2f3b90901211343n24f1abd5y6947ff98062017aa@mail.gmail.com> Message-ID: <191c3a030901211409g655f6a64n51649c7580a128c1@mail.gmail.com> lua: local event = freeswitch.Event("custom"); js: e = new Event("custom", "message"); in js you specify a subclass which means you would need to subscribe to it events plain custom message really just events plain custom is not a good idea because all the real events have a subclass once the custom keyword is seen in the event command each name after that is assumed to be a custom event name. For subclass we use the naming convention mod::event_name so a better test would be: e = new Event("custom", "my_js::test"); there is no point to specify a subclass unless the event type is custom. e = new Event("message"); would be a better example because you can easily subscribe to the "message" event. On Wed, Jan 21, 2009 at 3:43 PM, Michael Collins wrote: > Stephen, > > I've been able to duplicate this behavior on my Mac with r11333. It > seems to work with Lua but not with Javascript. I am going to discuss > it with the devs and possibly open a jira issue. In the meantime would > you be willing to try it with Lua, even just for testing? This worked > for me: > > -- Test sending custom events in Lua > local event = freeswitch.Event("custom"); > event:addHeader("Sample Custom Event", "no"); > event:fire(); > > I saved the above as /usr/local/freeswitch/scripts/event1.lua > > I then opened two terminal windows, one to freeswitch CLI and the > other a telnet into the event socket > On the event socket I logged in and listened for custom events: > telnet localhost 8021 > auth ClueCon > events plain custom > > On FS CLI I typed: > lua event1.lua > > On the event socket I immediately see this: > Sample Custom Event: no > Event-Name: CUSTOM > Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc > FreeSWITCH-Hostname: michael-collinss-macbook-pro.local > FreeSWITCH-IPv4: 192.168.1.5 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-01-21%2013%3A14%3A35 > Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A14%3A35%20GMT > Event-Date-Timestamp: 1232572475813346 > Event-Calling-File: switch_cpp.cpp > Event-Calling-Function: fire > Event-Calling-Line-Number: 295 > > However, when I do the same kind of thing with js it doesn't work: > > // Sample event sent from JavaScript > console_log("INFO","Starting event1.js sample event sender...\n"); > var msg = "Hello, welcome to the FreeSWITCH demo application > please enter some text into the chat box"; > e = new Event("custom", "message"); > e.addBody(msg); > e.fire(); > > > I saved the above as /usr/local/freeswitch/scripts/event1.js > I run it from the FS CLI: > jsrun event1.js > And I see my console message pop up but I don't see anything on the event > socket > However, if I do this at the event socket: > events plain all > > And then do jsrun event1.js from FS CLI then I do see my event on the > event socket like this: > Content-Length: 559 > Content-Type: text/event-plain > > Event-Subclass: message > Event-Name: CUSTOM > Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc > FreeSWITCH-Hostname: michael-collinss-macbook-pro.local > FreeSWITCH-IPv4: 192.168.1.5 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-01-21%2013%3A07%3A48 > Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A07%3A48%20GMT > Event-Date-Timestamp: 1232572068370864 > Event-Calling-File: mod_spidermonkey.c > Event-Calling-Function: event_fire > Event-Calling-Line-Number: 671 > Content-Length: 90 > > Hello, welcome to the FreeSWITCH demo application please enter some > text into the chat box > > So, there's definitely something going on, we just need to find out > what for sure. I'll be in touch. > -MC (mercutioviz) > > On Wed, Jan 21, 2009 at 12:00 PM, Michael Collins > wrote: > > thanks we'll have a look. Also, please use pastebin.freeswitch.org in > > the future because it makes it easier for us to find things. :) > > -MC > > > > On Wed, Jan 21, 2009 at 8:53 AM, Stephen Crosby > wrote: > >> Today I was able to see the event on the listener by subscribing to > >> all events. But I'd like to only subscribe to a subset if possible. I > >> thought that it would pop up when subscribing to CUSTOM events. I've > >> put it all together very neatly here: > >> http://pastebin.com/m6f8f7b43 > >> > >> --Stephen > >> > >> On Wed, Jan 21, 2009 at 6:03 AM, Michael Collins > wrote: > >>> can you create a pastebin with the two scripts in question? We'll take > >>> a look and see if we can figure out what's going on. > >>> Thanks, > >>> MC > >>> > >>> On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby > wrote: > >>>> 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 > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at 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 at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/268354f4/attachment-0001.html From msc at freeswitch.org Wed Jan 21 14:19:52 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 21 Jan 2009 14:19:52 -0800 Subject: [Freeswitch-users] firing events from javascript - working example needed In-Reply-To: <191c3a030901211409g655f6a64n51649c7580a128c1@mail.gmail.com> References: <11990ade0901202304k1d3b6bf6h75d4cf677eaf29ee@mail.gmail.com> <87f2f3b90901210603i39db5167rbc255cc78880a121@mail.gmail.com> <11990ade0901210853i44deba10oc856ab44334ac2a1@mail.gmail.com> <87f2f3b90901211200v547ffecbt57edf25edbae8874@mail.gmail.com> <87f2f3b90901211343n24f1abd5y6947ff98062017aa@mail.gmail.com> <191c3a030901211409g655f6a64n51649c7580a128c1@mail.gmail.com> Message-ID: Anthm, I will add the substance of this to the wiki. -MC Sent from my iPhone On Jan 21, 2009, at 2:09 PM, Anthony Minessale wrote: > > lua: > > local event = freeswitch.Event("custom"); > > js: > e = new Event("custom", "message"); > > > in js you specify a subclass which means you would need to subscribe > to it > events plain custom message > > really just > events plain custom is not a good idea because all the real events > have a subclass > once the custom keyword is seen in the event command each name after > that is assumed > to be a custom event name. > > For subclass we use the naming convention mod::event_name > so a better test would be: > > e = new Event("custom", "my_js::test"); > > there is no point to specify a subclass unless the event type is > custom. > > > e = new Event("message"); > > would be a better example because you can easily subscribe to the > "message" event. > > > > > > On Wed, Jan 21, 2009 at 3:43 PM, Michael Collins > wrote: > Stephen, > > I've been able to duplicate this behavior on my Mac with r11333. It > seems to work with Lua but not with Javascript. I am going to discuss > it with the devs and possibly open a jira issue. In the meantime would > you be willing to try it with Lua, even just for testing? This worked > for me: > > -- Test sending custom events in Lua > local event = freeswitch.Event("custom"); > event:addHeader("Sample Custom Event", "no"); > event:fire(); > > I saved the above as /usr/local/freeswitch/scripts/event1.lua > > I then opened two terminal windows, one to freeswitch CLI and the > other a telnet into the event socket > On the event socket I logged in and listened for custom events: > telnet localhost 8021 > auth ClueCon > events plain custom > > On FS CLI I typed: > lua event1.lua > > On the event socket I immediately see this: > Sample Custom Event: no > Event-Name: CUSTOM > Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc > FreeSWITCH-Hostname: michael-collinss-macbook-pro.local > FreeSWITCH-IPv4: 192.168.1.5 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-01-21%2013%3A14%3A35 > Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A14%3A35%20GMT > Event-Date-Timestamp: 1232572475813346 > Event-Calling-File: switch_cpp.cpp > Event-Calling-Function: fire > Event-Calling-Line-Number: 295 > > However, when I do the same kind of thing with js it doesn't work: > > // Sample event sent from JavaScript > console_log("INFO","Starting event1.js sample event sender...\n"); > var msg = "Hello, welcome to the FreeSWITCH demo application > please enter some text into the chat box"; > e = new Event("custom", "message"); > e.addBody(msg); > e.fire(); > > > I saved the above as /usr/local/freeswitch/scripts/event1.js > I run it from the FS CLI: > jsrun event1.js > And I see my console message pop up but I don't see anything on the > event socket > However, if I do this at the event socket: > events plain all > > And then do jsrun event1.js from FS CLI then I do see my event on the > event socket like this: > Content-Length: 559 > Content-Type: text/event-plain > > Event-Subclass: message > Event-Name: CUSTOM > Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc > FreeSWITCH-Hostname: michael-collinss-macbook-pro.local > FreeSWITCH-IPv4: 192.168.1.5 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-01-21%2013%3A07%3A48 > Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A07%3A48%20GMT > Event-Date-Timestamp: 1232572068370864 > Event-Calling-File: mod_spidermonkey.c > Event-Calling-Function: event_fire > Event-Calling-Line-Number: 671 > Content-Length: 90 > > Hello, welcome to the FreeSWITCH demo application please enter some > text into the chat box > > So, there's definitely something going on, we just need to find out > what for sure. I'll be in touch. > -MC (mercutioviz) > > On Wed, Jan 21, 2009 at 12:00 PM, Michael Collins > wrote: > > thanks we'll have a look. Also, please use pastebin.freeswitch.org > in > > the future because it makes it easier for us to find things. :) > > -MC > > > > On Wed, Jan 21, 2009 at 8:53 AM, Stephen Crosby > wrote: > >> Today I was able to see the event on the listener by subscribing to > >> all events. But I'd like to only subscribe to a subset if > possible. I > >> thought that it would pop up when subscribing to CUSTOM events. > I've > >> put it all together very neatly here: > >> http://pastebin.com/m6f8f7b43 > >> > >> --Stephen > >> > >> On Wed, Jan 21, 2009 at 6:03 AM, Michael Collins > wrote: > >>> can you create a pastebin with the two scripts in question? > We'll take > >>> a look and see if we can figure out what's going on. > >>> Thanks, > >>> MC > >>> > >>> On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby > wrote: > >>>> 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 > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at 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 at 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 at 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 at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/a2b33555/attachment.html From bdeacon at highergear.com Wed Jan 21 14:40:45 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Wed, 21 Jan 2009 14:40:45 -0800 Subject: [Freeswitch-users] Error: eval_some_python() Error importing module Message-ID: <1232577645.4022.176.camel@dev03.cal.highergear.com> Greetings, Couldn't find anything on the wiki or in the mail archives. (Let me know where you think a good home for this info might be on the wiki and I'd be more than happy to write something up in there.) I'm guessing I haven't done everything necessary to enable python on my machine. I have python-2.4.3 and python-devel 2.4.3-21 installed on the FS machine. Per the instructions, I uncommented the mod_python line from modules.conf and rebuilt my freeswitch instance. Vanilla functionality is working. I set PYTHONPATH to /usr/local/freeswitch/python before restarting the mod_python-enabled freeswitch (via modules.conf.xml) # echo $PYTHONPATH /usr/local/freeswitch/python # grep mod_python \ > /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml # ls $PYTHONPATH bar.py bar.pyc # cat $PYTHONPATH/bar.py stream.write("baz") The relevant entry in conf/dialplan/default.xml: (The bridge action is only there because it seemed unhappy unless it was going to actually try to do something.) The following output shows up in fs_cli when I dial 1235: > 2009-01-21 15:20:14 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/1000 at 10.48.5.207 Execute set(foo=${python(bar)}) > 2009-01-21 15:20:14 [NOTICE] mod_python.c:107 eval_some_python() Invoking py module: bar > 2009-01-21 15:20:14 [ERR] mod_python.c:121 eval_some_python() Error importing module > 2009-01-21 15:20:14 [DEBUG] switch_core_session.c:1254 switch_core_session_execute_application() sofia/internal/1000 at 10.48.5.207 Expanded String set(foo=) > 2009-01-21 15:20:14 [DEBUG] mod_dptools.c:699 set_function() sofia/internal/1000 at 10.48.5.207 SET [foo]=[UNDEF] So I'm guessing something dumb on my part. But there are so many dumb things I'm capable of doing... :) TIA, Brian From anthony.minessale at gmail.com Wed Jan 21 14:55:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 21 Jan 2009 16:55:10 -0600 Subject: [Freeswitch-users] Error: eval_some_python() Error importing module In-Reply-To: <1232577645.4022.176.camel@dev03.cal.highergear.com> References: <1232577645.4022.176.camel@dev03.cal.highergear.com> Message-ID: <191c3a030901211455v7c482983y7cbd638474e80c9@mail.gmail.com> add def fsapi(session, stream, env, args): stream.write("baz") see: http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py could make a good addition to the wiki On Wed, Jan 21, 2009 at 4:40 PM, Brian Deacon wrote: > Greetings, > > Couldn't find anything on the wiki or in the mail archives. (Let me > know where you think a good home for this info might be on the wiki and > I'd be more than happy to write something up in there.) > > I'm guessing I haven't done everything necessary to enable python on my > machine. I have python-2.4.3 and python-devel 2.4.3-21 installed on the > FS machine. > > Per the instructions, I uncommented the mod_python line from > modules.conf and rebuilt my freeswitch instance. Vanilla functionality > is working. I set PYTHONPATH to /usr/local/freeswitch/python before > restarting the mod_python-enabled freeswitch (via modules.conf.xml) > > # echo $PYTHONPATH > /usr/local/freeswitch/python > > # grep mod_python \ > > /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml > > > # ls $PYTHONPATH > bar.py bar.pyc > > # cat $PYTHONPATH/bar.py > stream.write("baz") > > > The relevant entry in conf/dialplan/default.xml: > > > > data="{group_confirm_file=vm-hello,group_confirm_key=4,call_timeout=60} > sofia/internal/1000,sofia/internal/1002" /> > > > > (The bridge action is only there because it seemed unhappy unless it was > going to actually try to do something.) > > The following output shows up in fs_cli when I dial 1235: > > > > 2009-01-21 15:20:14 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1000 at 10.48.5.207 Execute > set(foo=${python(bar)}) > > 2009-01-21 15:20:14 [NOTICE] mod_python.c:107 eval_some_python() Invoking > py module: bar > > 2009-01-21 15:20:14 [ERR] mod_python.c:121 eval_some_python() Error > importing module > > 2009-01-21 15:20:14 [DEBUG] switch_core_session.c:1254 > switch_core_session_execute_application() sofia/internal/1000 at 10.48.5.207Expanded String set(foo=) > > 2009-01-21 15:20:14 [DEBUG] mod_dptools.c:699 set_function() > sofia/internal/1000 at 10.48.5.207 SET [foo]=[UNDEF] > > > So I'm guessing something dumb on my part. But there are so many dumb > things I'm capable of doing... :) > > TIA, > Brian > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/119e4dfd/attachment-0001.html From msc at freeswitch.org Wed Jan 21 15:33:28 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Jan 2009 15:33:28 -0800 Subject: [Freeswitch-users] Error: eval_some_python() Error importing module In-Reply-To: <1232577645.4022.176.camel@dev03.cal.highergear.com> References: <1232577645.4022.176.camel@dev03.cal.highergear.com> Message-ID: <87f2f3b90901211533g2b2d98acw6bd3a390a97fb3b9@mail.gmail.com> On Wed, Jan 21, 2009 at 2:40 PM, Brian Deacon wrote: > Greetings, > > Couldn't find anything on the wiki or in the mail archives. (Let me > know where you think a good home for this info might be on the wiki and > I'd be more than happy to write something up in there.) Thanks, you're hired! :) We definitely need Python users to step up and help with the docs. There are only a few people so far who use Python so those who do use, and want to see it flourish, need to work on the documentation. Please start here: http://wiki.freeswitch.org/wiki/Mod_python Take it from there... -MC > > I'm guessing I haven't done everything necessary to enable python on my > machine. I have python-2.4.3 and python-devel 2.4.3-21 installed on the > FS machine. > > Per the instructions, I uncommented the mod_python line from > modules.conf and rebuilt my freeswitch instance. Vanilla functionality > is working. I set PYTHONPATH to /usr/local/freeswitch/python before > restarting the mod_python-enabled freeswitch (via modules.conf.xml) > > # echo $PYTHONPATH > /usr/local/freeswitch/python > > # grep mod_python \ >> /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml > > > # ls $PYTHONPATH > bar.py bar.pyc > > # cat $PYTHONPATH/bar.py > stream.write("baz") > > > The relevant entry in conf/dialplan/default.xml: > > > > data="{group_confirm_file=vm-hello,group_confirm_key=4,call_timeout=60} > sofia/internal/1000,sofia/internal/1002" /> > > > > (The bridge action is only there because it seemed unhappy unless it was > going to actually try to do something.) > > The following output shows up in fs_cli when I dial 1235: > > >> 2009-01-21 15:20:14 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/1000 at 10.48.5.207 Execute set(foo=${python(bar)}) >> 2009-01-21 15:20:14 [NOTICE] mod_python.c:107 eval_some_python() Invoking py module: bar >> 2009-01-21 15:20:14 [ERR] mod_python.c:121 eval_some_python() Error importing module >> 2009-01-21 15:20:14 [DEBUG] switch_core_session.c:1254 switch_core_session_execute_application() sofia/internal/1000 at 10.48.5.207 Expanded String set(foo=) >> 2009-01-21 15:20:14 [DEBUG] mod_dptools.c:699 set_function() sofia/internal/1000 at 10.48.5.207 SET [foo]=[UNDEF] > > > So I'm guessing something dumb on my part. But there are so many dumb things I'm capable of doing... :) > > TIA, > Brian > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From stevecrozz at gmail.com Wed Jan 21 15:45:54 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 21 Jan 2009 15:45:54 -0800 Subject: [Freeswitch-users] Error: eval_some_python() Error importing module In-Reply-To: <87f2f3b90901211533g2b2d98acw6bd3a390a97fb3b9@mail.gmail.com> References: <1232577645.4022.176.camel@dev03.cal.highergear.com> <87f2f3b90901211533g2b2d98acw6bd3a390a97fb3b9@mail.gmail.com> Message-ID: <11990ade0901211545o6ed1b6fqc57c0003447fc9a1@mail.gmail.com> I would love to see the documentation for python flourish. I would have chosen python for my recent development if I could have quickly figured out how to use it. I chose the javascript route instead because of all the examples on the wiki. --Stephen On Wed, Jan 21, 2009 at 3:33 PM, Michael Collins wrote: > On Wed, Jan 21, 2009 at 2:40 PM, Brian Deacon wrote: >> Greetings, >> >> Couldn't find anything on the wiki or in the mail archives. (Let me >> know where you think a good home for this info might be on the wiki and >> I'd be more than happy to write something up in there.) > Thanks, you're hired! :) > We definitely need Python users to step up and help with the docs. > There are only a few people so far who use Python so those who do use, > and want to see it flourish, need to work on the documentation. > Please start here: > http://wiki.freeswitch.org/wiki/Mod_python > Take it from there... > -MC >> >> I'm guessing I haven't done everything necessary to enable python on my >> machine. I have python-2.4.3 and python-devel 2.4.3-21 installed on the >> FS machine. >> >> Per the instructions, I uncommented the mod_python line from >> modules.conf and rebuilt my freeswitch instance. Vanilla functionality >> is working. I set PYTHONPATH to /usr/local/freeswitch/python before >> restarting the mod_python-enabled freeswitch (via modules.conf.xml) >> >> # echo $PYTHONPATH >> /usr/local/freeswitch/python >> >> # grep mod_python \ >>> /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml >> >> >> # ls $PYTHONPATH >> bar.py bar.pyc >> >> # cat $PYTHONPATH/bar.py >> stream.write("baz") >> >> >> The relevant entry in conf/dialplan/default.xml: >> >> >> >> > data="{group_confirm_file=vm-hello,group_confirm_key=4,call_timeout=60} >> sofia/internal/1000,sofia/internal/1002" /> >> >> >> >> (The bridge action is only there because it seemed unhappy unless it was >> going to actually try to do something.) >> >> The following output shows up in fs_cli when I dial 1235: >> >> >>> 2009-01-21 15:20:14 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/1000 at 10.48.5.207 Execute set(foo=${python(bar)}) >>> 2009-01-21 15:20:14 [NOTICE] mod_python.c:107 eval_some_python() Invoking py module: bar >>> 2009-01-21 15:20:14 [ERR] mod_python.c:121 eval_some_python() Error importing module >>> 2009-01-21 15:20:14 [DEBUG] switch_core_session.c:1254 switch_core_session_execute_application() sofia/internal/1000 at 10.48.5.207 Expanded String set(foo=) >>> 2009-01-21 15:20:14 [DEBUG] mod_dptools.c:699 set_function() sofia/internal/1000 at 10.48.5.207 SET [foo]=[UNDEF] >> >> >> So I'm guessing something dumb on my part. But there are so many dumb things I'm capable of doing... :) >> >> TIA, >> Brian >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From scott.ellis at novatex.com.au Wed Jan 21 16:05:56 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Thu, 22 Jan 2009 11:05:56 +1100 Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent? In-Reply-To: <191c3a030901211210p361c11d7q3ff45a8e73c02a95@mail.gmail.com> References: <496EBFDE.2070305@drlake.com.au> <496EDCB6.4020802@novatex.com.au> <496F0D38.5030904@novatex.com.au> <191c3a030901211210p361c11d7q3ff45a8e73c02a95@mail.gmail.com> Message-ID: <4977B864.2070209@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/b3469704/attachment.html From bdeacon at highergear.com Wed Jan 21 16:36:42 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Wed, 21 Jan 2009 16:36:42 -0800 Subject: [Freeswitch-users] Error: eval_some_python() Error importing module In-Reply-To: <191c3a030901211455v7c482983y7cbd638474e80c9@mail.gmail.com> References: <1232577645.4022.176.camel@dev03.cal.highergear.com> <191c3a030901211455v7c482983y7cbd638474e80c9@mail.gmail.com> Message-ID: <1232584602.4022.180.camel@dev03.cal.highergear.com> See, my powers of stupidity are legendary. :) I updated the one-liner code sample on the mod_python page (near the part that points to python_example.py) to include the def fsapi line. Works like a champ now! Thanks! Brian On Wed, 2009-01-21 at 16:55 -0600, Anthony Minessale wrote: > add > > def fsapi(session, stream, env, args): > stream.write("baz") > > > see: > http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py > > could make a good addition to the wiki > > > > > On Wed, Jan 21, 2009 at 4:40 PM, Brian Deacon > wrote: > Greetings, > > Couldn't find anything on the wiki or in the mail archives. > (Let me > know where you think a good home for this info might be on the > wiki and > I'd be more than happy to write something up in there.) > > I'm guessing I haven't done everything necessary to enable > python on my > machine. I have python-2.4.3 and python-devel 2.4.3-21 > installed on the > FS machine. > > Per the instructions, I uncommented the mod_python line from > modules.conf and rebuilt my freeswitch instance. Vanilla > functionality > is working. I set PYTHONPATH to /usr/local/freeswitch/python > before > restarting the mod_python-enabled freeswitch (via > modules.conf.xml) > > # echo $PYTHONPATH > /usr/local/freeswitch/python > > # grep mod_python \ > > /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml > > > # ls $PYTHONPATH > bar.py bar.pyc > > # cat $PYTHONPATH/bar.py > stream.write("baz") > > > The relevant entry in conf/dialplan/default.xml: > > > > data="{group_confirm_file=vm-hello,group_confirm_key=4,call_timeout=60} > sofia/internal/1000,sofia/internal/1002" /> > > > > (The bridge action is only there because it seemed unhappy > unless it was > going to actually try to do something.) > > The following output shows up in fs_cli when I dial 1235: > > > > 2009-01-21 15:20:14 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() > sofia/internal/1000 at 10.48.5.207 Execute set(foo= > ${python(bar)}) > > 2009-01-21 15:20:14 [NOTICE] mod_python.c:107 > eval_some_python() Invoking py module: bar > > 2009-01-21 15:20:14 [ERR] mod_python.c:121 > eval_some_python() Error importing module > > 2009-01-21 15:20:14 [DEBUG] switch_core_session.c:1254 > switch_core_session_execute_application() > sofia/internal/1000 at 10.48.5.207 Expanded String set(foo=) > > 2009-01-21 15:20:14 [DEBUG] mod_dptools.c:699 set_function() > sofia/internal/1000 at 10.48.5.207 SET [foo]=[UNDEF] > > > So I'm guessing something dumb on my part. But there are so > many dumb things I'm capable of doing... :) > > TIA, > Brian > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdeacon at highergear.com Wed Jan 21 16:42:05 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Wed, 21 Jan 2009 16:42:05 -0800 Subject: [Freeswitch-users] Error: eval_some_python() Error importing module In-Reply-To: <11990ade0901211545o6ed1b6fqc57c0003447fc9a1@mail.gmail.com> References: <1232577645.4022.176.camel@dev03.cal.highergear.com> <87f2f3b90901211533g2b2d98acw6bd3a390a97fb3b9@mail.gmail.com> <11990ade0901211545o6ed1b6fqc57c0003447fc9a1@mail.gmail.com> Message-ID: <1232584925.4022.186.camel@dev03.cal.highergear.com> Well, rescuing someone from the evils of javascript cleans so many years off my time in purgatory that I couldn't possibly pass up the chance. :) But I'm still probably another week or two from qualifying as N00b. I imagine I could probably handle transcoding javascript examples to python if it's mostly just a syntax issue, which I imagine is most cases. I could start whacking one out if someone told me which examples to steer away from because of javascript- or python-specific issues. Brian On Wed, 2009-01-21 at 15:45 -0800, Stephen Crosby wrote: > I would love to see the documentation for python flourish. I would > have chosen python for my recent development if I could have quickly > figured out how to use it. I chose the javascript route instead > because of all the examples on the wiki. > > --Stephen > > On Wed, Jan 21, 2009 at 3:33 PM, Michael Collins wrote: > > On Wed, Jan 21, 2009 at 2:40 PM, Brian Deacon wrote: > >> Greetings, > >> > >> Couldn't find anything on the wiki or in the mail archives. (Let me > >> know where you think a good home for this info might be on the wiki and > >> I'd be more than happy to write something up in there.) > > Thanks, you're hired! :) > > We definitely need Python users to step up and help with the docs. > > There are only a few people so far who use Python so those who do use, > > and want to see it flourish, need to work on the documentation. > > Please start here: > > http://wiki.freeswitch.org/wiki/Mod_python > > Take it from there... > > -MC > >> > >> I'm guessing I haven't done everything necessary to enable python on my > >> machine. I have python-2.4.3 and python-devel 2.4.3-21 installed on the > >> FS machine. > >> > >> Per the instructions, I uncommented the mod_python line from > >> modules.conf and rebuilt my freeswitch instance. Vanilla functionality > >> is working. I set PYTHONPATH to /usr/local/freeswitch/python before > >> restarting the mod_python-enabled freeswitch (via modules.conf.xml) > >> > >> # echo $PYTHONPATH > >> /usr/local/freeswitch/python > >> > >> # grep mod_python \ > >>> /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml > >> > >> > >> # ls $PYTHONPATH > >> bar.py bar.pyc > >> > >> # cat $PYTHONPATH/bar.py > >> stream.write("baz") > >> > >> > >> The relevant entry in conf/dialplan/default.xml: > >> > >> > >> > >> >> data="{group_confirm_file=vm-hello,group_confirm_key=4,call_timeout=60} > >> sofia/internal/1000,sofia/internal/1002" /> > >> > >> > >> > >> (The bridge action is only there because it seemed unhappy unless it was > >> going to actually try to do something.) > >> > >> The following output shows up in fs_cli when I dial 1235: > >> > >> > >>> 2009-01-21 15:20:14 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/1000 at 10.48.5.207 Execute set(foo=${python(bar)}) > >>> 2009-01-21 15:20:14 [NOTICE] mod_python.c:107 eval_some_python() Invoking py module: bar > >>> 2009-01-21 15:20:14 [ERR] mod_python.c:121 eval_some_python() Error importing module > >>> 2009-01-21 15:20:14 [DEBUG] switch_core_session.c:1254 switch_core_session_execute_application() sofia/internal/1000 at 10.48.5.207 Expanded String set(foo=) > >>> 2009-01-21 15:20:14 [DEBUG] mod_dptools.c:699 set_function() sofia/internal/1000 at 10.48.5.207 SET [foo]=[UNDEF] > >> > >> > >> So I'm guessing something dumb on my part. But there are so many dumb things I'm capable of doing... :) > >> > >> TIA, > >> Brian > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From stevecrozz at gmail.com Wed Jan 21 17:00:59 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 21 Jan 2009 17:00:59 -0800 Subject: [Freeswitch-users] Error: eval_some_python() Error importing module In-Reply-To: <1232584925.4022.186.camel@dev03.cal.highergear.com> References: <1232577645.4022.176.camel@dev03.cal.highergear.com> <87f2f3b90901211533g2b2d98acw6bd3a390a97fb3b9@mail.gmail.com> <11990ade0901211545o6ed1b6fqc57c0003447fc9a1@mail.gmail.com> <1232584925.4022.186.camel@dev03.cal.highergear.com> Message-ID: <11990ade0901211700j33a63f07r93abc7019e13d05@mail.gmail.com> I'd like to see an example of recording voicemail and a good conference application, tone generation, just a good IVR example would be nice too. --Stephen On Wed, Jan 21, 2009 at 4:42 PM, Brian Deacon wrote: > Well, rescuing someone from the evils of javascript cleans so many years > off my time in purgatory that I couldn't possibly pass up the chance. :) > > But I'm still probably another week or two from qualifying as N00b. I > imagine I could probably handle transcoding javascript examples to > python if it's mostly just a syntax issue, which I imagine is most > cases. > > I could start whacking one out if someone told me which examples to > steer away from because of javascript- or python-specific issues. > > Brian > > On Wed, 2009-01-21 at 15:45 -0800, Stephen Crosby wrote: >> I would love to see the documentation for python flourish. I would >> have chosen python for my recent development if I could have quickly >> figured out how to use it. I chose the javascript route instead >> because of all the examples on the wiki. >> >> --Stephen >> >> On Wed, Jan 21, 2009 at 3:33 PM, Michael Collins wrote: >> > On Wed, Jan 21, 2009 at 2:40 PM, Brian Deacon wrote: >> >> Greetings, >> >> >> >> Couldn't find anything on the wiki or in the mail archives. (Let me >> >> know where you think a good home for this info might be on the wiki and >> >> I'd be more than happy to write something up in there.) >> > Thanks, you're hired! :) >> > We definitely need Python users to step up and help with the docs. >> > There are only a few people so far who use Python so those who do use, >> > and want to see it flourish, need to work on the documentation. >> > Please start here: >> > http://wiki.freeswitch.org/wiki/Mod_python >> > Take it from there... >> > -MC >> >> >> >> I'm guessing I haven't done everything necessary to enable python on my >> >> machine. I have python-2.4.3 and python-devel 2.4.3-21 installed on the >> >> FS machine. >> >> >> >> Per the instructions, I uncommented the mod_python line from >> >> modules.conf and rebuilt my freeswitch instance. Vanilla functionality >> >> is working. I set PYTHONPATH to /usr/local/freeswitch/python before >> >> restarting the mod_python-enabled freeswitch (via modules.conf.xml) >> >> >> >> # echo $PYTHONPATH >> >> /usr/local/freeswitch/python >> >> >> >> # grep mod_python \ >> >>> /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml >> >> >> >> >> >> # ls $PYTHONPATH >> >> bar.py bar.pyc >> >> >> >> # cat $PYTHONPATH/bar.py >> >> stream.write("baz") >> >> >> >> >> >> The relevant entry in conf/dialplan/default.xml: >> >> >> >> >> >> >> >> > >> data="{group_confirm_file=vm-hello,group_confirm_key=4,call_timeout=60} >> >> sofia/internal/1000,sofia/internal/1002" /> >> >> >> >> >> >> >> >> (The bridge action is only there because it seemed unhappy unless it was >> >> going to actually try to do something.) >> >> >> >> The following output shows up in fs_cli when I dial 1235: >> >> >> >> >> >>> 2009-01-21 15:20:14 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/1000 at 10.48.5.207 Execute set(foo=${python(bar)}) >> >>> 2009-01-21 15:20:14 [NOTICE] mod_python.c:107 eval_some_python() Invoking py module: bar >> >>> 2009-01-21 15:20:14 [ERR] mod_python.c:121 eval_some_python() Error importing module >> >>> 2009-01-21 15:20:14 [DEBUG] switch_core_session.c:1254 switch_core_session_execute_application() sofia/internal/1000 at 10.48.5.207 Expanded String set(foo=) >> >>> 2009-01-21 15:20:14 [DEBUG] mod_dptools.c:699 set_function() sofia/internal/1000 at 10.48.5.207 SET [foo]=[UNDEF] >> >> >> >> >> >> So I'm guessing something dumb on my part. But there are so many dumb things I'm capable of doing... :) >> >> >> >> TIA, >> >> Brian >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Jan 21 17:14:58 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 21 Jan 2009 19:14:58 -0600 Subject: [Freeswitch-users] Error: eval_some_python() Error importing module In-Reply-To: <11990ade0901211700j33a63f07r93abc7019e13d05@mail.gmail.com> References: <1232577645.4022.176.camel@dev03.cal.highergear.com> <87f2f3b90901211533g2b2d98acw6bd3a390a97fb3b9@mail.gmail.com> <11990ade0901211545o6ed1b6fqc57c0003447fc9a1@mail.gmail.com> <1232584925.4022.186.camel@dev03.cal.highergear.com> <11990ade0901211700j33a63f07r93abc7019e13d05@mail.gmail.com> Message-ID: Most if not all of this functionality is done with FreeSWITCH without any need for python in the first place. For example you don't use python for voicemail, conferences, tone generation but maybe IVR... which is what any of the languages are for. /b On Jan 21, 2009, at 7:00 PM, Stephen Crosby wrote: > I'd like to see an example of recording voicemail and a good > conference application, tone generation, just a good IVR example would > be nice too. > > --Stephen From ajlong at worldlink.net Wed Jan 21 18:07:24 2009 From: ajlong at worldlink.net (Adam Long) Date: Wed, 21 Jan 2009 21:07:24 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670235F116EE@mse17be1.mse17.exchange.ms> References: <021001c97768$8c806a60$a5813f20$@net> <6E8D2069C08AA84A83D336E996AE4C670235F10C2A@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C670235F116EE@mse17be1.mse17.exchange.ms> Message-ID: <032a01c97c36$2b3fced0$81bf6c70$@net> Thanks Michael! FreeSWITCH.Managed.dll loads successfully without tinkering. Tks for the patch! Of course after adding the LD lib path. Thank you very much! Also saw you updated the BUG on Jira.. Thanks again great work! Only other thing to note is "languages/mod_managed" still needs to be manually added to modules.conf at this point. Perhaps a patch with this commented out by default would be a good idea for the future. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Giagnocavo Sent: Tuesday, January 20, 2009 5:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 OK, I just checked in a fix that should make it load, or at least print a helpful exception. However, you will need the freeswitch/mod folder in the LD path, so: export LD_LIBRARY_PATH=/usr/local/freeswitch/mod or edit the ld configs. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Giagnocavo Sent: Thursday, January 15, 2009 4:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 Last time I tried it I actually built from a snapshot, which should be less stable. It was on CentOS 5.2, however. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long Sent: Thursday, January 15, 2009 4:25 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 OK yah i'm experiencing exact same problem here CentOS 5.2 2.6.18-92.el5 #1 SMP Tue Jun 10 18:49:47 EDT 2008 i686 athlon i386 GNU/Linux I too have no problems at all on Windows. I'm going to try a Suse or Ubuntu prebuilt/packaged with Mono 2 I suspect it may be kernel/mono incompatibility. Did you compile mono from tarbal? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim B Sent: Thursday, January 15, 2009 6:18 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 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 at lists.freeswitch.org > Subject: Freeswitch-users Digest, Vol 31, Issue 77 > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 15 Jan 2009 00:20:53 -0800 > > Send Freeswitch-users mailing list submissions to > freeswitch-users at 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 at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at 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 > Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 > To: > Message-ID: > 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 at phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig > [root at phone2 mod_managed]# make > [root at phone2 mod_managed]# make install > > 3) Added to modules.conf.xml : > > > 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_0120 09 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090114/ 6a5facdc/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Thu, 15 Jan 2009 00:34:20 -0500 > From: Michael Giagnocavo > Subject: Re: [Freeswitch-users] Using mod_managed Linux/Mono 2.02 > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: > <6E8D2069C08AA84A83D336E996AE4C670235BBB97F at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long > Sent: Wednesday, January 14, 2009 3:45 PM > To: freeswitch-users at 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 at 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 > 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/ 73ac27e4/attachment-0001.html > > ------------------------------ > > Message: 3 > Date: Thu, 15 Jan 2009 17:50:30 +1100 > From: Scott Ellis > Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - > Freeswitch equivalent? > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: <496EDCB6.4020802 at 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 > Subject: [Freeswitch-users] Country specific tones - how to > contribute? > To: "freeswitch-users at lists.freeswitch.org" > > Message-ID: <496EE2BD.2050102 at 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 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 > Subject: Re: [Freeswitch-users] Country specific tones - how to > contribute? > To: freeswitch-users at lists.freeswitch.org > Message-ID: <20090115072405.GA15789 at jdc.jasonjgw.net> > Content-Type: text/plain; charset=us-ascii > > Scott Ellis 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 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 $${variable-name} syntax. > > There's an ITU document referred to on the wiki with the official definitions > of ringback and other tones for various countries. > > > > > ------------------------------ > > Message: 6 > Date: Thu, 15 Jan 2009 15:43:20 +0800 > From: "Juan Backson" > Subject: [Freeswitch-users] Changes in PlayAndGetDigits > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <27c25bc40901142343l34a3e99ftecf0df971e8e32f6 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi, > > Is there a change in the playAndGetDigits api? In the old release, > 11102, my lua script is working but is not working in the latest > release. > The error I am getting is " Error in playAndGetDigits expected 10..10 > args, got 9 ". > > Thanks, > JB > > > > ------------------------------ > > Message: 7 > Date: Thu, 15 Jan 2009 09:20:18 +0100 > From: Peter P GMX > Subject: Re: [Freeswitch-users] OpenZAP parse error [-3012] > [Q931E_INVALID_CRV] > To: freeswitch-users at lists.freeswitch.org > Message-ID: <496EF1C2.8020607 at gmx.net> > Content-Type: text/plain; charset=ISO-8859-1 > > Hello Michael, > > how much $$ 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 at lists.freeswitch.org/msg05792.h tml > 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 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 at 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 at 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 at 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 LiveT: Keep your life in sync. Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090121/0edc8ceb/attachment-0001.html From brian at freeswitch.org Wed Jan 21 18:16:18 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 21 Jan 2009 20:16:18 -0600 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: <032a01c97c36$2b3fced0$81bf6c70$@net> References: <021001c97768$8c806a60$a5813f20$@net> <6E8D2069C08AA84A83D336E996AE4C670235F10C2A@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C670235F116EE@mse17be1.mse17.exchange.ms> <032a01c97c36$2b3fced0$81bf6c70$@net> Message-ID: <59D4BA8D-9FCE-481C-A2B9-5D9C661395D3@freeswitch.org> Bet you could add /usr/local/freeswitch/mod to /etc/ld.so.conf and run ldconfig and accomplish the same thing ;) /b On Jan 21, 2009, at 8:07 PM, Adam Long wrote: > Thanks Michael! > > FreeSWITCH.Managed.dll loads successfully without tinkering. Tks > for the patch! > Of course after adding the LD lib path. > > Thank you very much! > Also saw you updated the BUG on Jira.. Thanks again great work! > > Only other thing to note is ?languages/mod_managed? still needs to > be manually added to modules.conf at this point. > Perhaps a patch with this commented out by default would be a good > idea for the future. > From ajlong at worldlink.net Wed Jan 21 18:56:40 2009 From: ajlong at worldlink.net (Adam Long) Date: Wed, 21 Jan 2009 21:56:40 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: <59D4BA8D-9FCE-481C-A2B9-5D9C661395D3@freeswitch.org> References: <021001c97768$8c806a60$a5813f20$@net> <6E8D2069C08AA84A83D336E996AE4C670235F10C2A@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C670235F116EE@mse17be1.mse17.exchange.ms> <032a01c97c36$2b3fced0$81bf6c70$@net> <59D4BA8D-9FCE-481C-A2B9-5D9C661395D3@freeswitch.org> Message-ID: <033e01c97c3d$0d146720$273d3560$@net> Yah I did try that ... doesn't seem to work. Which seems odd I had thought the same thing. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, January 21, 2009 9:16 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 Bet you could add /usr/local/freeswitch/mod to /etc/ld.so.conf and run ldconfig and accomplish the same thing ;) /b On Jan 21, 2009, at 8:07 PM, Adam Long wrote: > Thanks Michael! > > FreeSWITCH.Managed.dll loads successfully without tinkering. Tks > for the patch! > Of course after adding the LD lib path. > > Thank you very much! > Also saw you updated the BUG on Jira.. Thanks again great work! > > Only other thing to note is "languages/mod_managed" still needs to > be manually added to modules.conf at this point. > Perhaps a patch with this commented out by default would be a good > idea for the future. > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Wed Jan 21 18:58:30 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 21 Jan 2009 20:58:30 -0600 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: <033e01c97c3d$0d146720$273d3560$@net> References: <021001c97768$8c806a60$a5813f20$@net> <6E8D2069C08AA84A83D336E996AE4C670235F10C2A@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C670235F116EE@mse17be1.mse17.exchange.ms> <032a01c97c36$2b3fced0$81bf6c70$@net> <59D4BA8D-9FCE-481C-A2B9-5D9C661395D3@freeswitch.org> <033e01c97c3d$0d146720$273d3560$@net> Message-ID: <9FCA35AB-D024-43FB-BE02-AB4D2D7CD100@freeswitch.org> You have to make sure you do it in the same shell before you start FreeSWITCH or it won't work. How did you install mono on 5.2 I'll try it out also. /b On Jan 21, 2009, at 8:56 PM, Adam Long wrote: > Yah I did try that ... doesn't seem to work. > Which seems odd I had thought the same thing. From ajlong at worldlink.net Wed Jan 21 19:19:39 2009 From: ajlong at worldlink.net (Adam Long) Date: Wed, 21 Jan 2009 22:19:39 -0500 Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 In-Reply-To: <9FCA35AB-D024-43FB-BE02-AB4D2D7CD100@freeswitch.org> References: <021001c97768$8c806a60$a5813f20$@net> <6E8D2069C08AA84A83D336E996AE4C670235F10C2A@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C670235F116EE@mse17be1.mse17.exchange.ms> <032a01c97c36$2b3fced0$81bf6c70$@net> <59D4BA8D-9FCE-481C-A2B9-5D9C661395D3@freeswitch.org> <033e01c97c3d$0d146720$273d3560$@net> <9FCA35AB-D024-43FB-BE02-AB4D2D7CD100@freeswitch.org> Message-ID: <034101c97c40$43206f00$c9614d00$@net> I just used mono 2.2 src tarbal .. configured with --prefix=/usr --sysconfdir=/etc --localstatedir=/var And Freeswitch just configured as --prefix=/usr/local/freeswitch Nothing special. I did run ldconfig after editing files on the same shell as I used to start freeswitch I am manually loading mod_managed from the console after starting I don't auto load it. (just for now so I can see whats going on) -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, January 21, 2009 9:59 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 You have to make sure you do it in the same shell before you start FreeSWITCH or it won't work. How did you install mono on 5.2 I'll try it out also. /b On Jan 21, 2009, at 8:56 PM, Adam Long wrote: > Yah I did try that ... doesn't seem to work. > Which seems odd I had thought the same thing. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From scott.ellis at novatex.com.au Wed Jan 21 22:14:15 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Thu, 22 Jan 2009 17:14:15 +1100 Subject: [Freeswitch-users] Can I restrict a gateway to 1 call at a time? Message-ID: <49780EB7.5070102@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/f6691ee0/attachment.html From krice at suspicious.org Wed Jan 21 22:34:09 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 22 Jan 2009 00:34:09 -0600 Subject: [Freeswitch-users] Can I restrict a gateway to 1 call at a time? In-Reply-To: <49780EB7.5070102@novatex.com.au> Message-ID: You could use mod_limit for this... From: Scott Ellis Reply-To: Date: Thu, 22 Jan 2009 17:14:15 +1100 To: Subject: [Freeswitch-users] Can I restrict a gateway to 1 call at a time? I need to restrict the outbound calls to an SPA3000. Is there a way of marking a gateway to only allow one call at a time? Or, as the SPA3000 registers with FS, allowing the extension to only allow one outbound call. (Problem here is I have not worked out how to send a call to the FXS using the extension information, I have just been sending it as sofia/internal/$1 at 10.0.0.17:5061 ) If I do this I get a dial tone, but I can't figure how to send the number in this case. Obviously my understanding of dial plans is less than what it could be! This is all necessary, as if you send a second outbound call to the SPA3000, it will then think that there is an inbound call 1min 30 seconds later. Scott _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/34f6ff6f/attachment.html From helmut.kuper at ewetel.de Thu Jan 22 02:05:15 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 22 Jan 2009 11:05:15 +0100 Subject: [Freeswitch-users] Dialplan Question: Is extension local? Message-ID: <497844DB.3090702@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, is there a dialan or dptool function, which allows to check whether a user/extension is local or not ? What I try to do is this: Local extension? -> bridge -> bridge fails (on_busy, no answere, timeout, offline)->voicemail Every other call should go to PSTN. But unfortunately calls which should go to PSTN goes to voicemail ... I use default dialplan for local extensions which ships with FS regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl4RNsACgkQ4tZeNddg3dz26gCdHbAbrp0Z+YzIuVTewo+5tE7K GJcAniMGvoIE6yLCJJ3auVSJtKW/fma4 =5eQI -----END PGP SIGNATURE----- From krice at suspicious.org Thu Jan 22 02:19:05 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 22 Jan 2009 04:19:05 -0600 Subject: [Freeswitch-users] Dialplan Question: Is extension local? In-Reply-To: <497844DB.3090702@ewetel.de> Message-ID: You could write something in JS or lua that would check local numbers against a DB then forward them out if not (transfer) or send them to the local extension processor > From: Helmut Kuper > Reply-To: > Date: Thu, 22 Jan 2009 11:05:15 +0100 > To: > Subject: [Freeswitch-users] Dialplan Question: Is extension local? > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > is there a dialan or dptool function, which allows to check whether a > user/extension is local or not ? > > What I try to do is this: > > Local extension? -> bridge -> bridge fails (on_busy, no answere, > timeout, offline)->voicemail > > Every other call should go to PSTN. > > But unfortunately calls which should go to PSTN goes to voicemail ... > > I use default dialplan for local extensions which ships with FS > > regards > helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkl4RNsACgkQ4tZeNddg3dz26gCdHbAbrp0Z+YzIuVTewo+5tE7K > GJcAniMGvoIE6yLCJJ3auVSJtKW/fma4 > =5eQI > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pmhshz at gmail.com Thu Jan 22 03:59:39 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 22 Jan 2009 03:59:39 -0800 (PST) Subject: [Freeswitch-users] How to bridge without Answer? In-Reply-To: <191c3a030901210556n2d443179n17d8bbb9ed24b8ab@mail.gmail.com> References: <21583334.post@talk.nabble.com> <191c3a030901210556n2d443179n17d8bbb9ed24b8ab@mail.gmail.com> Message-ID: <21602715.post@talk.nabble.com> Thanks Anthony, There are some toll-free numbers I need to configure such that, originator does not need to charge to its users, even though they are answered on terminator side. Anthony Minessale-2 wrote: > > 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 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 at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-to-bridge-without-Answer--tp21583334p21602715.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From steveu at coppice.org Thu Jan 22 04:00:29 2009 From: steveu at coppice.org (Steve Underwood) Date: Thu, 22 Jan 2009 20:00:29 +0800 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <49770DF5.6070905@ewetel.de> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> <497485CC.6060906@voiceworks.pl> <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> <87f2f3b90901191046t6b8ac1e9hfa130e48f25a772b@mail.gmail.com> <49770DF5.6070905@ewetel.de> Message-ID: <49785FDD.3010302@coppice.org> Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > just an update about my progress in this. > > Currently I have working C code which creates a pcap file containing all > needed protocoll headers in front of the Q931 dump. I use libpcap to > create the pcap file. The protocol addresses of each protocol header > structure are dynamically set by the code to reflect direction, sequence > and timeline of each Q931 packet in whireshark. Wireshark can read and > decode the current packets generated by my C code correctly. > Have you figured out how well the Q.931 decoder in wireshark works? The decoders I have used, like T.38, are extremely buggy. I wonder if wireshark offers a shortcut that you have missed, for getting the data in. If you look through the supported protocols, there are some which never touch an IP network. For those to be useful, people must have a easy way to tie their data sources into wireshark. I've never looked into what they do, though. Steve From dave at 3c.co.uk Thu Jan 22 04:44:20 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 22 Jan 2009 12:44:20 +0000 Subject: [Freeswitch-users] How to bridge without Answer? In-Reply-To: <21602715.post@talk.nabble.com> References: <21583334.post@talk.nabble.com> <191c3a030901210556n2d443179n17d8bbb9ed24b8ab@mail.gmail.com> <21602715.post@talk.nabble.com> Message-ID: <49786A24.8010300@3c.co.uk> There's a whole bunch of reasons why you might not want to answer an inbound call: - intercept messages (e.g. "the cellphone you've called is switched off") - cost reduction on 1-800 calls, although you won't get a forward audio path from the caller until you do answer it - in one case, a company for whom I'd provided some IVR (back in the 1990s) had someone mail out some tens of thousands of cards with "You owe us X - you must call this (900) number now to avoid court proceedings" on; we were able to not answer the inbound leg of the call, but still play a recorded message to the caller informing them that they could just ignore it. Had we had to answer the inbound leg, they'd have been charged. --Dave > Thanks Anthony, > > There are some toll-free numbers I need to configure such that, originator > does not need to charge to its users, even though they are answered on > terminator side. > > > > > Anthony Minessale-2 wrote: > >> 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 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 at 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 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 >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > -- David Knell, Director, 3C Limited T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623 http://www.3c.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/f2628bd9/attachment.html From anthony.minessale at gmail.com Thu Jan 22 06:30:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 22 Jan 2009 08:30:43 -0600 Subject: [Freeswitch-users] How to bridge without Answer? In-Reply-To: <49786A24.8010300@3c.co.uk> References: <21583334.post@talk.nabble.com> <191c3a030901210556n2d443179n17d8bbb9ed24b8ab@mail.gmail.com> <21602715.post@talk.nabble.com> <49786A24.8010300@3c.co.uk> Message-ID: <191c3a030901220630y6401d13ene8302c6cf840d61a@mail.gmail.com> Daivd, I think you missed part of his question. You can easily choose not to answer an inbound call in FS by never explicitly answering it. you can call pre_answer instead or if you send the call to an app that requires media it's pre_answered automatically. pre_answer in FS terms is early media. You can run an ivr for instance completely in early_media (assuming the telco allows dtmf during early media) He asked if we can *bridge* the call outward to another endpoint and not pass the answer across when the far end answers. The bridge application is designed to bridge calls and the standard behavior when forwarding a call would be once the far end answers, pass the answer indication down the line. We do not currently have provision for supressing the answer as I stated. It would require a patch. On Thu, Jan 22, 2009 at 6:44 AM, David Knell wrote: > There's a whole bunch of reasons why you might not want to answer an > inbound call: > - intercept messages (e.g. "the cellphone you've called is switched off") > - cost reduction on 1-800 calls, although you won't get a forward audio > path from the > caller until you do answer it > - in one case, a company for whom I'd provided some IVR (back in the 1990s) > had > someone mail out some tens of thousands of cards with "You owe us X - you > must call > this (900) number now to avoid court proceedings" on; we were able to not > answer the > inbound leg of the call, but still play a recorded message to the caller > informing them > that they could just ignore it. Had we had to answer the inbound leg, > they'd have been > charged. > > --Dave > > Thanks Anthony, > > There are some toll-free numbers I need to configure such that, originator > does not need to charge to its users, even though they are answered on > terminator side. > > > > > Anthony Minessale-2 wrote: > > > 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 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 listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthmMSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conferencesip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > -- > David Knell, Director, 3C Limited > T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/acfe5fe3/attachment.html From krivushinme at rn-inform.tomsk.ru Thu Jan 22 01:37:57 2009 From: krivushinme at rn-inform.tomsk.ru (=?utf-8?b?0JrRgNC40LLRg9GI0LjQvSDQnNC40YXQsNC40Ls=?=) Date: Thu, 22 Jan 2009 15:37:57 +0600 Subject: [Freeswitch-users] mod_g729 Message-ID: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> If anyone want to play with it - http://freehg.org/u/deepwalker/fs_g729/ You use it on your own risk and compile it yourself. If you want to develop it - write me. I wrote it for russian community. -- ? ?????????, ???????? ?????? ??????? ?????????? ?????? ????????????????, ??? "??-??????" ?????? ? ?.??????, ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru mail: KrivushinME at rn-inform.tomsk.ru From helmut.kuper at ewetel.de Thu Jan 22 08:04:42 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 22 Jan 2009 17:04:42 +0100 Subject: [Freeswitch-users] Q931 decoding In-Reply-To: <49785FDD.3010302@coppice.org> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <49743CF8.6050804@ewetel.de> <497485CC.6060906@voiceworks.pl> <2ea4d47e0901191042r786df6eeo6da94643d98db221@mail.gmail.com> <87f2f3b90901191046t6b8ac1e9hfa130e48f25a772b@mail.gmail.com> <49770DF5.6070905@ewetel.de> <49785FDD.3010302@coppice.org> Message-ID: <4978991A.1010209@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Steve, wireshark decodes all things I needed to debug q931 and by now I haven't found a bug compared to my commercial "aurora duet" q931 monitor device. Well I spent some time to find a way in wireshark to get raw and pure q931 data in, but I didn't found a way except creating a protocoll stack as wireshark expect it. Nevertheless wireshark allways decodes from layer 1 successive to upper layers. This tells me that I definately have to encapsulate q931 somehow in a layer 1 link known by wireshark. Sa far as I know q931 is a layer 3 protocol in ISDN world, thus it can not be read directly by wireshark. whireshark knows a lot of layer 1 protocols - much more than me. And so I decide to use the protocols I know and wireshark knows. But if you find a direct way, please let me know, so I can speed up the logging of q931 packets into pcap file. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl4mRoACgkQ4tZeNddg3dzCNACeOzxwrY0VjvRBTkAlN+Pl3Qle J8QAoIme9CocEo2/RNVjs0T6psCRwMY/ =pbet -----END PGP SIGNATURE----- From helmut.kuper at ewetel.de Thu Jan 22 08:14:32 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 22 Jan 2009 17:14:32 +0100 Subject: [Freeswitch-users] voicemail web interface Message-ID: <49789B68.7060802@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried to get voicemail web interface up and running. I found that we need xml_rpc to open the 8080 port of FS. Then USer-profile needs "http-allowed-api=voicemail" parameter. After that I left a message for my account and tried to access my voicemails via web (I replaced fs.ip with my IP): http://fs.ip:8080/api/voicemail/web This led to a web page generating out of web-vm.tpl with 0 (zero) voicemails. If I call my voicebox to listen to the messages, there is one new message available. My phone gets a mwi saying there is 1 new message for me. I wonder how the page knows for whom it shall display the voicemails. After that I tried "http://fs.ip:8080/api/domains/this/api/voicemail/web" This delivered: Error 404 Not Found ABYSS Web Server for XML-RPC For C/C++ version Xmlrpc-c 1.14.99 Hm, any ideas ? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl4m2gACgkQ4tZeNddg3dzIogCgnZybNeIkfsFbyKzSmusgUP5b hc8AoLYYO8Fi5RHKLEk7eXxUfzj+udIW =ofO3 -----END PGP SIGNATURE----- From intralanman at freeswitch.org Thu Jan 22 08:31:15 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 22 Jan 2009 16:31:15 +0000 Subject: [Freeswitch-users] voicemail web interface In-Reply-To: <49789B68.7060802@ewetel.de> References: <49789B68.7060802@ewetel.de> Message-ID: <49789F53.6040206@freeswitch.org> Helmut Kuper wrote: > http://fs.ip:8080/api/voicemail/web > > This led to a web page generating out of web-vm.tpl with 0 (zero) > voicemails. make sure that the voicemail is left for user at ip rather than user at some.domain.name > If I call my voicebox to listen to the messages, there is > one new message available. My phone gets a mwi saying there is 1 new > message for me. > > I wonder how the page knows for whom it shall display the voicemails. > it should have prompted you for a user/pass when you hit that page... -Ray -------------- next part -------------- A non-text attachment was scrubbed... Name: intralanman.vcf Type: text/x-vcard Size: 209 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/f147e94e/attachment.vcf From intralanman at freeswitch.org Thu Jan 22 08:36:23 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 22 Jan 2009 16:36:23 +0000 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> Message-ID: <4978A087.70301@freeswitch.org> > You use it on your own risk Actually, the fact that this is in this post at all SHOULD tell you NOT to use it. Unless you have proper licensing for g.729, this is likely to cause issues. -Ray -------------- next part -------------- A non-text attachment was scrubbed... Name: intralanman.vcf Type: text/x-vcard Size: 218 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/ef8084d6/attachment.vcf From regs at kinetix.gr Thu Jan 22 08:45:38 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 22 Jan 2009 18:45:38 +0200 Subject: [Freeswitch-users] mod_radius_cdr questions and thoughts Message-ID: <4978A2B2.2020905@kinetix.gr> I am trying to implement a radius based solution using FS. I have seen that the mod_radius_cdr module is actively maintained. so I have a few questions/remarks : 1) When I place a call and my radius server is down, the call blocks forever instead of just radius_timeout * radius_retries seconds (I have declared only one server). I would expect that FS would stop trying to send an Acc Start packet after some time and get on with the call. 2) I have also noticed that FS sends only 1 packet (I waited for a minute) instead of 3 (default in the config) since the first (and second) attempt failed. If my server was up (the port was responding) but it returned a req. failed answer would the above time-out be valid? 3) When I tried to load the dictionary.freeswitch to my freeradius server, it complained : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[33]: dict_addattr: Duplicate attribute name NAS-Port-Id" How can I overcome this? When I "instructed" my freeradius to only load the freesitch dictionary I got this : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[257]: unknown option "Freeswitch"" How can I use the dictionary with the freeradius? (or other radius servers?) 4) The radius attributes included in the current requests are a) hard-coded, b) limited in number. I think many of us would like to use more attributes. Or even better define what to include (and what to put in them) using a config file (the same maybe?) 5) Does the module send accounting packets only for the a-leg of a call or for both legs? (Maybe that could be configurable too). If anyone is interested in the above questions/remarks please post a reply. I would really like to know how many of the mailing list users are also interested in FS radius support and your opinions on the matter. Regards, -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From cparker at segv.org Thu Jan 22 09:09:15 2009 From: cparker at segv.org (Chris Parker) Date: Thu, 22 Jan 2009 11:09:15 -0600 Subject: [Freeswitch-users] mod_radius_cdr questions and thoughts In-Reply-To: <4978A2B2.2020905@kinetix.gr> References: <4978A2B2.2020905@kinetix.gr> Message-ID: <6569777d0901220909q99e80abvb4c6253afb65dc81@mail.gmail.com> On Thu, Jan 22, 2009 at 10:45 AM, Apostolos Pantsiopoulos wrote: > I am trying to implement a radius based solution > using FS. I have seen that the mod_radius_cdr module > is actively maintained. so I have a few questions/remarks : > > 1) When I place a call and my radius server is down, the > call blocks forever instead of just radius_timeout * radius_retries > seconds (I have declared only one server). I would expect that > FS would stop trying to send an Acc Start packet after some > time and get on with the call. I have not seen this behavior. If you can duplicate this, and propose a patch, it would be gladly welcomed. > > > 2) I have also noticed that FS sends only 1 packet (I waited for a minute) > instead of 3 (default in the config) since the first (and second) > attempt failed. > If my server was up (the port was responding) but it returned a req. failed > answer would the above time-out be valid? I have not seen this behavior. > > > 3) When I tried to load the dictionary.freeswitch to my freeradius > server, it complained : Don't do that. The dictionary is for use with the radiusclient library. FreeRADIUS already includes a dictionary for FreeSWITCH VSAs ( you may need to uncomment it to have it loaded into FreeRADIUS ). > 4) The radius attributes included in the current requests are > a) hard-coded, b) limited in number. I think many of us would like to > use more attributes. Or even better define what to include (and what to > put in them) using a > config file (the same maybe?) This has been proposed. There isn't yet a mechanism, though the intent is to use a general purpose FS VSA for this. The code needs to be added to the mod_radius_cdr module to allow that to be a run_time configuration option. > 5) Does the module send accounting packets only for the a-leg > of a call or for both legs? (Maybe that could be configurable too). > > If anyone is interested in the above questions/remarks please post > a reply. I would really like to know how many of the mailing list users > are also interested in FS radius support and your opinions on the matter. > Again, patches are welcome. :) -Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/00baf301/attachment.html From stevecrozz at gmail.com Thu Jan 22 09:38:32 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 22 Jan 2009 09:38:32 -0800 Subject: [Freeswitch-users] Error: eval_some_python() Error importing module In-Reply-To: References: <1232577645.4022.176.camel@dev03.cal.highergear.com> <87f2f3b90901211533g2b2d98acw6bd3a390a97fb3b9@mail.gmail.com> <11990ade0901211545o6ed1b6fqc57c0003447fc9a1@mail.gmail.com> <1232584925.4022.186.camel@dev03.cal.highergear.com> <11990ade0901211700j33a63f07r93abc7019e13d05@mail.gmail.com> Message-ID: <11990ade0901220938l452634f5sbf41c7eb4246cc67@mail.gmail.com> The reason I'd like to see the examples isn't so I can have a basic voicemail or IVR app, but so I can extend an example with my own complicated domain logic. --Stephen On Wed, Jan 21, 2009 at 5:14 PM, Brian West wrote: > Most if not all of this functionality is done with FreeSWITCH without > any need for python in the first place. > > For example you don't use python for voicemail, conferences, tone > generation but maybe IVR... which is what any of the languages are for. > > /b > > On Jan 21, 2009, at 7:00 PM, Stephen Crosby wrote: > >> I'd like to see an example of recording voicemail and a good >> conference application, tone generation, just a good IVR example would >> be nice too. >> >> --Stephen > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From guillaume.renaud at gmail.com Thu Jan 22 09:42:34 2009 From: guillaume.renaud at gmail.com (Guillaume Renaud) Date: Thu, 22 Jan 2009 12:42:34 -0500 Subject: [Freeswitch-users] Recording ULAW files In-Reply-To: References: <7009019.301232564751905.JavaMail.root@zimbra> Message-ID: <8d3f3fad0901220942i46bdffc0ub2d51c6f9d23b80@mail.gmail.com> I use Sun's AU format to make ulaw raw files friendlier, you might find it usefull. 2009/1/21 Brian West > mod_sndfile already registered .ul and .al ;) > /b > > On Jan 21, 2009, at 1:05 PM, freeswitch-users at 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. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/b5b859d0/attachment-0001.html From brian at freeswitch.org Thu Jan 22 09:44:20 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 22 Jan 2009 11:44:20 -0600 Subject: [Freeswitch-users] Recording ULAW files In-Reply-To: <8d3f3fad0901220942i46bdffc0ub2d51c6f9d23b80@mail.gmail.com> References: <7009019.301232564751905.JavaMail.root@zimbra> <8d3f3fad0901220942i46bdffc0ub2d51c6f9d23b80@mail.gmail.com> Message-ID: <6CB44D4E-28A2-4836-8FBC-DA4F55CC4346@freeswitch.org> mod_sndfile can record those too ;) /b On Jan 22, 2009, at 11:42 AM, Guillaume Renaud wrote: > I use Sun's AU format to make ulaw raw files friendlier, you might > find it usefull. From regs at kinetix.gr Thu Jan 22 10:59:12 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 22 Jan 2009 20:59:12 +0200 Subject: [Freeswitch-users] mod_radius_cdr questions and thoughts In-Reply-To: <6569777d0901220909q99e80abvb4c6253afb65dc81@mail.gmail.com> References: <4978A2B2.2020905@kinetix.gr> <6569777d0901220909q99e80abvb4c6253afb65dc81@mail.gmail.com> Message-ID: <4978C200.7030403@kinetix.gr> Chris Parker wrote: > On Thu, Jan 22, 2009 at 10:45 AM, Apostolos Pantsiopoulos > > wrote: > > I am trying to implement a radius based solution > using FS. I have seen that the mod_radius_cdr module > is actively maintained. so I have a few questions/remarks : > > 1) When I place a call and my radius server is down, the > call blocks forever instead of just radius_timeout * radius_retries > seconds (I have declared only one server). I would expect that > FS would stop trying to send an Acc Start packet after some > time and get on with the call. > > > I have not seen this behavior. If you can duplicate this, and propose > a patch, it would be gladly welcomed. I rebuilt and retried and the behavior persists. The call progress freezes and I get the following in the log : 2009-01-22 20:48:32 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/internal/9333 at xxx.xxx.xxx.xxx) State ROUTING 2009-01-22 20:48:32 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/9333 at xx.xxx.xxx.xxx SOFIA ROUTING 2009-01-22 20:48:32 [DEBUG] mod_radius_cdr.c:152 my_on_routing() [mod_radius_cdr] Entering my_on_routing After I hangup the client and issue a shutdown in FS I get the following : 2009-01-22 20:50:50 [CRIT] sofia.c:794 sofia_profile_thread_run() Waiting for 1 session(s) repeatedly and FS never exits. > > > > 2) I have also noticed that FS sends only 1 packet (I waited for a > minute) > instead of 3 (default in the config) since the first (and second) > attempt failed. > If my server was up (the port was responding) but it returned a > req. failed > answer would the above time-out be valid? > > > I have not seen this behavior. The same here after the rebuild. > > > > 3) When I tried to load the dictionary.freeswitch to my freeradius > server, it complained : > > > Don't do that. The dictionary is for use with the radiusclient > library. FreeRADIUS already includes a dictionary for FreeSWITCH VSAs > ( you may need to uncomment it to have it loaded into FreeRADIUS ). I cannot find any reference to Freeswitch in the freeradius integrated dictionaries (in the share folder). Can you pinpoint the directory that a dictionary.freeswitch (or other FS related dictionary) resides? > > > 4) The radius attributes included in the current requests are > a) hard-coded, b) limited in number. I think many of us would like to > use more attributes. Or even better define what to include (and > what to > put in them) using a > config file (the same maybe?) > > > This has been proposed. There isn't yet a mechanism, though the > intent is to use a general purpose FS VSA for this. The code needs to > be added to the mod_radius_cdr module to allow that to be a run_time > configuration option. A general purpose VSA that holds only one value or many? Or a mix (array like)? > > > 5) Does the module send accounting packets only for the a-leg > of a call or for both legs? (Maybe that could be configurable too). > > If anyone is interested in the above questions/remarks please post > a reply. I would really like to know how many of the mailing list > users > are also interested in FS radius support and your opinions on the > matter. > > > Again, patches are welcome. :) > > -Chris > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/b6b411b8/attachment.html From msc at freeswitch.org Thu Jan 22 11:14:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Jan 2009 11:14:45 -0800 Subject: [Freeswitch-users] voicemail web interface In-Reply-To: <49789B68.7060802@ewetel.de> References: <49789B68.7060802@ewetel.de> Message-ID: <87f2f3b90901221114u2bb19921gb929af144b6aa808@mail.gmail.com> On Thu, Jan 22, 2009 at 8:14 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > I tried to get voicemail web interface up and running. I found that we > need xml_rpc to open the 8080 port of FS. Then USer-profile needs > "http-allowed-api=voicemail" parameter. After that I left a message for > my account and tried to access my voicemails via web (I replaced fs.ip > with my IP): > > http://fs.ip:8080/api/voicemail/web > > This led to a web page generating out of web-vm.tpl with 0 (zero) > voicemails. If I call my voicebox to listen to the messages, there is > one new message available. My phone gets a mwi saying there is 1 new > message for me. > > I wonder how the page knows for whom it shall display the voicemails. > > > > After that I tried "http://fs.ip:8080/api/domains/this/api/voicemail/web" > > This delivered: > > Error 404 > > Not Found > > ABYSS Web Server for XML-RPC For C/C++ version Xmlrpc-c 1.14.99 > > > Hm, any ideas ? This was my bad. I documented it incorrectly on the wiki. I've updated the wiki to reflect the proper URL which is: http://fs.ip:8080/domains/this/api/voicemail/web Give it a try and let me know if you have any issues. -MC > > > regards > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkl4m2gACgkQ4tZeNddg3dzIogCgnZybNeIkfsFbyKzSmusgUP5b > hc8AoLYYO8Fi5RHKLEk7eXxUfzj+udIW > =ofO3 > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Thu Jan 22 11:15:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 22 Jan 2009 13:15:04 -0600 Subject: [Freeswitch-users] voicemail web interface In-Reply-To: <49789F53.6040206@freeswitch.org> References: <49789B68.7060802@ewetel.de> <49789F53.6040206@freeswitch.org> Message-ID: <191c3a030901221115h6bd1505dx3014955d9ce9ac21@mail.gmail.com> Start by knowing the exact domain of your directory domain (the name= in the tag) Lets say it's helmut.com and your ip is 1.2.3.4 and your extension is 1000 so if helmut.com is a real domain on the internet you could point the dns so helmut.com points at FS then you could go to http://helmut.com:8080/api/voicemail/web now you can just supply 1000 as the user name with no domain and it will assumed to be helmut.com instead you can also go to http://1.2.3.4:8080/api/voicemail/web and specify the user name 1000 at helmut.com and now you can control the domain that way. finally you can go to http://1.2.3.4:8080/domains/helmut.com/api/voicemail/web and supply the user name 1000 because you have explicitly set the domain in the url. you should also be specifying the same domain as the argument to voicemail app On Thu, Jan 22, 2009 at 10:31 AM, Raymond Chandler < intralanman at freeswitch.org> wrote: > Helmut Kuper wrote: > >> http://fs.ip:8080/api/voicemail/web >> >> This led to a web page generating out of web-vm.tpl with 0 (zero) >> voicemails. >> > make sure that the voicemail is left for user at ip rather than > user at some.domain.name > >> If I call my voicebox to listen to the messages, there is >> one new message available. My phone gets a mwi saying there is 1 new >> message for me. >> >> I wonder how the page knows for whom it shall display the voicemails. >> >> > it should have prompted you for a user/pass when you hit that page... > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/3096b6c0/attachment.html From msc at freeswitch.org Thu Jan 22 11:34:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Jan 2009 11:34:46 -0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <4978A087.70301@freeswitch.org> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> Message-ID: <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> On Thu, Jan 22, 2009 at 8:36 AM, Raymond Chandler wrote: > >> You use it on your own risk Also, G.729 is patent encumbered big-time. Instead of lining the pockets of lawyers and mega-corporations by perpetuating the use of a crusty old codec we should all twist arms and get our providers, device makers, etc. to use Speex. -MC > > Actually, the fact that this is in this post at all SHOULD tell you NOT to > use it. Unless you have proper licensing for g.729, this is likely to cause > issues. > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From sias at cpdata.co.za Thu Jan 22 12:24:06 2009 From: sias at cpdata.co.za (Sias Mey) Date: Thu, 22 Jan 2009 22:24:06 +0200 Subject: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches Message-ID: <20090122202406.GA21067@cpdata.co.za> Hi, Im trying to originate calls from a conference and use javascript to watch out for hangup events so I can use the data in the session to flesh out some database info. However it seems that Im having some strangeness. It might just be my code. So I include that. I run FreeSwitch Version 1.0.trunk (11226) Dialplan: confout.js: is attached I use API calls to pull one user into a conference. Then I use more api calls to do a conference dial to loopback/confout-1001 This should run the js and then bridge extension 1001 into the same conference. (I have hardcoded the additional extension for testing). I dont know if there is another way to get a conference dial to run a javascript file for information logging, but I am open to enlightenment. Oh im using conference dial because that provides clear audible progress to the other conference memebers as to what is actually happening with the new call. Any help would be greatly apreciated, Thanks in advance. Sias -------------- next part -------------- A non-text attachment was scrubbed... Name: confout.js Type: application/javascript Size: 873 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/317b51e2/attachment-0001.bin From mrene_lists at avgs.ca Thu Jan 22 12:07:43 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 22 Jan 2009 15:07:43 -0500 Subject: [Freeswitch-users] api_hangup_hook not actually executing the command In-Reply-To: <497605DE.8010307@novatex.com.au> References: <497605DE.8010307@novatex.com.au> Message-ID: Its global_setvar not set_global Math On Tue, Jan 20, 2009 at 12:11 PM, Scott Ellis wrote: > The on answer is fine, the api_hangup_hook gets to the point where it > wants to execute, but then nothing happens. Any thoughts? > > > 2009-01-21 03:08:00 [DEBUG] switch_channel.c:1773 > switch_channel_perform_mark_answered() sofia/internal/Line_9 at 10.0.0.9 > execute on answer: set_global(10.0.0.19_INCALL=true)2009-01-21 03:08:00 > [DEBUG] mod_dptools.c:726 set_global_function() SET GLOBAL > [10.0.0.19_INCALL]=[true] > > 2009-01-21 03:08:09 [DEBUG] switch_core_state_machine.c:416 > switch_core_session_run() Hangup Command > set_global(10.0.0.19_INCALL=false): > > It does not execute - the global variable is not set...quite odd. > > Scott > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/5a292f31/attachment.html From brian at freeswitch.org Thu Jan 22 12:59:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 22 Jan 2009 14:59:35 -0600 Subject: [Freeswitch-users] api_hangup_hook not actually executing the command In-Reply-To: References: <497605DE.8010307@novatex.com.au> Message-ID: <04D6A331-1E76-427F-B6D1-52B9553F1D9E@freeswitch.org> I would use mod_limit and not futz with global anything. /b On Jan 22, 2009, at 2:07 PM, Mathieu Rene wrote: > Its global_setvar not set_global > > Math From brian at freeswitch.org Thu Jan 22 13:03:09 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 22 Jan 2009 15:03:09 -0600 Subject: [Freeswitch-users] Alternate config sets. Message-ID: FreeSWITCHers, I have started to put alternate config examples up for people to check out. Right now it only has the Softphone example... if you have ideas or examples to add to this please let me know: http://svn.freeswitch.org/svn/configs/softphone/ Thanks, Brian West FreeSWITCH.org From damin at nacs.net Thu Jan 22 13:11:35 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Thu, 22 Jan 2009 16:11:35 -0500 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> Message-ID: <001301c97cd6$02107b90$063172b0$@net> > >> You use it on your own risk > > Also, G.729 is patent encumbered big-time. Instead of lining the > pockets of lawyers and mega-corporations by perpetuating the use of a > crusty old codec we should all twist arms and get our providers, > device makers, etc. to use Speex. Yeah.. let me know when you get Cisco to add Speex support to IOS! ;) From brian at freeswitch.org Thu Jan 22 13:14:29 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 22 Jan 2009 15:14:29 -0600 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <001301c97cd6$02107b90$063172b0$@net> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> Message-ID: <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> They added iLBC. /b On Jan 22, 2009, at 3:11 PM, Gregory Boehnlein wrote: > > Yeah.. let me know when you get Cisco to add Speex support to IOS! ;) From msc at freeswitch.org Thu Jan 22 13:21:54 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 22 Jan 2009 13:21:54 -0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <001301c97cd6$02107b90$063172b0$@net> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> Message-ID: <238ADE09-54EF-469A-A522-9FD1653665E6@freeswitch.org> On Jan 22, 2009, at 1:11 PM, "Gregory Boehnlein" wrote: >>>> You use it on your own risk >> >> Also, G.729 is patent encumbered big-time. Instead of lining the >> pockets of lawyers and mega-corporations by perpetuating the use of a >> crusty old codec we should all twist arms and get our providers, >> device makers, etc. to use Speex. > > Yeah.. let me know when you get Cisco to add Speex support to IOS! ;) Who needs Crisco anyway? :p > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kristian.kielhofner at gmail.com Thu Jan 22 13:31:39 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 22 Jan 2009 16:31:39 -0500 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> Message-ID: <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> On Thu, Jan 22, 2009 at 4:14 PM, Brian West wrote: > They added iLBC. > > /b > Very true. I wish I knew who got them to do that! Speex would be awesome but I'm not holding my breath.... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Thu Jan 22 13:35:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 22 Jan 2009 15:35:53 -0600 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> Message-ID: Speex, while nice I think it would use more resources in some cases. /b On Jan 22, 2009, at 3:31 PM, Kristian Kielhofner wrote: > > Speex would be awesome but I'm not holding my breath.... From msc at freeswitch.org Thu Jan 22 13:40:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Jan 2009 13:40:46 -0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> Message-ID: <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> On Thu, Jan 22, 2009 at 1:35 PM, Brian West wrote: > Speex, while nice I think it would use more resources in some cases. True. Occasionally more resources, always less licensing fees... ;) -MC > > /b > > On Jan 22, 2009, at 3:31 PM, Kristian Kielhofner wrote: > >> >> Speex would be awesome but I'm not holding my breath.... > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kristian.kielhofner at gmail.com Thu Jan 22 13:55:32 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 22 Jan 2009 16:55:32 -0500 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> Message-ID: <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> On Thu, Jan 22, 2009 at 4:40 PM, Michael Collins wrote: > On Thu, Jan 22, 2009 at 1:35 PM, Brian West wrote: >> Speex, while nice I think it would use more resources in some cases. > > True. Occasionally more resources, always less licensing fees... ;) > -MC Less money on licensing, more money on DSP/processor resources. :) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From intralanman at freeswitch.org Thu Jan 22 14:08:53 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 22 Jan 2009 22:08:53 +0000 Subject: [Freeswitch-users] Alternate config sets. In-Reply-To: References: Message-ID: <4978EE75.9020507@freeswitch.org> Brian West wrote: > FreeSWITCHers, > I have started to put alternate config examples up for people to > check out. Right now it only has the Softphone example... if you have > ideas or examples to add to this please let me know: > i just added sbc, curl, and insideout config sets... all are pretty rough still, but should be shaping up over the next few days. -Ray -------------- next part -------------- A non-text attachment was scrubbed... Name: intralanman.vcf Type: text/x-vcard Size: 209 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/2144aa3d/attachment.vcf From anthony.minessale at gmail.com Thu Jan 22 14:25:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 22 Jan 2009 16:25:54 -0600 Subject: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches In-Reply-To: <20090122202406.GA21067@cpdata.co.za> References: <20090122202406.GA21067@cpdata.co.za> Message-ID: <191c3a030901221425p379ad74j7f866eee4596a6a9@mail.gmail.com> Try this (update to svn trunk first) then place your call as usual then in foo.js // dumps the event to text/plain env = request.dumpENV("text"); // dumps the event to text/xml xmlenv = request.dumpENV("xml"); // makes an XML obj from the xml text xinfo = new XML("" + xmlenv + ""); // dump the plain text event data consoleLog("info", env + "\n"); // dump the xml event data consoleLog("info", xmlenv + "\n"); // Get a header from the event object consoleLog("warning", "media ip was [" + request.getHeader("local_media_ip") + "]\n"); // Get the same header from the xml object consoleLog("warning", "media ip was [" + xinfo.event.headers.local_media_ip + "]\n"); On Thu, Jan 22, 2009 at 2:24 PM, Sias Mey wrote: > Hi, > > Im trying to originate calls from a conference and use javascript to > watch out for hangup events so I can use the data in the session to > flesh out some database info. However it seems that Im having some > strangeness. It might just be my code. So I include that. > > I run FreeSwitch Version 1.0.trunk (11226) > > Dialplan: > > > > > > > confout.js: > is attached > > I use API calls to pull one user into a conference. Then I use more api > calls to do a conference dial to loopback/confout-1001 > > This should run the js and then bridge extension 1001 into the same > conference. > (I have hardcoded the additional extension for testing). I dont know if > there is another way to get a conference dial to run a javascript file for > information logging, but I am open to enlightenment. > > Oh im using conference dial because that provides clear audible progress to > the other conference memebers as to what is actually happening with the new > call. > > Any help would be greatly apreciated, Thanks in advance. > Sias > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/f9466f95/attachment-0001.html From msc at freeswitch.org Thu Jan 22 15:04:11 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 22 Jan 2009 15:04:11 -0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> Message-ID: On Jan 22, 2009, at 1:55 PM, Kristian Kielhofner wrote: > On Thu, Jan 22, 2009 at 4:40 PM, Michael Collins > wrote: >> On Thu, Jan 22, 2009 at 1:35 PM, Brian West >> wrote: >>> Speex, while nice I think it would use more resources in some cases. >> >> True. Occasionally more resources, always less licensing fees... ;) >> -MC > > Less money on licensing, more money on DSP/processor resources. :) > That delta shrinks as processing power gets cheaper. I wonder if g729 licenses will get cheaper over time as well? I wouldn't take that bet. ;) -MC > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kristian.kielhofner at gmail.com Thu Jan 22 15:13:01 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 22 Jan 2009 18:13:01 -0500 Subject: [Freeswitch-users] mod_g729 In-Reply-To: References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> Message-ID: <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> On Thu, Jan 22, 2009 at 6:04 PM, Michael S Collins wrote: > > That delta shrinks as processing power gets cheaper. I wonder if g729 > licenses will get cheaper over time as well? I wouldn't take that > bet. ;) > > -MC Also remembers what happens to volume pricing. More of a break on licenses... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Thu Jan 22 15:15:13 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 22 Jan 2009 17:15:13 -0600 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> Message-ID: Not really what I would call a break... but at some point in the $1.6 million range you stop paying. /b On Jan 22, 2009, at 5:13 PM, Kristian Kielhofner wrote: > > Also remembers what happens to volume pricing. More of a break on > licenses... From msc at freeswitch.org Thu Jan 22 15:35:17 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 22 Jan 2009 15:35:17 -0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> Message-ID: <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> On Jan 22, 2009, at 3:15 PM, Brian West wrote: > Not really what I would call a break... but at some point in the $1.6 > million range you stop paying. > > /b Like I said, OSS FTW baby! -MC > > > On Jan 22, 2009, at 5:13 PM, Kristian Kielhofner wrote: > >> >> Also remembers what happens to volume pricing. More of a break on >> licenses... > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kristian.kielhofner at gmail.com Thu Jan 22 15:48:28 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 22 Jan 2009 18:48:28 -0500 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> Message-ID: <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> On Thu, Jan 22, 2009 at 6:35 PM, Michael S Collins wrote: > > > On Jan 22, 2009, at 3:15 PM, Brian West wrote: > >> Not really what I would call a break... but at some point in the $1.6 >> million range you stop paying. >> >> /b > > Like I said, OSS FTW baby! > -MC Quite the contrary. If Speex means you require faster, more capable DSPs, you are going to continue paying for them per unit. At $1.6mil you stop paying for G.729 licenses (basically a fixed cost at that point, regardless of volume or quantity). I'm sure most of the big guys paying for G.729 have no problem with that. I love OSS and Speex as much as any of us; I'm just trying to play devil's advocate and attempt to explain some of the strategy from those continuing to pay for G.729... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From msc at freeswitch.org Thu Jan 22 16:19:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Jan 2009 16:19:00 -0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> Message-ID: <87f2f3b90901221619i71eb28e5y123784cc85a98672@mail.gmail.com> On Thu, Jan 22, 2009 at 3:48 PM, Kristian Kielhofner wrote: > On Thu, Jan 22, 2009 at 6:35 PM, Michael S Collins wrote: >> >> >> On Jan 22, 2009, at 3:15 PM, Brian West wrote: >> >>> Not really what I would call a break... but at some point in the $1.6 >>> million range you stop paying. >>> >>> /b >> >> Like I said, OSS FTW baby! >> -MC > > Quite the contrary. > > If Speex means you require faster, more capable DSPs, you are going to > continue paying for them per unit. At $1.6mil you stop paying for > G.729 licenses (basically a fixed cost at that point, regardless of > volume or quantity). > > I'm sure most of the big guys paying for G.729 have no problem with that. > > I love OSS and Speex as much as any of us; I'm just trying to play > devil's advocate and attempt to explain some of the strategy from > those continuing to pay for G.729... You're a good devil! ;) I still have to wonder if g.729 is measurably "better" than Speex or any other 8kHz CELP-based codec. What is that measurement? That's how you can tell if it's "worth it" or not. As for me, I'd rather not get locked in. Patent encumbrances are nasty and I prefer to avoid them where possible. I think bigger businesses are starting to wise up to that fact as well. I think Polycom's model is going to catch on: codecs are "basically free" but are not open source. Not my favorite option but if I get a license to use the codec in perpetuity then I'm feeling pretty good. -MC > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveu at coppice.org Thu Jan 22 16:30:06 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 23 Jan 2009 08:30:06 +0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> Message-ID: <49790F8E.8000505@coppice.org> Kristian Kielhofner wrote: > On Thu, Jan 22, 2009 at 6:35 PM, Michael S Collins wrote: > >> On Jan 22, 2009, at 3:15 PM, Brian West wrote: >> >> >>> Not really what I would call a break... but at some point in the $1.6 >>> million range you stop paying. >>> >>> /b >>> >> Like I said, OSS FTW baby! >> -MC >> > > Quite the contrary. > > If Speex means you require faster, more capable DSPs, you are going to > continue paying for them per unit. At $1.6mil you stop paying for > G.729 licenses (basically a fixed cost at that point, regardless of > volume or quantity). > > I'm sure most of the big guys paying for G.729 have no problem with that. > > I love OSS and Speex as much as any of us; I'm just trying to play > devil's advocate and attempt to explain some of the strategy from > those continuing to pay for G.729... > Depends what you are after. Speex offers the quality of G.729 at around the same processing load. However, nobody seems to want to pay for the processing load of G.729. Almost everything uses G.729A. Half the processing load, but significantly poorer quality. VoIP is mostly a race to the bottom, and people wonder why it makes no money for provides. :-\ Regards, Steve From brian at freeswitch.org Thu Jan 22 16:32:18 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 22 Jan 2009 18:32:18 -0600 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <49790F8E.8000505@coppice.org> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org> Message-ID: <12659B55-E309-42F1-BB78-313CDDCBA7CF@freeswitch.org> makes me laugh... everyone else is racing to stay on 8k... I'm running in the other direction! : ) 16k, 32k and 48k voip... much better. /b On Jan 22, 2009, at 6:30 PM, Steve Underwood wrote: > Depends what you are after. Speex offers the quality of G.729 at > around > the same processing load. However, nobody seems to want to pay for the > processing load of G.729. Almost everything uses G.729A. Half the > processing load, but significantly poorer quality. > > VoIP is mostly a race to the bottom, and people wonder why it makes no > money for provides. :-\ > > Regards, > Steve From steveu at coppice.org Thu Jan 22 16:33:03 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 23 Jan 2009 08:33:03 +0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> Message-ID: <4979103F.9040300@coppice.org> Michael S Collins wrote: > That delta shrinks as processing power gets cheaper. I wonder if g729 > licenses will get cheaper over time as well? I wouldn't take that > bet. ;) > Economics 101: The pricing of the licences is directly related to G.729's lock on the market. The only reason for the prices to come down is if G.729A loses its grip on the IP phone market. Regards, Steve From msc at freeswitch.org Thu Jan 22 16:37:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Jan 2009 16:37:44 -0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <49790F8E.8000505@coppice.org> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org> Message-ID: <87f2f3b90901221637i2ce1e142h470f6413837a1aee@mail.gmail.com> On Thu, Jan 22, 2009 at 4:30 PM, Steve Underwood wrote: > Kristian Kielhofner wrote: >> On Thu, Jan 22, 2009 at 6:35 PM, Michael S Collins wrote: >> >>> On Jan 22, 2009, at 3:15 PM, Brian West wrote: >>> >>> >>>> Not really what I would call a break... but at some point in the $1.6 >>>> million range you stop paying. >>>> >>>> /b >>>> >>> Like I said, OSS FTW baby! >>> -MC >>> >> >> Quite the contrary. >> >> If Speex means you require faster, more capable DSPs, you are going to >> continue paying for them per unit. At $1.6mil you stop paying for >> G.729 licenses (basically a fixed cost at that point, regardless of >> volume or quantity). >> >> I'm sure most of the big guys paying for G.729 have no problem with that. >> >> I love OSS and Speex as much as any of us; I'm just trying to play >> devil's advocate and attempt to explain some of the strategy from >> those continuing to pay for G.729... >> > Depends what you are after. Speex offers the quality of G.729 at around > the same processing load. However, nobody seems to want to pay for the > processing load of G.729. Almost everything uses G.729A. Half the > processing load, but significantly poorer quality. > > VoIP is mostly a race to the bottom, and people wonder why it makes no > money for provides. :-\ Amen! A voice of reason. "Let's use a really, REALLY low-quality codec to save bandwidth and then make it so the tech support calls from the US go to places where the agents don't really speak English. That way it will be both hard to hear and difficult to understand. That'll save BOATLOADS of money!" Uh yeah. First off, Americans can barely speak English. Secondly, if you make it difficult for people to hear and/or understand each other then all that money you're saving with smaller bandwidth will go to things like higher telecom costs (longer phone calls) and higher payroll costs to cover the additional personnel needed by these call centers. As usual, thank you Steve for an enlightened viewpoint that makes others think about the big picture. -MC > Regards, > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Thu Jan 22 16:39:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Jan 2009 16:39:25 -0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <4979103F.9040300@coppice.org> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <4979103F.9040300@coppice.org> Message-ID: <87f2f3b90901221639x5823323bmc6caac7959fc2f60@mail.gmail.com> On Thu, Jan 22, 2009 at 4:33 PM, Steve Underwood wrote: > Michael S Collins wrote: >> That delta shrinks as processing power gets cheaper. I wonder if g729 >> licenses will get cheaper over time as well? I wouldn't take that >> bet. ;) >> > Economics 101: The pricing of the licences is directly related to > G.729's lock on the market. The only reason for the prices to come down > is if G.729A loses its grip on the IP phone market. In other words they only get cheaper if all of us band together and don't use them? ;) That works for me! -MC > > Regards, > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kristian.kielhofner at gmail.com Thu Jan 22 16:55:30 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 22 Jan 2009 19:55:30 -0500 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <49790F8E.8000505@coppice.org> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org> Message-ID: <2d9149cd0901221655k798a417dmfae848cd6186085c@mail.gmail.com> On Thu, Jan 22, 2009 at 7:30 PM, Steve Underwood wrote: > Depends what you are after. Speex offers the quality of G.729 at around > the same processing load. However, nobody seems to want to pay for the > processing load of G.729. Almost everything uses G.729A. Half the > processing load, but significantly poorer quality. Throughout most of this thread I've been using G.729 and G.729a interchangeably. That's sloppy on my part. Most of the time I meant G.729a because that is what most people (everyone?) uses. I was under the impression the quality difference wasn't that significant. I can't say if I've ever knowingly used "true" G.729. It certainly doesn't help that G.729a is allegedly completely compatible with G.729, to the point where I don't think it's even valid to use "G.729a" or annexa in an SDP... > VoIP is mostly a race to the bottom, and people wonder why it makes no > money for provides. :-\ Word Steve, word! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Thu Jan 22 16:57:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 22 Jan 2009 18:57:36 -0600 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <2d9149cd0901221655k798a417dmfae848cd6186085c@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org> <2d9149cd0901221655k798a417dmfae848cd6186085c@mail.gmail.com> Message-ID: <77C3777D-3C45-4757-BC72-0A10E65AAD6D@freeswitch.org> Nope its not valid.. tell Cisco this please! :P /b On Jan 22, 2009, at 6:55 PM, Kristian Kielhofner wrote: > I don't think it's even > valid to use "G.729a" or annexa in an SDP... From simon0922 at gmail.com Fri Jan 23 09:15:09 2009 From: simon0922 at gmail.com (Simon Leck) Date: Fri, 23 Jan 2009 09:15:09 -0800 Subject: [Freeswitch-users] freeswitch memory leak issue Message-ID: Hi All, I am experiencing memory leak issue in FreeSwitch? Anybody knows how this can be resolved, please email me. Thanks in advance to everybody for your kind assistance. Thanks Simon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/f215e9f7/attachment.html From brian at freeswitch.org Thu Jan 22 17:20:13 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 22 Jan 2009 19:20:13 -0600 Subject: [Freeswitch-users] freeswitch memory leak issue In-Reply-To: References: Message-ID: <8C8A023D-28BD-4996-82C6-A465E3FE292E@freeswitch.org> What rev? What OS? What exactly are you doing? /b On Jan 23, 2009, at 11:15 AM, Simon Leck wrote: > Hi All, > > I am experiencing memory leak issue in FreeSwitch? Anybody knows > how this can be resolved, please email me. > > Thanks in advance to everybody for your kind assistance. > > Thanks > Simon > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090122/d1c07c32/attachment.html From intralanman at freeswitch.org Thu Jan 22 17:20:25 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 23 Jan 2009 01:20:25 +0000 Subject: [Freeswitch-users] freeswitch memory leak issue In-Reply-To: References: Message-ID: <49791B59.9000507@freeswitch.org> what's the bug number for this issue? if the answer is "there isn't one", then please post one -Ray Simon Leck wrote: > > Hi All, > > > > I am experiencing memory leak issue in FreeSwitch? Anybody knows how > this can be resolved, please email me. > > > > Thanks in advance to everybody for your kind assistance. > > > > Thanks > > Simon > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/ad96450e/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: intralanman.vcf Type: text/x-vcard Size: 209 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/ad96450e/attachment.vcf From jlists at skopis.com Thu Jan 22 17:27:42 2009 From: jlists at skopis.com (John Skopis (Lists)) Date: Thu, 22 Jan 2009 19:27:42 -0600 Subject: [Freeswitch-users] VMWare voice quality In-Reply-To: <87f2f3b90901150932r2d7bf58cxd4d63b30e464e48f@mail.gmail.com> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <87f2f3b90901150932r2d7bf58cxd4d63b30e464e48f@mail.gmail.com> Message-ID: <49791D0E.5060601@skopis.com> Michael Collins wrote: > If anyone figures this out please post it to this thread. I am working > on a wiki page for the VMWare appliance and I would like to be able to > inform people on how to handle this situation. I had some issues under vmware fusion. They were resolved by adding clock=pit [1] to the kernel boot params and switching to host-only networking, and running natd + ipfw on the host system. The vmware natd would probably also work. haven't tried myself. The clock=pit is the big kicker. Also, recompiling the kernel with HZ=100 might help to reduce the load on the host system. [2] Though, with a small number of vms/vcpus on decent hw the number of context switches probably won't have much of an effect. [1] www.vmware.com/pdf/vmware_timekeeping.pdf [2] http://communities.vmware.com/docs/DOC-3580 > > Also, IIUC, those running VMWare Fusion on Macs are not experiencing > this, correct? What about those using a hypervisor like ESXi? Any > known issues? > > Thanks, > MC > > On Thu, Jan 15, 2009 at 9:15 AM, Ken Rice wrote: >> On 1/15/09 11:01 AM, "Remko Kloosterman" wrote: >> >>> Hello Ken, hello all, >>> >>> I just read about the FreeSWITCH VMware applicance. I'm curious about >>> your experiences with the audio quality on VMWare, so here's a new >>> thread. >>> >>> I've installed freeswitch on VMware Server for Windows. The IVR audio >>> always plays choppy, while the server itself has no performance issues. >>> The same poor voice quality also goes for Asterisk or Yate, even on a >>> very fast VMware ESX system. >>> >>> Did you experience the same and/or do you have pointers on how to >>> troubleshoot and fix this? >> >> There is a high resolution timer you need to enable on vmware... I'm not >> familiar enuff with all the versions of vmware to advise there that switch >> is, but they have a couple of articles on it in their knowledge base >> >> >> From kristian.kielhofner at gmail.com Thu Jan 22 17:59:08 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 22 Jan 2009 20:59:08 -0500 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <77C3777D-3C45-4757-BC72-0A10E65AAD6D@freeswitch.org> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org> <2d9149cd0901221655k798a417dmfae848cd6186085c@mail.gmail.com> <77C3777D-3C45-4757-BC72-0A10E65AAD6D@freeswitch.org> Message-ID: <2d9149cd0901221759w79eb12abk15eda119e4ae099a@mail.gmail.com> Haha, I knew I'd seen it *somewhere*! ;) On Thu, Jan 22, 2009 at 7:57 PM, Brian West wrote: > Nope its not valid.. tell Cisco this please! :P > > /b > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From simon0922 at gmail.com Fri Jan 23 11:17:42 2009 From: simon0922 at gmail.com (Simon Leck) Date: Fri, 23 Jan 2009 11:17:42 -0800 Subject: [Freeswitch-users] freeswitch memory leak issue In-Reply-To: <8C8A023D-28BD-4996-82C6-A465E3FE292E@freeswitch.org> References: <8C8A023D-28BD-4996-82C6-A465E3FE292E@freeswitch.org> Message-ID: Hi Brian, Thanks for your prompt reply. We are currently using FreeSwitch rev 1.02. As for OS, we are using CentOS 5.2. We are using Java application to connect to MySQL database by JDBC APIs, the issue we faced is that after many calls are made, the java application will reflect the error message shown below. Furthermore the java application will not release the memory even though, the application had disconnect from the connected session Here are the errors message, "Caused by: java.lang.OutOfMemoryError: PermGen space" Thanks again Brian in advance for your kind assistance. Thanks Simon From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, January 22, 2009 5:20 PM To: freeswitch-users at lists.freeswitch.org Subject: [?? Probable Spam] Re: [Freeswitch-users] freeswitch memory leak issue What rev? What OS? What exactly are you doing? /b On Jan 23, 2009, at 11:15 AM, Simon Leck wrote: Hi All, I am experiencing memory leak issue in FreeSwitch? Anybody knows how this can be resolved, please email me. Thanks in advance to everybody for your kind assistance. Thanks Simon _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/9b660f8d/attachment.html From dave at 3c.co.uk Thu Jan 22 21:09:32 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 23 Jan 2009 05:09:32 +0000 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <49790F8E.8000505@coppice.org> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org> Message-ID: <4979510C.90303@3c.co.uk> Steve Underwood wrote: > Depends what you are after. Speex offers the quality of G.729 at around > the same processing load. However, nobody seems to want to pay for the > processing load of G.729. Almost everything uses G.729A. Half the > processing load, but significantly poorer quality. > > VoIP is mostly a race to the bottom, and people wonder why it makes no > money for provides. :-\ > And, at the wholesale level, it makes no sense whatsoever to compress calls any more: bandwidth is so cheap (and has been for a while) that the loss in call quality - especially from tandem compressions - and the increased processing requirements and other bits of expense do not stack up. Case in point: we moved a route from G.711 to G.729, and saw the ACD drop from over 10 to under 7 minutes. It was a route to mobiles, so the audio was being recompressed with the GSM codec on its way to the handsets. Economically, had we carried on using G.729, we'd have lost about 30% of our margin on that route. --Dave -- David Knell, Director, 3C Limited T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623 http://www.3c.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/a2af423c/attachment.html From regs at kinetix.gr Thu Jan 22 06:52:16 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 22 Jan 2009 16:52:16 +0200 Subject: [Freeswitch-users] mod_radius_cdr questions and thoughts Message-ID: <49788820.1000208@kinetix.gr> I am trying to implement a radius based solution using FS. I have seen that the mod_radius_cdr module is actively maintained. so I have a few questions/remarks : 1) When I place a call and my radius server is down, the call blocks forever instead of just radius_timeout * radius_retries seconds (I have declared only one server). I would expect that FS would stop trying to send an Acc Start packet after some time and get on with the call. 2) I have also noticed that FS sends only 1 packet (I waited for a minute) instead of 3 (default in the config) since the first (and second) attempt failed. If my server was up (the port was responding) but it returned a req. failed answer would the above time-out be valid? 3) When I tried to load the dictionary.freeswitch to my freeradius server, it complained : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[33]: dict_addattr: Duplicate attribute name NAS-Port-Id" How can I overcome this? When I "instructed" my freeradius to only load the freesitch dictionary I got this : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[257]: unknown option "Freeswitch"" How can I use the dictionary with the freeradius? (or other radius servers?) 4) The radius attributes included in the current requests are a) hard-coded, b) limited in number. I think many of us would like to use more attributes. Or even better define what to include (and what to put in them) using a config file (the same maybe?) 5) Does the module send accounting packets only for the a-leg of a call or for both legs? (Maybe that could be configurable too). If anyone is interested in the above questions/remarks please post a reply. I would really like to know how many of the mailing list users are also interested in FS radius support and your opinions on the matter. Regards, -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From krivushinme at rn-inform.tomsk.ru Thu Jan 22 22:49:19 2009 From: krivushinme at rn-inform.tomsk.ru (=?utf-8?b?0JrRgNC40LLRg9GI0LjQvSDQnNC40YXQsNC40Ls=?=) Date: Fri, 23 Jan 2009 12:49:19 +0600 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <87f2f3b90901221619i71eb28e5y123784cc85a98672@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <87f2f3b90901221619i71eb28e5y123784cc85a98672@mail.gmail.com> Message-ID: <200901231249.19207.krivushinme@rn-inform.tomsk.ru> I wrote it becouse in Russia * is very popular. And it have g729. I want to make popular FS in my country. We have not patent issues, but I more like speex and celt - it's better in my opinion - 8 kHz is past century! It's century of wb VoIP and not 8khz TDM! : ) Only for those who very need g729 in countries like Russia I made it public. I dont tell you use it. In fact I tell to not use it - but if you cannot, you can : ) -- ? ?????????, ???????? ?????? ??????? ?????????? ?????? ????????????????, ??? "??-??????" ?????? ? ?.??????, ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru mail: KrivushinME at rn-inform.tomsk.ru From helmut.kuper at ewetel.de Fri Jan 23 00:14:16 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 23 Jan 2009 09:14:16 +0100 Subject: [Freeswitch-users] voicemail web interface In-Reply-To: <191c3a030901221115h6bd1505dx3014955d9ce9ac21@mail.gmail.com> References: <49789B68.7060802@ewetel.de> <49789F53.6040206@freeswitch.org> <191c3a030901221115h6bd1505dx3014955d9ce9ac21@mail.gmail.com> Message-ID: <49797C58.7000607@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, well, was all my fault ... I was logged in with xml rpc user (freeswitch/work). Both urls working the same way on my side: entered in both pages simply extension number + vm-password. Both pages results in the same output. Very very impressive stuff :) Unfortunately I didn't get the nice flash players playing the messages. When I click on play button player's title bar changes to "1. undefined" and browser's status line shows: "Transferring data from ..." regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl5fFgACgkQ4tZeNddg3dy+JACgh0tXzvVaFfdErWqqsIJq1q1I S3kAni8H97R4JRyW4sOEUc7+au8/SRp6 =hqxE -----END PGP SIGNATURE----- From sias at cpdata.co.za Fri Jan 23 01:42:01 2009 From: sias at cpdata.co.za (Sias Mey) Date: Fri, 23 Jan 2009 11:42:01 +0200 Subject: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches In-Reply-To: <191c3a030901221425p379ad74j7f866eee4596a6a9@mail.gmail.com> References: <20090122202406.GA21067@cpdata.co.za> <191c3a030901221425p379ad74j7f866eee4596a6a9@mail.gmail.com> Message-ID: <20090123094201.GA6386@cpdata.co.za> Hmm ok... updated to the latest SVN and tried your suggsestion however all I can see happening in the console is 2009-01-23 11:35:44 [ERR] hangup.js:2 mod_spidermonkey() ReferenceError: request is not defined (obviously I renamed foo.js to hangup here) thanks again for the help. On Thu, Jan 22, 2009 at 04:25:54PM -0600, Anthony Minessale wrote: > Try this (update to svn trunk first) > data="{api_hangup_hook=jsapi::foo.js}sofia/default/[1]user at dest.com"/> > then place your call as usual > then in foo.js > // dumps the event to text/plain > env = request.dumpENV("text"); > // dumps the event to text/xml > xmlenv = request.dumpENV("xml"); > // makes an XML obj from the xml text > xinfo = new XML("" + xmlenv + ""); > // dump the plain text event data > consoleLog("info", env + "\n"); > // dump the xml event data > consoleLog("info", xmlenv + "\n"); > // Get a header from the event object > consoleLog("warning", "media ip was [" + > request.getHeader("local_media_ip") + "]\n"); > // Get the same header from the xml object > consoleLog("warning", "media ip was [" + > xinfo.event.headers.local_media_ip + "]\n"); > > On Thu, Jan 22, 2009 at 2:24 PM, Sias Mey <[2]sias at cpdata.co.za> wrote: > > Hi, > Im trying to originate calls from a conference and use javascript to > watch out for hangup events so I can use the data in the session to > flesh out some database info. However it seems that Im having some > strangeness. It might just be my code. So I include that. > I run FreeSwitch Version 1.0.trunk (11226) > Dialplan: > > expression="^confout-(10\d{2})$"> > > > > confout.js: > is attached > I use API calls to pull one user into a conference. Then I use more > api calls to do a conference dial to loopback/confout-1001 > This should run the js and then bridge extension 1001 into the same > conference. > (I have hardcoded the additional extension for testing). I dont know > if there is another way to get a conference dial to run a javascript > file for information logging, but I am open to enlightenment. > Oh im using conference dial because that provides clear audible > progress to the other conference memebers as to what is actually > happening with the new call. > Any help would be greatly apreciated, Thanks in advance. > Sias > _______________________________________________ > Freeswitch-users mailing list > [3]Freeswitch-users at lists.freeswitch.org > [4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[5]http://lists.freeswitch.org/mailman/options/freeswitc > h-users > [6]http://www.freeswitch.org > > -- > Anthony Minessale II > FreeSWITCH [7]http://www.freeswitch.org/ > ClueCon [8]http://www.cluecon.com/ > AIM: anthm > [9]MSN:anthony_minessale at hotmail.com > GTALK/JABBER/[10]PAYPAL:anthony.minessale at gmail.com > IRC: [11]irc.freenode.net #freeswitch > FreeSWITCH Developer Conference > [12]sip:888 at conference.freeswitch.org > [13]iax:guest at conference.freeswitch.org/888 > [14]googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > References > > 1. mailto:user at dest.com > 2. mailto:sias at cpdata.co.za > 3. mailto:Freeswitch-users at lists.freeswitch.org > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 5. http://lists.freeswitch.org/mailman/options/freeswitch-users > 6. http://www.freeswitch.org/ > 7. http://www.freeswitch.org/ > 8. http://www.cluecon.com/ > 9. mailto:MSN%3Aanthony_minessale at hotmail.com > 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 11. http://irc.freenode.net/ > 12. mailto:sip%3A888 at conference.freeswitch.org > 13. http://iax:guest at conference.freeswitch.org/888 > 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sias at cpdata.co.za Fri Jan 23 01:50:36 2009 From: sias at cpdata.co.za (Sias Mey) Date: Fri, 23 Jan 2009 11:50:36 +0200 Subject: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches In-Reply-To: <191c3a030901221425p379ad74j7f866eee4596a6a9@mail.gmail.com> References: <20090122202406.GA21067@cpdata.co.za> <191c3a030901221425p379ad74j7f866eee4596a6a9@mail.gmail.com> Message-ID: <20090123095036.GA6568@cpdata.co.za> Wait sory ignore my previous reply... I only just realized you were actually routing through the javascript xml_rpc module. and I didnt actually have the api.js file in my scripts dir. let me see what this does before you worry about it any more ;-) On Thu, Jan 22, 2009 at 04:25:54PM -0600, Anthony Minessale wrote: > Try this (update to svn trunk first) > data="{api_hangup_hook=jsapi::foo.js}sofia/default/[1]user at dest.com"/> > then place your call as usual > then in foo.js > // dumps the event to text/plain > env = request.dumpENV("text"); > // dumps the event to text/xml > xmlenv = request.dumpENV("xml"); > // makes an XML obj from the xml text > xinfo = new XML("" + xmlenv + ""); > // dump the plain text event data > consoleLog("info", env + "\n"); > // dump the xml event data > consoleLog("info", xmlenv + "\n"); > // Get a header from the event object > consoleLog("warning", "media ip was [" + > request.getHeader("local_media_ip") + "]\n"); > // Get the same header from the xml object > consoleLog("warning", "media ip was [" + > xinfo.event.headers.local_media_ip + "]\n"); > > On Thu, Jan 22, 2009 at 2:24 PM, Sias Mey <[2]sias at cpdata.co.za> wrote: > > Hi, > Im trying to originate calls from a conference and use javascript to > watch out for hangup events so I can use the data in the session to > flesh out some database info. However it seems that Im having some > strangeness. It might just be my code. So I include that. > I run FreeSwitch Version 1.0.trunk (11226) > Dialplan: > > expression="^confout-(10\d{2})$"> > > > > confout.js: > is attached > I use API calls to pull one user into a conference. Then I use more > api calls to do a conference dial to loopback/confout-1001 > This should run the js and then bridge extension 1001 into the same > conference. > (I have hardcoded the additional extension for testing). I dont know > if there is another way to get a conference dial to run a javascript > file for information logging, but I am open to enlightenment. > Oh im using conference dial because that provides clear audible > progress to the other conference memebers as to what is actually > happening with the new call. > Any help would be greatly apreciated, Thanks in advance. > Sias > _______________________________________________ > Freeswitch-users mailing list > [3]Freeswitch-users at lists.freeswitch.org > [4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[5]http://lists.freeswitch.org/mailman/options/freeswitc > h-users > [6]http://www.freeswitch.org > > -- > Anthony Minessale II > FreeSWITCH [7]http://www.freeswitch.org/ > ClueCon [8]http://www.cluecon.com/ > AIM: anthm > [9]MSN:anthony_minessale at hotmail.com > GTALK/JABBER/[10]PAYPAL:anthony.minessale at gmail.com > IRC: [11]irc.freenode.net #freeswitch > FreeSWITCH Developer Conference > [12]sip:888 at conference.freeswitch.org > [13]iax:guest at conference.freeswitch.org/888 > [14]googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > References > > 1. mailto:user at dest.com > 2. mailto:sias at cpdata.co.za > 3. mailto:Freeswitch-users at lists.freeswitch.org > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 5. http://lists.freeswitch.org/mailman/options/freeswitch-users > 6. http://www.freeswitch.org/ > 7. http://www.freeswitch.org/ > 8. http://www.cluecon.com/ > 9. mailto:MSN%3Aanthony_minessale at hotmail.com > 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 11. http://irc.freenode.net/ > 12. mailto:sip%3A888 at conference.freeswitch.org > 13. http://iax:guest at conference.freeswitch.org/888 > 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Fri Jan 23 02:00:05 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 23 Jan 2009 11:00:05 +0100 Subject: [Freeswitch-users] voicemail web interface In-Reply-To: <49797C58.7000607@ewetel.de> References: <49789B68.7060802@ewetel.de> <49789F53.6040206@freeswitch.org> <191c3a030901221115h6bd1505dx3014955d9ce9ac21@mail.gmail.com> <49797C58.7000607@ewetel.de> Message-ID: <49799525.2080203@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, Update to this: I found that slim.swf only plays mp3 files. wav files, which I have are ignored without an error message. renaming .wav to .mp3 doesn't help. slim.swf sends only then a http request, when the file to be loaded has a mp3 suffix. regards helmut Am 23.01.2009 09:14, schrieb Helmut Kuper: > Hello, > > well, was all my fault ... I was logged in with xml rpc user > (freeswitch/work). Both urls working the same way on my side: > entered in both pages simply extension number + vm-password. > > Both pages results in the same output. > > Very very impressive stuff :) > > Unfortunately I didn't get the nice flash players playing the messages. > When I click on play button player's title bar changes to "1. undefined" > and browser's status line shows: "Transferring data from ..." > > regards > Helmut _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl5lSQACgkQ4tZeNddg3dxYMgCgnUT8US/c3QzHwRABXMKfSAja K8wAoJSQKq/nLmX6/Jyp1PurACBpnnI/ =38C/ -----END PGP SIGNATURE----- From brian at freeswitch.org Fri Jan 23 02:03:07 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 23 Jan 2009 04:03:07 -0600 Subject: [Freeswitch-users] mod_radius_cdr questions and thoughts In-Reply-To: <49788820.1000208@kinetix.gr> References: <49788820.1000208@kinetix.gr> Message-ID: <50DC9878-0BAC-42EB-AB55-237AEC3134C5@freeswitch.org> Did you mean to send this twice? /b On Jan 22, 2009, at 8:52 AM, Apostolos Pantsiopoulos wrote: > I am trying to implement a radius based solution > using FS. I have seen that the mod_radius_cdr module > is actively maintained. so I have a few questions/remarks : > > 1) When I place a call and my radius server is down, the > call blocks forever instead of just radius_timeout * radius_retries > seconds (I have declared only one server). I would expect that > FS would stop trying to send an Acc Start packet after some > time and get on with the call. > > 2) I have also noticed that FS sends only 1 packet (I waited for a > minute) > instead of 3 (default in the config) since the first (and second) > attempt failed. > If my server was up (the port was responding) but it returned a req. > failed > answer would the above time-out be valid? > > 3) When I tried to load the dictionary.freeswitch to my freeradius > server, it complained : > > "Errors reading dictionary: dict_init: > /usr/local/etc/raddb/dictionary.freeswitch[33]: dict_addattr: > Duplicate > attribute name NAS-Port-Id" > > How can I overcome this? > > When I "instructed" my freeradius to only load the freesitch > dictionary > I got this : > > "Errors reading dictionary: dict_init: > /usr/local/etc/raddb/dictionary.freeswitch[257]: unknown option > "Freeswitch"" > > How can I use the dictionary with the freeradius? (or other radius > servers?) > > 4) The radius attributes included in the current requests are > a) hard-coded, b) limited in number. I think many of us would like to > use more attributes. Or even better define what to include (and what > to > put in them) using a > config file (the same maybe?) > > 5) Does the module send accounting packets only for the a-leg > of a call or for both legs? (Maybe that could be configurable too). > > If anyone is interested in the above questions/remarks please post > a reply. I would really like to know how many of the mailing list > users > are also interested in FS radius support and your opinions on the > matter. > > Regards, > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sias at cpdata.co.za Fri Jan 23 02:03:26 2009 From: sias at cpdata.co.za (Sias Mey) Date: Fri, 23 Jan 2009 12:03:26 +0200 Subject: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches In-Reply-To: <20090123095036.GA6568@cpdata.co.za> References: <20090122202406.GA21067@cpdata.co.za> <191c3a030901221425p379ad74j7f866eee4596a6a9@mail.gmail.com> <20090123095036.GA6568@cpdata.co.za> Message-ID: <20090123100326.GA6830@cpdata.co.za> Woot greater win. Thanks you so much for that pointer. although i did have to change the dialplan line to (space between jsapi and foo.js instead of ::) and im not sure if the api.js file actually made any difference.. but it did point me in the right direction. On Fri, Jan 23, 2009 at 11:50:36AM +0200, Sias Mey wrote: > Wait sory ignore my previous reply... > > I only just realized you were actually routing through the javascript > xml_rpc module. and I didnt actually have the api.js file in my scripts > dir. > > let me see what this does before you worry about it any more ;-) > On Thu, Jan 22, 2009 at 04:25:54PM -0600, Anthony Minessale wrote: > > Try this (update to svn trunk first) > > > data="{api_hangup_hook=jsapi::foo.js}sofia/default/[1]user at dest.com"/> > > then place your call as usual > > then in foo.js > > // dumps the event to text/plain > > env = request.dumpENV("text"); > > // dumps the event to text/xml > > xmlenv = request.dumpENV("xml"); > > // makes an XML obj from the xml text > > xinfo = new XML("" + xmlenv + ""); > > // dump the plain text event data > > consoleLog("info", env + "\n"); > > // dump the xml event data > > consoleLog("info", xmlenv + "\n"); > > // Get a header from the event object > > consoleLog("warning", "media ip was [" + > > request.getHeader("local_media_ip") + "]\n"); > > // Get the same header from the xml object > > consoleLog("warning", "media ip was [" + > > xinfo.event.headers.local_media_ip + "]\n"); > > > > On Thu, Jan 22, 2009 at 2:24 PM, Sias Mey <[2]sias at cpdata.co.za> wrote: > > > > Hi, > > Im trying to originate calls from a conference and use javascript to > > watch out for hangup events so I can use the data in the session to > > flesh out some database info. However it seems that Im having some > > strangeness. It might just be my code. So I include that. > > I run FreeSwitch Version 1.0.trunk (11226) > > Dialplan: > > > > > expression="^confout-(10\d{2})$"> > > > > > > > > confout.js: > > is attached > > I use API calls to pull one user into a conference. Then I use more > > api calls to do a conference dial to loopback/confout-1001 > > This should run the js and then bridge extension 1001 into the same > > conference. > > (I have hardcoded the additional extension for testing). I dont know > > if there is another way to get a conference dial to run a javascript > > file for information logging, but I am open to enlightenment. > > Oh im using conference dial because that provides clear audible > > progress to the other conference memebers as to what is actually > > happening with the new call. > > Any help would be greatly apreciated, Thanks in advance. > > Sias > > _______________________________________________ > > Freeswitch-users mailing list > > [3]Freeswitch-users at lists.freeswitch.org > > [4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:[5]http://lists.freeswitch.org/mailman/options/freeswitc > > h-users > > [6]http://www.freeswitch.org > > > > -- > > Anthony Minessale II > > FreeSWITCH [7]http://www.freeswitch.org/ > > ClueCon [8]http://www.cluecon.com/ > > AIM: anthm > > [9]MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/[10]PAYPAL:anthony.minessale at gmail.com > > IRC: [11]irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > > [12]sip:888 at conference.freeswitch.org > > [13]iax:guest at conference.freeswitch.org/888 > > [14]googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > References > > > > 1. mailto:user at dest.com > > 2. mailto:sias at cpdata.co.za > > 3. mailto:Freeswitch-users at lists.freeswitch.org > > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 5. http://lists.freeswitch.org/mailman/options/freeswitch-users > > 6. http://www.freeswitch.org/ > > 7. http://www.freeswitch.org/ > > 8. http://www.cluecon.com/ > > 9. mailto:MSN%3Aanthony_minessale at hotmail.com > > 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com > > 11. http://irc.freenode.net/ > > 12. mailto:sip%3A888 at conference.freeswitch.org > > 13. http://iax:guest at conference.freeswitch.org/888 > > 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Laurent.Fabre at kirranet.com Fri Jan 23 03:22:38 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Fri, 23 Jan 2009 12:22:38 +0100 Subject: [Freeswitch-users] voicemail web interface In-Reply-To: <49799525.2080203@ewetel.de> References: <49789B68.7060802@ewetel.de> <49789F53.6040206@freeswitch.org> <191c3a030901221115h6bd1505dx3014955d9ce9ac21@mail.gmail.com> <49797C58.7000607@ewetel.de> <49799525.2080203@ewetel.de> Message-ID: I was having the same issue :) I'm glad you found the workaround. -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61 laurent.fabre at kirranet.com -----Message d'origine----- De?: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Helmut Kuper Envoy??: vendredi 23 janvier 2009 11:00 ??: freeswitch-users at lists.freeswitch.org Objet?: Re: [Freeswitch-users] voicemail web interface -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, Update to this: I found that slim.swf only plays mp3 files. wav files, which I have are ignored without an error message. renaming .wav to .mp3 doesn't help. slim.swf sends only then a http request, when the file to be loaded has a mp3 suffix. regards helmut Am 23.01.2009 09:14, schrieb Helmut Kuper: > Hello, > > well, was all my fault ... I was logged in with xml rpc user > (freeswitch/work). Both urls working the same way on my side: > entered in both pages simply extension number + vm-password. > > Both pages results in the same output. > > Very very impressive stuff :) > > Unfortunately I didn't get the nice flash players playing the messages. > When I click on play button player's title bar changes to "1. undefined" > and browser's status line shows: "Transferring data from ..." > > regards > Helmut _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl5lSQACgkQ4tZeNddg3dxYMgCgnUT8US/c3QzHwRABXMKfSAja K8wAoJSQKq/nLmX6/Jyp1PurACBpnnI/ =38C/ -----END PGP SIGNATURE----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From regs at kinetix.gr Fri Jan 23 03:44:24 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Fri, 23 Jan 2009 13:44:24 +0200 Subject: [Freeswitch-users] mod_radius_cdr questions and thoughts In-Reply-To: <50DC9878-0BAC-42EB-AB55-237AEC3134C5@freeswitch.org> References: <49788820.1000208@kinetix.gr> <50DC9878-0BAC-42EB-AB55-237AEC3134C5@freeswitch.org> Message-ID: <4979AD98.70409@kinetix.gr> the first one would not appear in the mailing list 2 hours after I had sent it so I figured there was something wrong and I resent it Brian West wrote: > Did you mean to send this twice? > > /b > > On Jan 22, 2009, at 8:52 AM, Apostolos Pantsiopoulos wrote: > > >> I am trying to implement a radius based solution >> using FS. I have seen that the mod_radius_cdr module >> is actively maintained. so I have a few questions/remarks : >> >> 1) When I place a call and my radius server is down, the >> call blocks forever instead of just radius_timeout * radius_retries >> seconds (I have declared only one server). I would expect that >> FS would stop trying to send an Acc Start packet after some >> time and get on with the call. >> >> 2) I have also noticed that FS sends only 1 packet (I waited for a >> minute) >> instead of 3 (default in the config) since the first (and second) >> attempt failed. >> If my server was up (the port was responding) but it returned a req. >> failed >> answer would the above time-out be valid? >> >> 3) When I tried to load the dictionary.freeswitch to my freeradius >> server, it complained : >> >> "Errors reading dictionary: dict_init: >> /usr/local/etc/raddb/dictionary.freeswitch[33]: dict_addattr: >> Duplicate >> attribute name NAS-Port-Id" >> >> How can I overcome this? >> >> When I "instructed" my freeradius to only load the freesitch >> dictionary >> I got this : >> >> "Errors reading dictionary: dict_init: >> /usr/local/etc/raddb/dictionary.freeswitch[257]: unknown option >> "Freeswitch"" >> >> How can I use the dictionary with the freeradius? (or other radius >> servers?) >> >> 4) The radius attributes included in the current requests are >> a) hard-coded, b) limited in number. I think many of us would like to >> use more attributes. Or even better define what to include (and what >> to >> put in them) using a >> config file (the same maybe?) >> >> 5) Does the module send accounting packets only for the a-leg >> of a call or for both legs? (Maybe that could be configurable too). >> >> If anyone is interested in the above questions/remarks please post >> a reply. I would really like to know how many of the mailing list >> users >> are also interested in FS radius support and your opinions on the >> matter. >> >> Regards, >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/9e96bab2/attachment-0001.html From rehan at supertec.com Fri Jan 23 15:59:08 2009 From: rehan at supertec.com (Rehan Allah Wala) Date: Fri, 23 Jan 2009 16:59:08 -0700 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <200901231249.19207.krivushinme@rn-inform.tomsk.ru> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru>, <87f2f3b90901221619i71eb28e5y123784cc85a98672@mail.gmail.com>, <200901231249.19207.krivushinme@rn-inform.tomsk.ru> Message-ID: <4979F75C.12740.BA74FA2@rehan.supertec.com> Spacibah Balshoi When are you making g723 for the Russians? > I wrote it becouse in Russia * is very popular. And it have g729. > > I want to make popular FS in my country. We have not patent issues, but I more > like speex and celt - it's better in my opinion - 8 kHz is past century! It's > century of wb VoIP and not 8khz TDM! : ) > > Only for those who very need g729 in countries like Russia I made it public. I > dont tell you use it. In fact I tell to not use it - but if you cannot, you > can : ) > > -- > , > , > "-" ., > . . +7 913 865 78 66 > icq: 218 744 127 > xmpp: KrivushinME at jabber.ru > mail: KrivushinME at rn-inform.tomsk.ru > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Rehan Ahmed AllahWala Msn/Yahoo/GoogleTalk/Email: Rehan at Rehan.com http://www.supertec.com/ - Internet Telephony Solutions Http://www.DIDX.net - DID Number Market Place. Don't Remember Me ? Visit http://www.Rehan.com ~~~~~~~~~~~~~~~~~~~ "First they ignore you, then they laugh at you, then they fight you, then you win." By Gandhi. "Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi From steveu at coppice.org Fri Jan 23 04:13:51 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 23 Jan 2009 20:13:51 +0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <2d9149cd0901221655k798a417dmfae848cd6186085c@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org> <2d9149cd0901221655k798a417dmfae848cd6186085c@mail.gmail.com> Message-ID: <4979B47F.3010505@coppice.org> Kristian Kielhofner wrote: > On Thu, Jan 22, 2009 at 7:30 PM, Steve Underwood wrote: > >> Depends what you are after. Speex offers the quality of G.729 at around >> the same processing load. However, nobody seems to want to pay for the >> processing load of G.729. Almost everything uses G.729A. Half the >> processing load, but significantly poorer quality. >> > > Throughout most of this thread I've been using G.729 and G.729a > interchangeably. That's sloppy on my part. Most of the time I meant > G.729a because that is what most people (everyone?) uses. > > I was under the impression the quality difference wasn't that > significant. I can't say if I've ever knowingly used "true" G.729. > It certainly doesn't help that G.729a is allegedly completely > compatible with G.729, to the point where I don't think it's even > valid to use "G.729a" or annexa in an SDP... > There is no reason whatsoever to differentiate between G.729 and G.729A in the SDP as the bit streams are 100% compatible. Annex A data is just encoded and decoded in less optimal ways. Regards, Steve From anthony.minessale at gmail.com Fri Jan 23 06:00:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jan 2009 08:00:56 -0600 Subject: [Freeswitch-users] voicemail web interface In-Reply-To: References: <49789B68.7060802@ewetel.de> <49789F53.6040206@freeswitch.org> <191c3a030901221115h6bd1505dx3014955d9ce9ac21@mail.gmail.com> <49797C58.7000607@ewetel.de> <49799525.2080203@ewetel.de> Message-ID: <191c3a030901230600q6ceaa51bqb78a94bb899b8fb5@mail.gmail.com> in your voicemail prefs you have to change the format to mp3 to be able to play them on the web interface. in voicemail.conf.xml (you must have mod_shout loaded for this option) You also may have to change the recording rate to 11025 I know you do with gmail's player but try it without this setting first. On Fri, Jan 23, 2009 at 5:22 AM, Laurent Fabre wrote: > I was having the same issue :) I'm glad you found the workaround. > > -- Laurent FABRE > Directeur g?n?ral > 10, rue d'Aumale > 75009 Paris > Tel: +33.(0)1.42.81.28.20 > Mob: +33.(0)6.75.75.02.96 > Fax: +33.(0)1.70.24.74.61 > laurent.fabre at kirranet.com > > > -----Message d'origine----- > De : freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] De la part de Helmut Kuper > Envoy? : vendredi 23 janvier 2009 11:00 > ? : freeswitch-users at lists.freeswitch.org > Objet : Re: [Freeswitch-users] voicemail web interface > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > Update to this: I found that slim.swf only plays mp3 files. wav files, > which I have are ignored without an error message. renaming .wav to .mp3 > doesn't help. slim.swf sends only then a http request, when the file to > be loaded has a mp3 suffix. > > regards > helmut > > > Am 23.01.2009 09:14, schrieb Helmut Kuper: > > Hello, > > > > well, was all my fault ... I was logged in with xml rpc user > > (freeswitch/work). Both urls working the same way on my side: > > entered in both pages simply extension number + vm-password. > > > > Both pages results in the same output. > > > > Very very impressive stuff :) > > > > Unfortunately I didn't get the nice flash players playing the messages. > > When I click on play button player's title bar changes to "1. undefined" > > and browser's status line shows: "Transferring data from ..." > > > > regards > > Helmut > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkl5lSQACgkQ4tZeNddg3dxYMgCgnUT8US/c3QzHwRABXMKfSAja > K8wAoJSQKq/nLmX6/Jyp1PurACBpnnI/ > =38C/ > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/645829b8/attachment.html From anthony.minessale at gmail.com Fri Jan 23 06:10:24 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jan 2009 08:10:24 -0600 Subject: [Freeswitch-users] freeswitch memory leak issue In-Reply-To: References: <8C8A023D-28BD-4996-82C6-A465E3FE292E@freeswitch.org> Message-ID: <191c3a030901230610r7883bc3bw2f254a8e66696f64@mail.gmail.com> Make sure your mysql lib is the right one, you need the re entrant version look for _r We have seen people getting leaks in mysql in the past in several applications when not using the correct lib. There is not much else we can do to keep you from having a leak in Java and/or mysql client lib. If you can run it in valgrind and attribute it to FreeSWITCH actually holding the memory we can have a look. On Fri, Jan 23, 2009 at 1:17 PM, Simon Leck wrote: > Hi Brian, > > > > Thanks for your prompt reply. We are currently using FreeSwitch rev 1.02. > As for OS, we are using CentOS 5.2. > > > > We are using Java application to connect to MySQL database by JDBC APIs, > the issue we faced is that after many calls are made, the java application > will reflect the error message shown below. Furthermore the java application > will not release the memory even though, the application had disconnect from > the connected session > > > > Here are the errors message, > > "Caused by: java.lang.OutOfMemoryError: PermGen space" > > > > > > Thanks again Brian in advance for your kind assistance. > > > > Thanks > > Simon > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Thursday, January 22, 2009 5:20 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [?? Probable Spam] Re: [Freeswitch-users] freeswitch memory > leak issue > > > > What rev? What OS? What exactly are you doing? > > > > /b > > > > On Jan 23, 2009, at 11:15 AM, Simon Leck wrote: > > > > Hi All, > > > > I am experiencing memory leak issue in FreeSwitch? Anybody knows how this > can be resolved, please email me. > > > > Thanks in advance to everybody for your kind assistance. > > > > Thanks > > Simon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/e586c552/attachment-0001.html From anthony.minessale at gmail.com Fri Jan 23 06:13:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jan 2009 08:13:16 -0600 Subject: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches In-Reply-To: <20090123100326.GA6830@cpdata.co.za> References: <20090122202406.GA21067@cpdata.co.za> <191c3a030901221425p379ad74j7f866eee4596a6a9@mail.gmail.com> <20090123095036.GA6568@cpdata.co.za> <20090123100326.GA6830@cpdata.co.za> Message-ID: <191c3a030901230613t6335d8ic48e3ce22786803f@mail.gmail.com> That was the change i checked into trunk to allow app::arg as well as apparg that doesn't work for you? When i said update it was down to the minute i sent the email that the change was added. On Fri, Jan 23, 2009 at 4:03 AM, Sias Mey wrote: > Woot greater win. > > Thanks you so much for that pointer. > > although i did have to change the dialplan line to > data="{api_hangup_hook=jsapi foo.js}sofia/default/[1]user at dest.com"/> > (space between jsapi and foo.js instead of ::) > > and im not sure if the api.js file actually made any difference.. but it > did point me in the right direction. > > On Fri, Jan 23, 2009 at 11:50:36AM +0200, Sias Mey wrote: > > Wait sory ignore my previous reply... > > > > I only just realized you were actually routing through the javascript > > xml_rpc module. and I didnt actually have the api.js file in my scripts > > dir. > > > > let me see what this does before you worry about it any more ;-) > > On Thu, Jan 22, 2009 at 04:25:54PM -0600, Anthony Minessale wrote: > > > Try this (update to svn trunk first) > > > > > data="{api_hangup_hook=jsapi::foo.js}sofia/default/[1]user at dest.com > "/> > > > then place your call as usual > > > then in foo.js > > > // dumps the event to text/plain > > > env = request.dumpENV("text"); > > > // dumps the event to text/xml > > > xmlenv = request.dumpENV("xml"); > > > // makes an XML obj from the xml text > > > xinfo = new XML("" + xmlenv + ""); > > > // dump the plain text event data > > > consoleLog("info", env + "\n"); > > > // dump the xml event data > > > consoleLog("info", xmlenv + "\n"); > > > // Get a header from the event object > > > consoleLog("warning", "media ip was [" + > > > request.getHeader("local_media_ip") + "]\n"); > > > // Get the same header from the xml object > > > consoleLog("warning", "media ip was [" + > > > xinfo.event.headers.local_media_ip + "]\n"); > > > > > > On Thu, Jan 22, 2009 at 2:24 PM, Sias Mey <[2]sias at cpdata.co.za> > wrote: > > > > > > Hi, > > > Im trying to originate calls from a conference and use javascript > to > > > watch out for hangup events so I can use the data in the session > to > > > flesh out some database info. However it seems that Im having some > > > strangeness. It might just be my code. So I include that. > > > I run FreeSwitch Version 1.0.trunk (11226) > > > Dialplan: > > > > > > > > expression="^confout-(10\d{2})$"> > > > > > > > > > > > > confout.js: > > > is attached > > > I use API calls to pull one user into a conference. Then I use > more > > > api calls to do a conference dial to loopback/confout-1001 > > > This should run the js and then bridge extension 1001 into the > same > > > conference. > > > (I have hardcoded the additional extension for testing). I dont > know > > > if there is another way to get a conference dial to run a > javascript > > > file for information logging, but I am open to enlightenment. > > > Oh im using conference dial because that provides clear audible > > > progress to the other conference memebers as to what is actually > > > happening with the new call. > > > Any help would be greatly apreciated, Thanks in advance. > > > Sias > > > _______________________________________________ > > > Freeswitch-users mailing list > > > [3]Freeswitch-users at lists.freeswitch.org > > > [4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:[5] > http://lists.freeswitch.org/mailman/options/freeswitc > > > h-users > > > [6]http://www.freeswitch.org > > > > > > -- > > > Anthony Minessale II > > > FreeSWITCH [7]http://www.freeswitch.org/ > > > ClueCon [8]http://www.cluecon.com/ > > > AIM: anthm > > > [9]MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/[10]PAYPAL:anthony.minessale at gmail.com > > > IRC: [11]irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > > > [12]sip:888 at conference.freeswitch.org > > > [13]iax:guest at conference.freeswitch.org/888 > > > [14]googletalk:conf+888 at conference.freeswitch.org > > > pstn:213-799-1400 > > > > > > References > > > > > > 1. mailto:user at dest.com > > > 2. mailto:sias at cpdata.co.za > > > 3. mailto:Freeswitch-users at lists.freeswitch.org > > > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > 5. http://lists.freeswitch.org/mailman/options/freeswitch-users > > > 6. http://www.freeswitch.org/ > > > 7. http://www.freeswitch.org/ > > > 8. http://www.cluecon.com/ > > > 9. mailto:MSN%3Aanthony_minessale at hotmail.com > > > 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com > > > 11. http://irc.freenode.net/ > > > 12. mailto:sip%3A888 at conference.freeswitch.org > > > 13. http://iax:guest at conference.freeswitch.org/888 > > > 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/f402dd4b/attachment.html From helmut.kuper at ewetel.de Fri Jan 23 07:54:09 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 23 Jan 2009 16:54:09 +0100 Subject: [Freeswitch-users] voicemail web interface In-Reply-To: <191c3a030901230600q6ceaa51bqb78a94bb899b8fb5@mail.gmail.com> References: <49789B68.7060802@ewetel.de> <49789F53.6040206@freeswitch.org> <191c3a030901221115h6bd1505dx3014955d9ce9ac21@mail.gmail.com> <49797C58.7000607@ewetel.de> <49799525.2080203@ewetel.de> <191c3a030901230600q6ceaa51bqb78a94bb899b8fb5@mail.gmail.com> Message-ID: <4979E821.9000705@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, yes, I expected that. I enhanced the voicemail code, so it looks for the file suffix (mp3 or wav) and do a embed tag which calls my quicktime plugin. I know QT isn't available for linux ... :/ Disadvantage: every voicemail file is loaded when web-voicemail page is loaded ... But the hack does the job for me at least for now. regards Helmut Am 23.01.2009 15:00, schrieb Anthony Minessale: > in your voicemail prefs you have to change the format to mp3 to be able > to play them on the web interface. > > in voicemail.conf.xml > > > (you must have mod_shout loaded for this option) > > > You also may have to change the recording rate to 11025 > I know you do with gmail's player but try it without this setting first. > > > On Fri, Jan 23, 2009 at 5:22 AM, Laurent Fabre > > wrote: -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl56CEACgkQ4tZeNddg3dxs0gCfQeqjk3IX2CXl0M8rjBAYXEsr 0a4AnA0GNYd0CxUjZgW1660mOIwCccgJ =gMet -----END PGP SIGNATURE----- From odermann at googlemail.com Fri Jan 23 08:04:27 2009 From: odermann at googlemail.com (Dennis) Date: Fri, 23 Jan 2009 17:04:27 +0100 Subject: [Freeswitch-users] Conference and socket outbound In-Reply-To: <191c3a030811150928y59c85b94n7ffbb1417a281bf7@mail.gmail.com> References: <5e414ed0811150801g713024ffv2b22c7d8ccdda4f0@mail.gmail.com> <191c3a030811150928y59c85b94n7ffbb1417a281bf7@mail.gmail.com> Message-ID: <5e414ed0901230804i7b2ebfa3he883ef6670a7ce22@mail.gmail.com> is it possible to define a profile and its params for a conference dynamically over socket outbound? in the moment, if we want to have multiple profiles for different clients, we (have to) setup a profile in the conference.conf - otherwise we get an error in fs. because we have multipple fs-servers and multiple clients using conference, this is not very compfortable to setup. therefor it would be great, if we could call a conference profile name and set params dynamically over the socket, without having to edit the conference.conf thanks From Laurent.Fabre at kirranet.com Fri Jan 23 08:19:15 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Fri, 23 Jan 2009 17:19:15 +0100 Subject: [Freeswitch-users] Gateway Params Message-ID: Hi, Can I enclose the params of the directory gateway element in ? I mean I sure can but is the parser going to choke on it or not ? Thanks in advance, -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61<+33170247461> laurent.fabre at kirranet.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/bdced5bc/attachment-0001.html From brian at freeswitch.org Fri Jan 23 08:17:13 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 23 Jan 2009 10:17:13 -0600 Subject: [Freeswitch-users] Gateway Params In-Reply-To: References: Message-ID: <00DB9735-E007-42D3-AB82-F28C8041AA27@freeswitch.org> If its not done exactly like it says in the defaults ie it WILL not parse correctly. I think we might wanna change this to be inside a params tag... but we have to be backwards compatible too. /b On Jan 23, 2009, at 10:19 AM, Laurent Fabre wrote: > Hi, > > Can I enclose the params of the directory gateway element in > ? I mean I sure can but is the parser going to > choke on it or not ? > > Thanks in advance, > > -- Laurent FABRE > Directeur g?n?ral > 10, rue d'Aumale > 75009 Paris > Tel: +33.(0)1.42.81.28.20 > Mob: +33.(0)6.75.75.02.96 > Fax: +33.(0)1.70.24.74.61 > laurent.fabre at kirranet.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/b51bb72b/attachment.html From Laurent.Fabre at kirranet.com Fri Jan 23 08:39:08 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Fri, 23 Jan 2009 17:39:08 +0100 Subject: [Freeswitch-users] Gateway Params In-Reply-To: <00DB9735-E007-42D3-AB82-F28C8041AA27@freeswitch.org> References: <00DB9735-E007-42D3-AB82-F28C8041AA27@freeswitch.org> Message-ID: Thanks. That's what I was afraid of. Can I add variables inside the gateway or it's not supported either ? -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61<+33170247461> laurent.fabre at kirranet.com De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Brian West Envoy? : vendredi 23 janvier 2009 17:17 ? : freeswitch-users at lists.freeswitch.org Objet : Re: [Freeswitch-users] Gateway Params If its not done exactly like it says in the defaults ie it WILL not parse correctly. I think we might wanna change this to be inside a params tag... but we have to be backwards compatible too. /b On Jan 23, 2009, at 10:19 AM, Laurent Fabre wrote: Hi, Can I enclose the params of the directory gateway element in ? I mean I sure can but is the parser going to choke on it or not ? Thanks in advance, -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61<+33170247461> laurent.fabre at kirranet.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/cb806f01/attachment.html From brian at freeswitch.org Fri Jan 23 08:32:50 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 23 Jan 2009 10:32:50 -0600 Subject: [Freeswitch-users] Gateway Params In-Reply-To: References: <00DB9735-E007-42D3-AB82-F28C8041AA27@freeswitch.org> Message-ID: <887703A5-4DEB-4F9F-9CA9-1B0F75750128@freeswitch.org> Actually now that I think of it.. we might do both in gateways now. Let me dig into the code. /b On Jan 23, 2009, at 10:39 AM, Laurent Fabre wrote: > Thanks. That?s what I was afraid of. Can I add variables inside the > gateway or it?s not supported either ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/7b88d40f/attachment.html From cstomi.levlist at gmail.com Fri Jan 23 08:45:19 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Fri, 23 Jan 2009 17:45:19 +0100 Subject: [Freeswitch-users] record session in fifo Message-ID: <4979F41F.70605@gmail.com> Hello, we would like to distribute calls with fifo and record these sessions but we'd like to skip the recording while the caller is waiting. (we don't need to record the hold music, just the speech with the fifo consumer.) I tried but it doesn't work because the channel is answered immediately when the caller is pushed into the fifo. (I don't know if there exists any other channel flag that could be use here) I also tried fifo_record_template. but it records the session from the point of view of the consumer's session, and after the bridge the recording is stopped. we would like to record the whole session into a single file even after calltransfers moreover we'd like to use some kind of predcitive dialing which 1, originate a loopback channel via event socket 2, loopback-b channel is hunting the dialplan, wich decide routing, caller_id, the need for recordings and so forth, and bridge a sofia call 3. the record_session is running on the sofia channel with bridge_pre_execute magic vars 4 loopback-a channel is pushed into the fifo 5 a script get the fifo::info via event socket 6 originate a call to the consumer with the proper strategy with &fifo out application 7 sofia channel is bridged to the consumer 8 loopback channels die after transfers everything is recorded into one file. but the problem here is again the unwanted recording in the fifo while the caller is waiting Could you please advise me any solution, if there is? Thank you, Tamas From ivica.lists at googlemail.com Fri Jan 23 03:53:57 2009 From: ivica.lists at googlemail.com (Ivica Samija) Date: Fri, 23 Jan 2009 12:53:57 +0100 Subject: [Freeswitch-users] Few question regarding move from Asterisk to FS Message-ID: Hi all, our company have implemented two Asterisk servers to: - connect two company sites - transition to IP telephony - cut down TCO regarding telephony Our interconnection schema: --T1/E1 provider1--< > --T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 > ---SIP provider3---< > | SIP trunk | < Asterisk2 >--T1/E1/ trunk--< propriety PBX2 > On each site we have number of IP phones connected to Asterisk and analog phones connected to propriety PBX. Features implemented on Asterisk boxes are: - IVR - queue - conference - DISA - BLF - transfer calls All was more or less good until we had to implement call rating (we have to keep track cost made on each extension for statistic). Company policy is that implementation has to be in house. We hit brick wall because Asterisk have inaccurate CDRs (transfers, forwards,...) I am looking in FS for last few weeks and it seams to me that it can replace our Asterisk boxes, and more :). I am a little confused with XML config but it seams to me that it is worth of learning. For most of my question I have found answers in documentation wiki and on list archive, but I have just a few question still without answer. I am sorry if the answers are out there but I was to clumsy to find them. In that case some info or link woud be great. So here we go: 1) OpenZap is stable enough that it can be used in production? As you can see we depend on 4 zap trunks. From ivica.lists at googlemail.com Fri Jan 23 04:20:35 2009 From: ivica.lists at googlemail.com (Ivica Samija) Date: Fri, 23 Jan 2009 13:20:35 +0100 Subject: [Freeswitch-users] Few question regarding move from Asterisk to FS - resend Message-ID: Last message was incomplete, sorry for that, resending. Hi all, our company have implemented two Asterisk servers to: - connect two company sites - transition to IP telephony - cut down TCO regarding telephony Our interconnection schema: --T1/E1 provider1--< > --T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 > ---SIP provider3---< > | SIP trunk | < Asterisk2 >--T1/E1/ trunk--< propriety PBX2 > On each site we have number of IP phones connected to Asterisk and analog phones connected to propriety PBX. Features implemented on Asterisk boxes are: - IVR - queue - conference - DISA - BLF - transfer calls All was more or less good until we had to implement call rating (we have to keep track cost made on each extension for statistic). Company policy is that implementation has to be in house. We hit brick wall because Asterisk have inaccurate CDRs (transfers, forwards,...) I am looking in FS for last few weeks and it seams to me that it can replace our Asterisk boxes, and more :). I am a little confused with XML config but it seams to me that it is worth of learning. For most of my question I have found answers in documentation wiki and on list archive, but I have just a few question still without answer. I am sorry if the answers are out there but I was to clumsy to find them. In that case some info or link would be great. So here we go: 1) OpenZap is stable enough that it can be used in production? As you can see we depend on 4 zap trunks. We use OpenVox T1/E1 cards (D210P and D410P) with wct4xxp module. We use ccs, hdb3, crc4, loadzone=it in zaptel.conf We use switchtype=euroisdn, pridialplan=international, pridialplan=unknown, pri_cpe and pri_net signaling. Any suggestions are welcome. 2) Is it possible do bridge to Zap group instead of channel something like Dial(Zap/g4/${EXTEN:1},60,t)? 3) If I understood correct I can talk with SQLite (or other DB) only through Lua, Javascript,... there is nothing similar to Set(mobile=${DB(mob/${EXTEN:1})}) 4) mod_nibblebill, is it possible to get resulting cost per channel in some channel variable and than put value in custom cdr field Sorry on bad english, not a native speaker. Best regards, Ivica From mike at jerris.com Fri Jan 23 09:38:02 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 23 Jan 2009 12:38:02 -0500 Subject: [Freeswitch-users] Conference and socket outbound In-Reply-To: <5e414ed0901230804i7b2ebfa3he883ef6670a7ce22@mail.gmail.com> References: <5e414ed0811150801g713024ffv2b22c7d8ccdda4f0@mail.gmail.com> <191c3a030811150928y59c85b94n7ffbb1417a281bf7@mail.gmail.com> <5e414ed0901230804i7b2ebfa3he883ef6670a7ce22@mail.gmail.com> Message-ID: <95C5CB2E-6FA7-42A6-9832-3A266EF8801A@jerris.com> On Jan 23, 2009, at 11:04 AM, Dennis wrote: > is it possible to define a profile and its params for a conference > dynamically over socket outbound? > > in the moment, if we want to have multiple profiles for different > clients, we (have to) setup a profile in the conference.conf - > otherwise we get an error in fs. > because we have multipple fs-servers and multiple clients using > conference, this is not very compfortable to setup. > > therefor it would be great, if we could call a conference profile name > and set params dynamically over the socket, without having to edit the > conference.conf You can use something like mod_xml_curl to dynamically serve up the conference.conf xml Mike From mrene_lists at avgs.ca Fri Jan 23 09:39:07 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 23 Jan 2009 12:39:07 -0500 Subject: [Freeswitch-users] Few question regarding move from Asterisk to FS - resend In-Reply-To: References: Message-ID: 1) A lot of people use openzap in production environements 2) probably, even if openzap doesnt implement it (which I think it does), you can use call groups to achieve the same results 3) freeswitch has a db application/api that does the same. 4) it sets the "nibble_total_billed" channel variable, which you can use in your cdrs. Mathieu On Fri, Jan 23, 2009 at 7:20 AM, Ivica Samija wrote: > Last message was incomplete, sorry for that, resending. > > Hi all, > our company have implemented two Asterisk servers to: > - connect two company sites > - transition to IP telephony > - cut down TCO regarding telephony > > Our interconnection schema: > > --T1/E1 provider1--< > > --T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 > > ---SIP provider3---< > > | > SIP trunk > | > < Asterisk2 >--T1/E1/ trunk--< propriety PBX2 > > > On each site we have number of IP phones connected to Asterisk and > analog phones connected to propriety PBX. > Features implemented on Asterisk boxes are: > - IVR > - queue > - conference > - DISA > - BLF > - transfer calls > All was more or less good until we had to implement call rating (we > have to keep track cost made on each extension for statistic). > Company policy is that implementation has to be in house. We hit brick > wall because Asterisk have inaccurate CDRs (transfers, forwards,...) > I am looking in FS for last few weeks and it seams to me that it can > replace our Asterisk boxes, and more :). > I am a little confused with XML config but it seams to me that it is > worth of learning. > For most of my question I have found answers in documentation wiki and > on list archive, but I have just a few question still without answer. > I am sorry if the answers are out there but I was to clumsy to find > them. In that case some info or link would be great. > So here we go: > 1) OpenZap is stable enough that it can be used in production? > As you can see we depend on 4 zap trunks. > We use OpenVox T1/E1 cards (D210P and D410P) with wct4xxp module. > We use ccs, hdb3, crc4, loadzone=it in zaptel.conf > We use switchtype=euroisdn, pridialplan=international, pridialplan=unknown, > pri_cpe and pri_net signaling. > Any suggestions are welcome. > > 2) Is it possible do bridge to Zap group instead of channel > something like Dial(Zap/g4/${EXTEN:1},60,t)? > > 3) If I understood correct I can talk with SQLite (or other DB) only > through Lua, Javascript,... > there is nothing similar to Set(mobile=${DB(mob/${EXTEN:1})}) > > 4) mod_nibblebill, is it possible to get resulting cost per channel in > some channel variable > and than put value in custom cdr field > > Sorry on bad english, not a native speaker. > Best regards, > Ivica > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/87f9e18f/attachment.html From msc at freeswitch.org Fri Jan 23 09:51:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Jan 2009 09:51:41 -0800 Subject: [Freeswitch-users] record session in fifo In-Reply-To: <4979F41F.70605@gmail.com> References: <4979F41F.70605@gmail.com> Message-ID: <87f2f3b90901230951i72f80803kbd73de277c4da8f6@mail.gmail.com> Tamas, These are very specific requirements. FreeSWITCH certainly has the tools necessary to do it all but it requires some knowledge and skill on your part. I think you are right to be looking at the event socket for this. You need some sort of 3PCC - 3rd Party Call Control - which is most likely going to be a set of scripts that talk to the event socket, each one doing a specific task, i.e. turning on recording at the correct point in the call. This is a big-time serious project. If you don't have much experience in programming in an environment like this then you would be better off hiring someone to do the work. I doubt that you will be able to build something like this simply by asking around. -MC On Fri, Jan 23, 2009 at 8:45 AM, Tamas Cseke wrote: > Hello, > > we would like to distribute calls with fifo and record these sessions > but we'd like to skip the recording while the caller is waiting. > (we don't need to record the hold music, just the speech with the fifo > consumer.) > > I tried > > > > > but it doesn't work because the channel is answered immediately when the > caller is pushed into the fifo. > (I don't know if there exists any other channel flag that could be use here) > > I also tried fifo_record_template. > but it records the session from the point of view of the consumer's > session, and after the bridge the recording is stopped. > we would like to record the whole session into a single file even after > calltransfers > > moreover we'd like to use some kind of predcitive dialing > which > 1, originate a loopback channel via event socket > 2, loopback-b channel is hunting the dialplan, wich decide routing, > caller_id, the need for recordings and so forth, and bridge a sofia call > 3. the record_session is running on the sofia channel with > bridge_pre_execute magic vars > 4 loopback-a channel is pushed into the fifo > 5 a script get the fifo::info via event socket > 6 originate a call to the consumer with the proper strategy with &fifo > out application > 7 sofia channel is bridged to the consumer > 8 loopback channels die > > after transfers everything is recorded into one file. > but the problem here is again the unwanted recording in the fifo while > the caller is waiting > > Could you please advise me any solution, if there is? > > > Thank you, > Tamas > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Fri Jan 23 10:45:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jan 2009 12:45:00 -0600 Subject: [Freeswitch-users] record session in fifo In-Reply-To: <4979F41F.70605@gmail.com> References: <4979F41F.70605@gmail.com> Message-ID: <191c3a030901231045m784f0cd7s6994b763d02b3e26@mail.gmail.com> please test latest trunk. Patch added to pause media bugs while not in a bridge which should pause recordings and cut out the moh. On Fri, Jan 23, 2009 at 10:45 AM, Tamas Cseke wrote: > Hello, > > we would like to distribute calls with fifo and record these sessions > but we'd like to skip the recording while the caller is waiting. > (we don't need to record the hold music, just the speech with the fifo > consumer.) > > I tried > > > > > but it doesn't work because the channel is answered immediately when the > caller is pushed into the fifo. > (I don't know if there exists any other channel flag that could be use > here) > > I also tried fifo_record_template. > but it records the session from the point of view of the consumer's > session, and after the bridge the recording is stopped. > we would like to record the whole session into a single file even after > calltransfers > > moreover we'd like to use some kind of predcitive dialing > which > 1, originate a loopback channel via event socket > 2, loopback-b channel is hunting the dialplan, wich decide routing, > caller_id, the need for recordings and so forth, and bridge a sofia call > 3. the record_session is running on the sofia channel with > bridge_pre_execute magic vars > 4 loopback-a channel is pushed into the fifo > 5 a script get the fifo::info via event socket > 6 originate a call to the consumer with the proper strategy with &fifo > out application > 7 sofia channel is bridged to the consumer > 8 loopback channels die > > after transfers everything is recorded into one file. > but the problem here is again the unwanted recording in the fifo while > the caller is waiting > > Could you please advise me any solution, if there is? > > > Thank you, > Tamas > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/a9df5680/attachment.html From msc at freeswitch.org Fri Jan 23 11:57:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Jan 2009 11:57:02 -0800 Subject: [Freeswitch-users] Few question regarding move from Asterisk to FS In-Reply-To: References: Message-ID: <87f2f3b90901231157s3ed62304k1042cdfafc88028@mail.gmail.com> On Fri, Jan 23, 2009 at 3:53 AM, Ivica Samija wrote: > Hi all, > our company have implemented two Asterisk servers to: > - connect two company sites > - transition to IP telephony > - cut down TCO regarding telephony > > Our interconnection schema: > > --T1/E1 provider1--< > > --T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 > > ---SIP provider3---< > > | > SIP trunk > | > < Asterisk2 >--T1/E1/ trunk--< propriety PBX2 > > > On each site we have number of IP phones connected to Asterisk and > analog phones connected to propriety PBX. > Features implemented on Asterisk boxes are: > - IVR > - queue > - conference > - DISA > - BLF > - transfer calls > All was more or less good until we had to implement call rating (we > have to keep track cost made on each extension for statistic). > Company policy is that implementation has to be in house. We hit brick > wall because Asterisk have inaccurate CDRs (transfers, forwards,...) > I am looking in FS for last few weeks and it seams to me that it can > replace our Asterisk boxes, and more :). > I am a little confused with XML config but it seams to me that it is > worth of learning. > For most of my question I have found answers in documentation wiki and > on list archive, but I have just a few question still without answer. > I am sorry if the answers are out there but I was to clumsy to find > them. In that case some info or link woud be great. > So here we go: > 1) OpenZap is stable enough that it can be used in production? As you > can see we depend on 4 zap trunks. I would highly recommend doing a test with your provider and your PBX to see if there are any interop issues. Also, OpenZAP has a PRI issue where channels don't go back to DOWN (i.e. idle/on-hook) properly. It's a known issue that is being worked on but it's being done in a volunteer's spare time so there's no ETA on that. I would recommend that you hop on IRC and join both #freeswitch and #openzap channels. User Cypromis is putting some openzap stuff into production next week so you might want to pick his brain on the issues that he sees. -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Jan 23 12:05:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Jan 2009 12:05:31 -0800 Subject: [Freeswitch-users] Few question regarding move from Asterisk to FS - resend In-Reply-To: References: Message-ID: <87f2f3b90901231205j404acea6u1887fc3463406af6@mail.gmail.com> FYI, sorry, I responded to the other email first! Per Mathieu's replies, yes FreeSWITCH can do all of those things that you mentioned. The key for you will be to unlearn "the Asterisk way" because much of the way Asterisk does things is a result of working around inherent limitations in the system. FreeSWITCH is designed to avoid those limitations wherever possible, so in many cases there are ways to accomplish what you are doing with Asterisk without copying what Asterisk does. My advice to you is to get some test boxes to play with. Try doing various things with FreeSWITCH to see if it does what you need it to do. And go slowly. FreeSWITCH does all sorts of things so give yourself some time to learn it all. -MC On Fri, Jan 23, 2009 at 9:39 AM, Mathieu Rene wrote: > 1) A lot of people use openzap in production environements > 2) probably, even if openzap doesnt implement it (which I think it does), > you can use call groups to achieve the same results > 3) freeswitch has a db application/api that does the same. > 4) it sets the "nibble_total_billed" channel variable, which you can use in > your cdrs. > > > Mathieu > > > On Fri, Jan 23, 2009 at 7:20 AM, Ivica Samija > wrote: >> >> Last message was incomplete, sorry for that, resending. >> >> Hi all, >> our company have implemented two Asterisk servers to: >> - connect two company sites >> - transition to IP telephony >> - cut down TCO regarding telephony >> >> Our interconnection schema: >> >> --T1/E1 provider1--< > >> --T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 > >> ---SIP provider3---< > >> | >> SIP trunk >> | >> < Asterisk2 >--T1/E1/ trunk--< propriety PBX2 > >> >> On each site we have number of IP phones connected to Asterisk and >> analog phones connected to propriety PBX. >> Features implemented on Asterisk boxes are: >> - IVR >> - queue >> - conference >> - DISA >> - BLF >> - transfer calls >> All was more or less good until we had to implement call rating (we >> have to keep track cost made on each extension for statistic). >> Company policy is that implementation has to be in house. We hit brick >> wall because Asterisk have inaccurate CDRs (transfers, forwards,...) >> I am looking in FS for last few weeks and it seams to me that it can >> replace our Asterisk boxes, and more :). >> I am a little confused with XML config but it seams to me that it is >> worth of learning. >> For most of my question I have found answers in documentation wiki and >> on list archive, but I have just a few question still without answer. >> I am sorry if the answers are out there but I was to clumsy to find >> them. In that case some info or link would be great. >> So here we go: >> 1) OpenZap is stable enough that it can be used in production? >> As you can see we depend on 4 zap trunks. >> We use OpenVox T1/E1 cards (D210P and D410P) with wct4xxp module. >> We use ccs, hdb3, crc4, loadzone=it in zaptel.conf >> We use switchtype=euroisdn, pridialplan=international, >> pridialplan=unknown, >> pri_cpe and pri_net signaling. >> Any suggestions are welcome. >> >> 2) Is it possible do bridge to Zap group instead of channel >> something like Dial(Zap/g4/${EXTEN:1},60,t)? >> >> 3) If I understood correct I can talk with SQLite (or other DB) only >> through Lua, Javascript,... >> there is nothing similar to Set(mobile=${DB(mob/${EXTEN:1})}) >> >> 4) mod_nibblebill, is it possible to get resulting cost per channel in >> some channel variable >> and than put value in custom cdr field >> >> Sorry on bad english, not a native speaker. >> Best regards, >> Ivica >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From telles-listas at devel-it.com.br Fri Jan 23 12:16:56 2009 From: telles-listas at devel-it.com.br (Rodrigo P. Telles) Date: Fri, 23 Jan 2009 18:16:56 -0200 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <4979510C.90303@3c.co.uk> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <7C9FDCAF-E480-404F-9BA6-25BBA168A19F@freeswitch.org> <2d9149cd0901221331v23108d3el845d56db200d5bf@mail.gmail.com> <87f2f3b90901221340p3e639e78i689394c0dadb5425@mail.gmail.com> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org> <4979510C.90303@3c.co.uk> Message-ID: <497A25B8.3010504@devel-it.com.br> Hi Dave, Down here in Brazil, the bandwidth costs is very high (around U$ 400.00/Mb) so it should be valid only for a "non" third world country. G729 and G723.1 is almost a law here, if you don't play at least with G729 your ITSP is out of mark share! My 2 cents from a third world country. Regards, Rodrigo Telles Em 23-01-2009 03:09, David Knell escreveu: > Steve Underwood wrote: >> Depends what you are after. Speex offers the quality of G.729 at around >> the same processing load. However, nobody seems to want to pay for the >> processing load of G.729. Almost everything uses G.729A. Half the >> processing load, but significantly poorer quality. >> >> VoIP is mostly a race to the bottom, and people wonder why it makes no >> money for provides. :-\ >> > And, at the wholesale level, it makes no sense whatsoever to compress calls > any more: bandwidth is so cheap (and has been for a while) that the loss in > call quality - especially from tandem compressions - and the increased > processing requirements and other bits of expense do not stack up. Case in > point: we moved a route from G.711 to G.729, and saw the ACD drop from > over 10 to under 7 minutes. It was a route to mobiles, so the audio was > being > recompressed with the GSM codec on its way to the handsets. Economically, > had we carried on using G.729, we'd have lost about 30% of our margin on > that route. > > --Dave From msc at freeswitch.org Fri Jan 23 12:29:04 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Jan 2009 12:29:04 -0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <497A25B8.3010504@devel-it.com.br> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org> <4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br> Message-ID: <87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> On Fri, Jan 23, 2009 at 12:16 PM, Rodrigo P. Telles wrote: > Hi Dave, > > Down here in Brazil, the bandwidth costs is very high (around U$ 400.00/Mb) so it should be valid only for a "non" third > world country. > G729 and G723.1 is almost a law here, if you don't play at least with G729 your ITSP is out of mark share! > > My 2 cents from a third world country. What is the patent and licensing situation in Brazil? Those are also factors. $10/port might be cheap in the US but in Brazil it could be much more? (I'm asking...) -MC > > Regards, > Rodrigo Telles > > Em 23-01-2009 03:09, David Knell escreveu: >> Steve Underwood wrote: >>> Depends what you are after. Speex offers the quality of G.729 at around >>> the same processing load. However, nobody seems to want to pay for the >>> processing load of G.729. Almost everything uses G.729A. Half the >>> processing load, but significantly poorer quality. >>> >>> VoIP is mostly a race to the bottom, and people wonder why it makes no >>> money for provides. :-\ >>> >> And, at the wholesale level, it makes no sense whatsoever to compress calls >> any more: bandwidth is so cheap (and has been for a while) that the loss in >> call quality - especially from tandem compressions - and the increased >> processing requirements and other bits of expense do not stack up. Case in >> point: we moved a route from G.711 to G.729, and saw the ACD drop from >> over 10 to under 7 minutes. It was a route to mobiles, so the audio was >> being >> recompressed with the GSM codec on its way to the handsets. Economically, >> had we carried on using G.729, we'd have lost about 30% of our margin on >> that route. >> >> --Dave > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Fri Jan 23 12:37:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jan 2009 14:37:19 -0600 Subject: [Freeswitch-users] Gateway Params In-Reply-To: <887703A5-4DEB-4F9F-9CA9-1B0F75750128@freeswitch.org> References: <00DB9735-E007-42D3-AB82-F28C8041AA27@freeswitch.org> <887703A5-4DEB-4F9F-9CA9-1B0F75750128@freeswitch.org> Message-ID: <191c3a030901231237u74635fbdwd063f12140bf2deb@mail.gmail.com> now all of the above is true in latest trunk On Fri, Jan 23, 2009 at 10:32 AM, Brian West wrote: > Actually now that I think of it.. we might do both in gateways now. Let me > dig into the code. > /b > > On Jan 23, 2009, at 10:39 AM, Laurent Fabre wrote: > > Thanks. That's what I was afraid of. Can I add variables inside the gateway > or it's not supported either ? > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/bed9f0ad/attachment.html From sicfslist at gmail.com Fri Jan 23 14:04:40 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 23 Jan 2009 16:04:40 -0600 Subject: [Freeswitch-users] Auto dialing ... Message-ID: <35b355e90901231404lfb937b5x1152b4703ce88130@mail.gmail.com> OK ... Here goes another I'm doing this with AST ... but I want to move it to FS. Searched via google site:lists.freeswitch.org auto dialer and others ... nothing useful. Today I have a platform for auto dialing with AST (centrally managed ... about 10 machines) and we do this: -- Remote machines query central DB for numbers to call based on certain configs -- Use AMI to generate the call -- If call gets answered, extension info queried via rta (central db again) The nice thing about all of this is it's relatively easy to manage (through one central web interface we built) and it works ... the bad part is reporting ... as anyone knows on this list that has used AST for auto dialing in this way (via .call or AMI) every call looks like it fails instead of showing a real cause code. So ... conceptually I'm trying to accomplish the same thing ... Today we use FS a lot for termination of VoIP traffic ... all done via XML_CURL ... which is awesome! Would like to do something like: -- originate request -- on answer XML_CURL posts info -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/5a3aca0a/attachment.html From testeador01 at gmail.com Fri Jan 23 14:14:46 2009 From: testeador01 at gmail.com (Milena) Date: Fri, 23 Jan 2009 17:14:46 -0500 Subject: [Freeswitch-users] My fs_cli stopped working, can you help me please? Message-ID: I use the stable version of fs 1.0.2 testing out different options of configuration; Last thing I was doing was trying to fix some issues with dingaling-google talk cause i hear no audio at all from an external ip, and i still didnt get it to work; so i tried deleting the sofia_* files on the folder db/ as suggested on another thread about nat related issues (oops?), and then restarting fs. After this, trying to execute bin/fs_cli doesn't work anymore; it tries to connect cause i get the output on the box: 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1791 listener_run() Connection Open from 127.0.0.1:33055 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1914 listener_run() Session complete, waiting for children 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1938 listener_run() Connection Closed but on the console where im trying to run the cli nothing happens, I tried rebooting the machine; i tried with freeswitch make current, it didn't fix it; i tried all the way from configure to make and make install and it didn't fix my problem either. What should i run or change on the configurations to fix this? Thank you very much. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/88ba0e3e/attachment.html From sicfslist at gmail.com Fri Jan 23 14:15:10 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 23 Jan 2009 16:15:10 -0600 Subject: [Freeswitch-users] auto dialing question ... Message-ID: <35b355e90901231415q782b5ba9mba2bfc7fe6345fd3@mail.gmail.com> Sorry for the double post ... actually hit send too early ... OK ... Here goes another I'm doing this with AST ... but I want to move it to FS. Searched via google site:lists.freeswitch.org auto dialer and others ... nothing useful. Today I have a platform for auto dialing with AST (centrally managed ... about 10 machines) and we do this: -- Remote machines query central DB for numbers to call based on certain configs -- Use AMI to generate the call -- If call gets answered, extension info queried via rta (central db again) The nice thing about all of this is it's relatively easy to manage (through one central web interface we built) and it works ... the bad part is reporting ... So ... conceptually I'm trying to accomplish the same thing ... Today we use FS a lot for termination of VoIP traffic ... all done via XML_CURL ... which is awesome (not to xml cdr ... and the "proxying" of media) ... Would like to do something like: -- originate request (looks simple enough) -- on answer XML_CURL posts info But for the life of me I can't figure out how to translate this into the xml response ... [campaign] exten => 100,1,ANSWER() exten => 100,n,WAIT(2) exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile) exten => 100,n,WAITEXTEN(10) exten => 100,n,HANGUP() exten => 1,1,PLAYBACK(goodbye) .... and so on ... I've looked at the ivr.conf stuff but it's all static and all of this has to be manageable via a web interface .... meaning dumping into a DB and returning an XML response seems reasonable ... but trying to stick or modify static text files from the web interface is too much text parsing and bad things will happen ... Any thoughts or pointing me in the right direction would be appreciated. Shelby -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/717257a6/attachment-0001.html From egghunt at gmail.com Fri Jan 23 14:33:46 2009 From: egghunt at gmail.com (Arnaldo de Moraes Pereira) Date: Fri, 23 Jan 2009 20:33:46 -0200 Subject: [Freeswitch-users] Auto dialing ... In-Reply-To: <35b355e90901231404lfb937b5x1152b4703ce88130@mail.gmail.com> References: <35b355e90901231404lfb937b5x1152b4703ce88130@mail.gmail.com> Message-ID: On Fri, Jan 23, 2009 at 8:04 PM, Shelby Ramsey wrote: > OK ... Here goes another I'm doing this with AST ... but I want to move it > to FS. Searched via google site:lists.freeswitch.org auto dialer and > others ... nothing useful. > Today I have a platform for auto dialing with AST (centrally managed ... > about 10 machines) and we do this: > -- Remote machines query central DB for numbers to call based on certain > configs > -- Use AMI to generate the call > -- If call gets answered, extension info queried via rta (central db > again) > > The nice thing about all of this is it's relatively easy to manage (through > one central web interface we built) and it works ... the bad part is > reporting ... as anyone knows on this list that has used AST for auto > dialing in this way (via .call or AMI) every call looks like it fails > instead of showing a real cause code. > > So ... conceptually I'm trying to accomplish the same thing ... > > Today we use FS a lot for termination of VoIP traffic ... all done via > XML_CURL ... which is awesome! > > Would like to do something like: > -- originate request > -- on answer XML_CURL posts info > Auto dialing is one of the many areas where freeswitch is much superior than asterisk. You can accomplish what you need in some ways, one would be to listen for a CHANNEL_ANSWER event on event_socket interface and, then, take whatever needed action. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 http://lustyscripps.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/b8576570/attachment.html From brian at freeswitch.org Fri Jan 23 14:38:23 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 23 Jan 2009 16:38:23 -0600 Subject: [Freeswitch-users] Auto dialing ... In-Reply-To: References: <35b355e90901231404lfb937b5x1152b4703ce88130@mail.gmail.com> Message-ID: http://wiki.freeswitch.org/wiki/Mod_commands#originate /b On Jan 23, 2009, at 4:33 PM, Arnaldo de Moraes Pereira wrote: > On Fri, Jan 23, 2009 at 8:04 PM, Shelby Ramsey > wrote: > OK ... Here goes another I'm doing this with AST ... but I want to > move it to FS. Searched via google site:lists.freeswitch.org auto > dialer and others ... nothing useful. > > Today I have a platform for auto dialing with AST (centrally > managed ... about 10 machines) and we do this: > -- Remote machines query central DB for numbers to call based on > certain configs > -- Use AMI to generate the call > -- If call gets answered, extension info queried via rta (central > db again) > > The nice thing about all of this is it's relatively easy to manage > (through one central web interface we built) and it works ... the > bad part is reporting ... as anyone knows on this list that has used > AST for auto dialing in this way (via .call or AMI) every call looks > like it fails instead of showing a real cause code. > > So ... conceptually I'm trying to accomplish the same thing ... > > Today we use FS a lot for termination of VoIP traffic ... all done > via XML_CURL ... which is awesome! > > Would like to do something like: > -- originate request > -- on answer XML_CURL posts info > > Auto dialing is one of the many areas where freeswitch is much > superior than asterisk. You can accomplish what you need in some > ways, one would be to listen for a CHANNEL_ANSWER event on > event_socket interface and, then, take whatever needed action. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 > http://lustyscripps.wordpress.com > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/cc4f576a/attachment.html From msc at freeswitch.org Fri Jan 23 14:39:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Jan 2009 14:39:34 -0800 Subject: [Freeswitch-users] auto dialing question ... In-Reply-To: <35b355e90901231415q782b5ba9mba2bfc7fe6345fd3@mail.gmail.com> References: <35b355e90901231415q782b5ba9mba2bfc7fe6345fd3@mail.gmail.com> Message-ID: <87f2f3b90901231439u553031f4u29c1f27670fe5c78@mail.gmail.com> On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey wrote: > Sorry for the double post ... actually hit send too early ... > OK ... Here goes another I'm doing this with AST ... but I want to move it > to FS. Searched via google site:lists.freeswitch.org auto dialer and others > ... nothing useful. > Today I have a platform for auto dialing with AST (centrally managed ... > about 10 machines) and we do this: > -- Remote machines query central DB for numbers to call based on certain > configs > -- Use AMI to generate the call > -- If call gets answered, extension info queried via rta (central db > again) > The nice thing about all of this is it's relatively easy to manage (through > one central web interface we built) and it works ... the bad part is > reporting ... > So ... conceptually I'm trying to accomplish the same thing ... > Today we use FS a lot for termination of VoIP traffic ... all done via > XML_CURL ... which is awesome (not to xml cdr ... and the "proxying" of > media) ... > Would like to do something like: > -- originate request (looks simple enough) > -- on answer XML_CURL posts info Several choices, depending upon how much you want it handled inside the dialplan vs. handled in the scripting language. For the sake of testing you could do something like this: Then have: This would have any answered call go to the "ivr-answer" extension while unanswered calls could stay in the ivr-start extension to get properly handled. (Busy, no answer, invalid/SIT, etc.) You could then have the "ivr-answer" extension do whatever is appropriate, like listen for digits, play announcement, beg for money, etc. :) -MC > But for the life of me I can't figure out how to translate this into the xml > response ... > [campaign] > exten => 100,1,ANSWER() > exten => 100,n,WAIT(2) > exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile) > exten => 100,n,WAITEXTEN(10) > exten => 100,n,HANGUP() > exten => 1,1,PLAYBACK(goodbye) > .... and so on ... > I've looked at the ivr.conf stuff but it's all static and all of this has to > be manageable via a web interface .... meaning dumping into a DB and > returning an XML response seems reasonable ... but trying to stick or modify > static text files from the web interface is too much text parsing and bad > things will happen ... > Any thoughts or pointing me in the right direction would be appreciated. > Shelby > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Fri Jan 23 14:40:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jan 2009 16:40:50 -0600 Subject: [Freeswitch-users] Auto dialing ... In-Reply-To: <35b355e90901231404lfb937b5x1152b4703ce88130@mail.gmail.com> References: <35b355e90901231404lfb937b5x1152b4703ce88130@mail.gmail.com> Message-ID: <191c3a030901231440i1532cd04ub1cb61f44d756c85@mail.gmail.com> Does AST mean Asterisk Open Source PBX ? If so, then yes I am familiar with it's archetechure as I am a former developer from that project. You have 3 choices with FreeSWITCH 1) You can open a dedicated connection to mod_event_socket or XMLRPC per call and issue the originate command from there: This will block until you know for sure the outcome of the attempt. If it's success it will give you the uuid if not it gives you the cause code. 2) You can use a single mod_event_socket or XMLRPC connection to send all calls but use the bgapi mechanism which will do the same as above only asynchronously, The command will return immediately and the result will be fired as an event that you can pick up on the same or different event_socket connection or other event consumer such as a custom C,perl,lua etc module. 3) You can use mod_xml_cdr to generate detailed 1-file-per-leg call files that will tell you when where and why the calls failed or did not fail. On Fri, Jan 23, 2009 at 4:04 PM, Shelby Ramsey wrote: > OK ... Here goes another I'm doing this with AST ... but I want to move it > to FS. Searched via google site:lists.freeswitch.org auto dialer and > others ... nothing useful. > Today I have a platform for auto dialing with AST (centrally managed ... > about 10 machines) and we do this: > -- Remote machines query central DB for numbers to call based on certain > configs > -- Use AMI to generate the call > -- If call gets answered, extension info queried via rta (central db > again) > > The nice thing about all of this is it's relatively easy to manage (through > one central web interface we built) and it works ... the bad part is > reporting ... as anyone knows on this list that has used AST for auto > dialing in this way (via .call or AMI) every call looks like it fails > instead of showing a real cause code. > > So ... conceptually I'm trying to accomplish the same thing ... > > Today we use FS a lot for termination of VoIP traffic ... all done via > XML_CURL ... which is awesome! > > Would like to do something like: > -- originate request > -- on answer XML_CURL posts info > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/290f76a7/attachment.html From msc at freeswitch.org Fri Jan 23 14:49:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Jan 2009 14:49:42 -0800 Subject: [Freeswitch-users] My fs_cli stopped working, can you help me please? In-Reply-To: References: Message-ID: <87f2f3b90901231449x7e0e7785pd87455f36e670d54@mail.gmail.com> just for kicks, can you try a raw telnet session, just to make sure the the event socket is working properly? telnet localhost 8021 auth ClueCon (press enter twice) If you get something like this: h-3.2# telnet localhost 8021 Trying ::1... telnet: connect to address ::1: Connection refused Trying fe80::1... telnet: connect to address fe80::1: Connection refused Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Content-Type: auth/request auth ClueCon Content-Type: command/reply Reply-Text: +OK accepted Then the event socket stuff is working. To exit out, type "exit" then enter twice. Let us know what happens. -MC On Fri, Jan 23, 2009 at 2:14 PM, Milena wrote: > > I use the stable version of fs 1.0.2 testing out different options of > configuration; > Last thing I was doing was trying to fix some issues with dingaling-google > talk cause i hear no audio at all from an external ip, and i still didnt get > it to work; > > so i tried deleting the sofia_* files on the folder db/ as suggested on > another thread about nat related issues (oops?), and then restarting fs. > After this, trying to execute bin/fs_cli doesn't work anymore; it tries to > connect cause i get the output on the box: > 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1791 listener_run() > Connection Open from 127.0.0.1:33055 > 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1914 listener_run() Session > complete, waiting for children > 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1938 listener_run() > Connection Closed > > but on the console where im trying to run the cli nothing happens, > I tried rebooting the machine; i tried with freeswitch make current, it > didn't fix it; i tried all the way from configure to make and make install > and it didn't fix my problem either. > > What should i run or change on the configurations to fix this? > > Thank you very much. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Fri Jan 23 14:55:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jan 2009 16:55:03 -0600 Subject: [Freeswitch-users] auto dialing question ... In-Reply-To: <87f2f3b90901231439u553031f4u29c1f27670fe5c78@mail.gmail.com> References: <35b355e90901231415q782b5ba9mba2bfc7fe6345fd3@mail.gmail.com> <87f2f3b90901231439u553031f4u29c1f27670fe5c78@mail.gmail.com> Message-ID: <191c3a030901231455g2650cd7brd3736b96490fd55b@mail.gmail.com> Does AST mean Asterisk Open Source PBX ? If so, then yes I am familiar with it's archetechure as I am a former developer from that project. You have 3 choices with FreeSWITCH 1) You can open a dedicated connection to mod_event_socket or XMLRPC per call and issue the originate command from there: This will block until you know for sure the outcome of the attempt. If it's success it will give you the uuid if not it gives you the cause code. 2) You can use a single mod_event_socket or XMLRPC connection to send all calls but use the bgapi mechanism which will do the same as above only asynchronously, The command will return immediately and the result will be fired as an event that you can pick up on the same or different event_socket connection or other event consumer such as a custom C,perl,lua etc module. 3) You can use mod_xml_cdr to generate detailed 1-file-per-leg call files that will tell you when where and why the calls failed or did not fail. On Fri, Jan 23, 2009 at 4:39 PM, Michael Collins wrote: > On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey > wrote: > > Sorry for the double post ... actually hit send too early ... > > OK ... Here goes another I'm doing this with AST ... but I want to move > it > > to FS. Searched via google site:lists.freeswitch.org auto dialer and > others > > ... nothing useful. > > Today I have a platform for auto dialing with AST (centrally managed ... > > about 10 machines) and we do this: > > -- Remote machines query central DB for numbers to call based on > certain > > configs > > -- Use AMI to generate the call > > -- If call gets answered, extension info queried via rta (central db > > again) > > The nice thing about all of this is it's relatively easy to manage > (through > > one central web interface we built) and it works ... the bad part is > > reporting ... > > So ... conceptually I'm trying to accomplish the same thing ... > > Today we use FS a lot for termination of VoIP traffic ... all done via > > XML_CURL ... which is awesome (not to xml cdr ... and the "proxying" of > > media) ... > > Would like to do something like: > > -- originate request (looks simple enough) > > -- on answer XML_CURL posts info > > Several choices, depending upon how much you want it handled inside > the dialplan vs. handled in the scripting language. For the sake of > testing you could do something like this: > > > > > > > > Then have: > > > > > > > This would have any answered call go to the "ivr-answer" extension > while unanswered calls could stay in the ivr-start extension to get > properly handled. (Busy, no answer, invalid/SIT, etc.) > > You could then have the "ivr-answer" extension do whatever is > appropriate, like listen for digits, play announcement, beg for money, > etc. :) > > -MC > > > But for the life of me I can't figure out how to translate this into the > xml > > response ... > > [campaign] > > exten => 100,1,ANSWER() > > exten => 100,n,WAIT(2) > > exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile) > > exten => 100,n,WAITEXTEN(10) > > exten => 100,n,HANGUP() > > exten => 1,1,PLAYBACK(goodbye) > > .... and so on ... > > I've looked at the ivr.conf stuff but it's all static and all of this has > to > > be manageable via a web interface .... meaning dumping into a DB and > > returning an XML response seems reasonable ... but trying to stick or > modify > > static text files from the web interface is too much text parsing and bad > > things will happen ... > > Any thoughts or pointing me in the right direction would be appreciated. > > Shelby > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/712297b9/attachment.html From mark at markehle.net Fri Jan 23 14:47:38 2009 From: mark at markehle.net (Mark) Date: Fri, 23 Jan 2009 17:47:38 -0500 Subject: [Freeswitch-users] Freeswitch and an SPA941 In-Reply-To: <191c3a030901231440i1532cd04ub1cb61f44d756c85@mail.gmail.com> References: <35b355e90901231404lfb937b5x1152b4703ce88130@mail.gmail.com> <191c3a030901231440i1532cd04ub1cb61f44d756c85@mail.gmail.com> Message-ID: <20090123174738.15443m1etywdn5kw@markehle.net> Hello - I am brand-new to VOIP and have set up freeswitch on our pFsense firewall. It works great with X-lite. I would like to hook up a linksys SPA941 to the system, and am having trouble finding any documentation on how to configure it. Where do I start? Thanks - Mark From brian at freeswitch.org Fri Jan 23 15:19:42 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 23 Jan 2009 17:19:42 -0600 Subject: [Freeswitch-users] Freeswitch and an SPA941 In-Reply-To: <20090123174738.15443m1etywdn5kw@markehle.net> References: <35b355e90901231404lfb937b5x1152b4703ce88130@mail.gmail.com> <191c3a030901231440i1532cd04ub1cb61f44d756c85@mail.gmail.com> <20090123174738.15443m1etywdn5kw@markehle.net> Message-ID: <38A378D2-EDC6-45E5-B4CF-E80BC72C5568@freeswitch.org> Mark, Welcome to the FreeSWITCH Community. Check this out http://voxilla.com/tools/device-configuration-wizard/linksys-spa941-configuration-wizard Also when you post to the list please click new message, input the freeswitch-users at lists.freeswitch.org and do a new subject and click send. When you reply to an existing email, change the subject and contents you hijack a thread which can cause your email to go un-noticed which could result in your question not being answered sometimes. ;) Anyway the config wizard would help you get started. Thanks, Brian On Jan 23, 2009, at 4:47 PM, Mark wrote: > Hello - > > I am brand-new to VOIP and have set up freeswitch on our pFsense > firewall. It works great with X-lite. I would like to hook up a > linksys SPA941 to the system, and am having trouble finding any > documentation on how to configure it. Where do I start? > > Thanks - > > Mark > From pauld at versafon.com Fri Jan 23 15:35:41 2009 From: pauld at versafon.com (pauld) Date: Fri, 23 Jan 2009 18:35:41 -0500 Subject: [Freeswitch-users] bgapi uuid_kill question Message-ID: <1223981236.71232753768369.JavaMail.james@versafon31.versafon.com> Hi, Trying to kill a channel (hangup) when it's in CHANNEL_EXECUTE state as per mod_socket_event event information I use bgapi uuid_kill - but get response "ERR no such channel!". Same happens if I use unique_id or channel_name as an argument to uuid_kill. What I am doing wrong? Help would much appreciated. Thx. From brian at freeswitch.org Fri Jan 23 15:39:53 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 23 Jan 2009 17:39:53 -0600 Subject: [Freeswitch-users] bgapi uuid_kill question In-Reply-To: <1223981236.71232753768369.JavaMail.james@versafon31.versafon.com> References: <1223981236.71232753768369.JavaMail.james@versafon31.versafon.com> Message-ID: <0569E2A1-83C8-444C-A812-AB300A64864B@freeswitch.org> Don't use the core_uuid.. use the session uuid. /b On Jan 23, 2009, at 5:35 PM, pauld wrote: > Hi, > Trying to kill a channel (hangup) when it's in CHANNEL_EXECUTE state > as > per mod_socket_event event information I use bgapi uuid_kill > > - but get response "ERR no such channel!". Same happens if I use > unique_id or channel_name > as an argument to uuid_kill. What I am doing wrong? > Help would much appreciated. > Thx. From sicfslist at gmail.com Fri Jan 23 15:39:59 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 23 Jan 2009 17:39:59 -0600 Subject: [Freeswitch-users] auto dialing question ... In-Reply-To: <191c3a030901231455g2650cd7brd3736b96490fd55b@mail.gmail.com> References: <35b355e90901231415q782b5ba9mba2bfc7fe6345fd3@mail.gmail.com> <87f2f3b90901231439u553031f4u29c1f27670fe5c78@mail.gmail.com> <191c3a030901231455g2650cd7brd3736b96490fd55b@mail.gmail.com> Message-ID: <35b355e90901231539s1a2b938crd24279933efbbe2@mail.gmail.com> Anthony / Michael, Thanks for the quick responses. What I don't want to do is "drive the call" (by that listen on a socket ... do this on this event ... or anything else that my very limited FS foo would break) ... Just want to start it and then give it instructions on where to go. So I guess a better question would be ... how do I give directions to FS for this (and I get the 1st part ... that's obvious ... really lost on the DTMF digit part) ... and please keep in mind we're talking hundreds of extensions / IVR's and distributed machines so I can't have any dependancy on static conf files other than maybe something like what Michael mentioned where I point every call to something: [campaign] exten => 100,1,ANSWER() exten => 100,n,PLAYBACK(somefile) exten => 100,n,BACKGROUND(somefile) exten => 100,n,WAITEXTEN(4) exten => 100,n,HANGUP() but in that same context is someone triggers DTMF: exten => 1,1,DOSOMETHING exten => 2,1,DOSOMETHING I was imaging issuing originate via XML_RPC ... something like originate sofia/$ANI@$IP $SOMEEXTEN then on answer when FS tries to connect to $SOMEEXTEN it will ask me what to do via xml_curl ... where I would normally respond with something like this:
The challenge I've got is I have no idea how to do stuff like the IVR mentioned above (the playback part is easy) ... but I can't grasp conceptually how to get the "context" with "multiple extensions" part back to FS via this method (is it possible?)... Sorry for what is probably a very simple answer and any AST references (but I've been using it in heavy production environments for about 5 years). Just trying to "port" what I do today without making my brain melt out of my ears (and it doesn't take much for that to happen). Shelby PS ... Really enjoy the list. I usually fall out of my chair laughing once a day from your remarks Anthony. Keep it coming! On Fri, Jan 23, 2009 at 4:55 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Does AST mean Asterisk Open Source PBX ? > > If so, then yes I am familiar with it's archetechure as I am a former > developer from that project. > > You have 3 choices with FreeSWITCH > > 1) You can open a dedicated connection to mod_event_socket or XMLRPC per > call and issue the originate command from there: > This will block until you know for sure the outcome of the attempt. If > it's success it will give you the uuid if not it gives you the cause code. > > 2) You can use a single mod_event_socket or XMLRPC connection to send all > calls but use the bgapi mechanism which will do the same as above > only asynchronously, The command will return immediately and the result > will be fired as an event that you can pick up on the same or different > event_socket connection or > other event consumer such as a custom C,perl,lua etc module. > > 3) You can use mod_xml_cdr to generate detailed 1-file-per-leg call files > that will tell you when where and why the calls failed or did not fail. > > > > On Fri, Jan 23, 2009 at 4:39 PM, Michael Collins wrote: > >> On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey >> wrote: >> > Sorry for the double post ... actually hit send too early ... >> > OK ... Here goes another I'm doing this with AST ... but I want to move >> it >> > to FS. Searched via google site:lists.freeswitch.org auto dialer and >> others >> > ... nothing useful. >> > Today I have a platform for auto dialing with AST (centrally managed ... >> > about 10 machines) and we do this: >> > -- Remote machines query central DB for numbers to call based on >> certain >> > configs >> > -- Use AMI to generate the call >> > -- If call gets answered, extension info queried via rta (central db >> > again) >> > The nice thing about all of this is it's relatively easy to manage >> (through >> > one central web interface we built) and it works ... the bad part is >> > reporting ... >> > So ... conceptually I'm trying to accomplish the same thing ... >> > Today we use FS a lot for termination of VoIP traffic ... all done via >> > XML_CURL ... which is awesome (not to xml cdr ... and the "proxying" of >> > media) ... >> > Would like to do something like: >> > -- originate request (looks simple enough) >> > -- on answer XML_CURL posts info >> >> Several choices, depending upon how much you want it handled inside >> the dialplan vs. handled in the scripting language. For the sake of >> testing you could do something like this: >> >> >> >> >> >> >> >> Then have: >> >> >> >> >> >> >> This would have any answered call go to the "ivr-answer" extension >> while unanswered calls could stay in the ivr-start extension to get >> properly handled. (Busy, no answer, invalid/SIT, etc.) >> >> You could then have the "ivr-answer" extension do whatever is >> appropriate, like listen for digits, play announcement, beg for money, >> etc. :) >> >> -MC >> >> > But for the life of me I can't figure out how to translate this into the >> xml >> > response ... >> > [campaign] >> > exten => 100,1,ANSWER() >> > exten => 100,n,WAIT(2) >> > exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile) >> > exten => 100,n,WAITEXTEN(10) >> > exten => 100,n,HANGUP() >> > exten => 1,1,PLAYBACK(goodbye) >> > .... and so on ... >> > I've looked at the ivr.conf stuff but it's all static and all of this >> has to >> > be manageable via a web interface .... meaning dumping into a DB and >> > returning an XML response seems reasonable ... but trying to stick or >> modify >> > static text files from the web interface is too much text parsing and >> bad >> > things will happen ... >> > Any thoughts or pointing me in the right direction would be appreciated. >> > Shelby >> > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/4c99bf36/attachment-0001.html From anthony.minessale at gmail.com Fri Jan 23 15:55:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jan 2009 17:55:06 -0600 Subject: [Freeswitch-users] auto dialing question ... In-Reply-To: <35b355e90901231539s1a2b938crd24279933efbbe2@mail.gmail.com> References: <35b355e90901231415q782b5ba9mba2bfc7fe6345fd3@mail.gmail.com> <87f2f3b90901231439u553031f4u29c1f27670fe5c78@mail.gmail.com> <191c3a030901231455g2650cd7brd3736b96490fd55b@mail.gmail.com> <35b355e90901231539s1a2b938crd24279933efbbe2@mail.gmail.com> Message-ID: <191c3a030901231555w492357cbob4cfc4420ae4b745@mail.gmail.com> Try it from the FS CLI first to see how to do it. originate so type this in the console replacing the sofia url with one of your choice: originate sofia/default/100 at dom.com 9998 XML default when you answer the call you should hear the tetris song. so you can use any ext at context+dialplan combo with those args. You can do the same over xmlrpc or mod_event_socket On Fri, Jan 23, 2009 at 5:39 PM, Shelby Ramsey wrote: > Anthony / Michael, > Thanks for the quick responses. What I don't want to do is "drive the > call" (by that listen on a socket ... do this on this event ... or anything > else that my very limited FS foo would break) ... Just want to start it and > then give it instructions on where to go. > > So I guess a better question would be ... how do I give directions to FS > for this (and I get the 1st part ... that's obvious ... really lost on the > DTMF digit part) ... and please keep in mind we're talking hundreds of > extensions / IVR's and distributed machines so I can't have any dependancy > on static conf files other than maybe something like what Michael mentioned > where I point every call to something: > > [campaign] > exten => 100,1,ANSWER() > exten => 100,n,PLAYBACK(somefile) > exten => 100,n,BACKGROUND(somefile) > exten => 100,n,WAITEXTEN(4) > exten => 100,n,HANGUP() > > but in that same context is someone triggers DTMF: > exten => 1,1,DOSOMETHING > exten => 2,1,DOSOMETHING > > I was imaging issuing originate via XML_RPC ... something like originate > sofia/$ANI@$IP $SOMEEXTEN then on answer when FS tries to connect to > $SOMEEXTEN it will ask me what to do via xml_curl ... where I would normally > respond with something like this: > > type="freeswitch/xml">
> field="destination_number" expression=""> data="hangup_after_bridge=true"/> data="continue_on_fail=true"/> data="call_timeout=180"/> data="proxy_media=true"/> data="pass_rfc2833=true"/> data="accountcode=$CUSTOMER" /> data="origination_caller_id_name=NULL" /> data="origination_caller_id_number=$CIDNUM" /> data="effective_caller_id_name=NULL" /> data="effective_caller_id_number=$CIDNUM" /> data="userfield=$BUNCHOFCRAPFORMYCDR" /> data="sofia/external/$ANI@$PROVIDERIP" /> > >
> > The challenge I've got is I have no idea how to do stuff like the IVR > mentioned above (the playback part is easy) ... but I can't grasp > conceptually how to get the "context" with "multiple extensions" part back > to FS via this method (is it possible?)... > > Sorry for what is probably a very simple answer and any AST references (but > I've been using it in heavy production environments for about 5 years). Just > trying to "port" what I do today without making my brain melt out of my ears > (and it doesn't take much for that to happen). > > Shelby > > PS ... Really enjoy the list. I usually fall out of my chair laughing once > a day from your remarks Anthony. Keep it coming! > > > On Fri, Jan 23, 2009 at 4:55 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Does AST mean Asterisk Open Source PBX ? >> >> If so, then yes I am familiar with it's archetechure as I am a former >> developer from that project. >> >> You have 3 choices with FreeSWITCH >> >> 1) You can open a dedicated connection to mod_event_socket or XMLRPC per >> call and issue the originate command from there: >> This will block until you know for sure the outcome of the attempt. >> If it's success it will give you the uuid if not it gives you the cause >> code. >> >> 2) You can use a single mod_event_socket or XMLRPC connection to send all >> calls but use the bgapi mechanism which will do the same as above >> only asynchronously, The command will return immediately and the >> result will be fired as an event that you can pick up on the same or >> different event_socket connection or >> other event consumer such as a custom C,perl,lua etc module. >> >> 3) You can use mod_xml_cdr to generate detailed 1-file-per-leg call files >> that will tell you when where and why the calls failed or did not fail. >> >> >> >> On Fri, Jan 23, 2009 at 4:39 PM, Michael Collins wrote: >> >>> On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey >>> wrote: >>> > Sorry for the double post ... actually hit send too early ... >>> > OK ... Here goes another I'm doing this with AST ... but I want to >>> move it >>> > to FS. Searched via google site:lists.freeswitch.org auto dialer and >>> others >>> > ... nothing useful. >>> > Today I have a platform for auto dialing with AST (centrally managed >>> ... >>> > about 10 machines) and we do this: >>> > -- Remote machines query central DB for numbers to call based on >>> certain >>> > configs >>> > -- Use AMI to generate the call >>> > -- If call gets answered, extension info queried via rta (central db >>> > again) >>> > The nice thing about all of this is it's relatively easy to manage >>> (through >>> > one central web interface we built) and it works ... the bad part is >>> > reporting ... >>> > So ... conceptually I'm trying to accomplish the same thing ... >>> > Today we use FS a lot for termination of VoIP traffic ... all done via >>> > XML_CURL ... which is awesome (not to xml cdr ... and the "proxying" >>> of >>> > media) ... >>> > Would like to do something like: >>> > -- originate request (looks simple enough) >>> > -- on answer XML_CURL posts info >>> >>> Several choices, depending upon how much you want it handled inside >>> the dialplan vs. handled in the scripting language. For the sake of >>> testing you could do something like this: >>> >>> >>> >>> >>> >>> >>> >>> Then have: >>> >>> >>> >>> >>> >>> >>> This would have any answered call go to the "ivr-answer" extension >>> while unanswered calls could stay in the ivr-start extension to get >>> properly handled. (Busy, no answer, invalid/SIT, etc.) >>> >>> You could then have the "ivr-answer" extension do whatever is >>> appropriate, like listen for digits, play announcement, beg for money, >>> etc. :) >>> >>> -MC >>> >>> > But for the life of me I can't figure out how to translate this into >>> the xml >>> > response ... >>> > [campaign] >>> > exten => 100,1,ANSWER() >>> > exten => 100,n,WAIT(2) >>> > exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile) >>> > exten => 100,n,WAITEXTEN(10) >>> > exten => 100,n,HANGUP() >>> > exten => 1,1,PLAYBACK(goodbye) >>> > .... and so on ... >>> > I've looked at the ivr.conf stuff but it's all static and all of this >>> has to >>> > be manageable via a web interface .... meaning dumping into a DB and >>> > returning an XML response seems reasonable ... but trying to stick or >>> modify >>> > static text files from the web interface is too much text parsing and >>> bad >>> > things will happen ... >>> > Any thoughts or pointing me in the right direction would be >>> appreciated. >>> > Shelby >>> > >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at 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 at 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 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 >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/9b54acce/attachment.html From msc at freeswitch.org Fri Jan 23 15:55:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Jan 2009 15:55:26 -0800 Subject: [Freeswitch-users] auto dialing question ... In-Reply-To: <35b355e90901231539s1a2b938crd24279933efbbe2@mail.gmail.com> References: <35b355e90901231415q782b5ba9mba2bfc7fe6345fd3@mail.gmail.com> <87f2f3b90901231439u553031f4u29c1f27670fe5c78@mail.gmail.com> <191c3a030901231455g2650cd7brd3736b96490fd55b@mail.gmail.com> <35b355e90901231539s1a2b938crd24279933efbbe2@mail.gmail.com> Message-ID: <87f2f3b90901231555i6d1690a4p25ecece2340acc4e@mail.gmail.com> I see your dilemma. To keep things dynamic you definitely want to use your XML_CURL stuff. Like you said, nothing static. Can you post a sample call-flow in plain English? I'm curious about something. I don't want to say anything else. Just post a simple call flow: Initiate call Wait for answer/busy/timeout/invalid On busy/timeout/invalid: update db and move on On answer play greeting, accept digit, route based on digit Something like that would help me conceptualize what you are trying to do. Thanks MC P.S. - I've done a little bit of outbound IVR calling so hopefully I can assist you. On Fri, Jan 23, 2009 at 3:39 PM, Shelby Ramsey wrote: > Anthony / Michael, > Thanks for the quick responses. What I don't want to do is "drive the call" > (by that listen on a socket ... do this on this event ... or anything else > that my very limited FS foo would break) ... Just want to start it and then > give it instructions on where to go. > So I guess a better question would be ... how do I give directions to FS for > this (and I get the 1st part ... that's obvious ... really lost on the DTMF > digit part) ... and please keep in mind we're talking hundreds of extensions > / IVR's and distributed machines so I can't have any dependancy on static > conf files other than maybe something like what Michael mentioned where I > point every call to something: > [campaign] > exten => 100,1,ANSWER() > exten => 100,n,PLAYBACK(somefile) > exten => 100,n,BACKGROUND(somefile) > exten => 100,n,WAITEXTEN(4) > exten => 100,n,HANGUP() > but in that same context is someone triggers DTMF: > exten => 1,1,DOSOMETHING > exten => 2,1,DOSOMETHING > I was imaging issuing originate via XML_RPC ... something like originate > sofia/$ANI@$IP $SOMEEXTEN then on answer when FS tries to connect to > $SOMEEXTEN it will ask me what to do via xml_curl ... where I would normally > respond with something like this: > type="freeswitch/xml">
> field="destination_number" expression=""> data="hangup_after_bridge=true"/> data="continue_on_fail=true"/> data="call_timeout=180"/> data="proxy_media=true"/> data="pass_rfc2833=true"/> data="accountcode=$CUSTOMER" /> data="origination_caller_id_name=NULL" /> data="origination_caller_id_number=$CIDNUM" /> data="effective_caller_id_name=NULL" /> data="effective_caller_id_number=$CIDNUM" /> data="userfield=$BUNCHOFCRAPFORMYCDR" /> data="sofia/external/$ANI@$PROVIDERIP" /> > >
> The challenge I've got is I have no idea how to do stuff like the IVR > mentioned above (the playback part is easy) ... but I can't grasp > conceptually how to get the "context" with "multiple extensions" part back > to FS via this method (is it possible?)... > Sorry for what is probably a very simple answer and any AST references (but > I've been using it in heavy production environments for about 5 years). Just > trying to "port" what I do today without making my brain melt out of my ears > (and it doesn't take much for that to happen). > Shelby > PS ... Really enjoy the list. I usually fall out of my chair laughing once a > day from your remarks Anthony. Keep it coming! > > On Fri, Jan 23, 2009 at 4:55 PM, Anthony Minessale > wrote: >> >> Does AST mean Asterisk Open Source PBX ? >> >> If so, then yes I am familiar with it's archetechure as I am a former >> developer from that project. >> >> You have 3 choices with FreeSWITCH >> >> 1) You can open a dedicated connection to mod_event_socket or XMLRPC per >> call and issue the originate command from there: >> This will block until you know for sure the outcome of the attempt. >> If it's success it will give you the uuid if not it gives you the cause >> code. >> >> 2) You can use a single mod_event_socket or XMLRPC connection to send all >> calls but use the bgapi mechanism which will do the same as above >> only asynchronously, The command will return immediately and the >> result will be fired as an event that you can pick up on the same or >> different event_socket connection or >> other event consumer such as a custom C,perl,lua etc module. >> >> 3) You can use mod_xml_cdr to generate detailed 1-file-per-leg call files >> that will tell you when where and why the calls failed or did not fail. >> >> >> On Fri, Jan 23, 2009 at 4:39 PM, Michael Collins >> wrote: >>> >>> On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey >>> wrote: >>> > Sorry for the double post ... actually hit send too early ... >>> > OK ... Here goes another I'm doing this with AST ... but I want to >>> > move it >>> > to FS. Searched via google site:lists.freeswitch.org auto dialer and >>> > others >>> > ... nothing useful. >>> > Today I have a platform for auto dialing with AST (centrally managed >>> > ... >>> > about 10 machines) and we do this: >>> > -- Remote machines query central DB for numbers to call based on >>> > certain >>> > configs >>> > -- Use AMI to generate the call >>> > -- If call gets answered, extension info queried via rta (central db >>> > again) >>> > The nice thing about all of this is it's relatively easy to manage >>> > (through >>> > one central web interface we built) and it works ... the bad part is >>> > reporting ... >>> > So ... conceptually I'm trying to accomplish the same thing ... >>> > Today we use FS a lot for termination of VoIP traffic ... all done via >>> > XML_CURL ... which is awesome (not to xml cdr ... and the "proxying" >>> > of >>> > media) ... >>> > Would like to do something like: >>> > -- originate request (looks simple enough) >>> > -- on answer XML_CURL posts info >>> >>> Several choices, depending upon how much you want it handled inside >>> the dialplan vs. handled in the scripting language. For the sake of >>> testing you could do something like this: >>> >>> >>> >>> >>> >>> >>> >>> Then have: >>> >>> >>> >>> >>> >>> >>> This would have any answered call go to the "ivr-answer" extension >>> while unanswered calls could stay in the ivr-start extension to get >>> properly handled. (Busy, no answer, invalid/SIT, etc.) >>> >>> You could then have the "ivr-answer" extension do whatever is >>> appropriate, like listen for digits, play announcement, beg for money, >>> etc. :) >>> >>> -MC >>> >>> > But for the life of me I can't figure out how to translate this into >>> > the xml >>> > response ... >>> > [campaign] >>> > exten => 100,1,ANSWER() >>> > exten => 100,n,WAIT(2) >>> > exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile) >>> > exten => 100,n,WAITEXTEN(10) >>> > exten => 100,n,HANGUP() >>> > exten => 1,1,PLAYBACK(goodbye) >>> > .... and so on ... >>> > I've looked at the ivr.conf stuff but it's all static and all of this >>> > has to >>> > be manageable via a web interface .... meaning dumping into a DB and >>> > returning an XML response seems reasonable ... but trying to stick or >>> > modify >>> > static text files from the web interface is too much text parsing and >>> > bad >>> > things will happen ... >>> > Any thoughts or pointing me in the right direction would be >>> > appreciated. >>> > Shelby >>> > >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at 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 at 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 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 >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From pauld at versafon.com Fri Jan 23 15:56:59 2009 From: pauld at versafon.com (pauld) Date: Fri, 23 Jan 2009 18:56:59 -0500 Subject: [Freeswitch-users] bgapi uuid_kill question In-Reply-To: <0569E2A1-83C8-444C-A812-AB300A64864B@freeswitch.org> References: <1223981236.71232753768369.JavaMail.james@versafon31.versafon.com> <0569E2A1-83C8-444C-A812-AB300A64864B@freeswitch.org> Message-ID: <1748763617.81232755044367.JavaMail.james@versafon31.versafon.com> Oh, But where do I get it from? It's not seem to be available from CHANNEL_EXECUTE event or any other channel related event. Brian West wrote: > Don't use the core_uuid.. use the session uuid. > > /b > > On Jan 23, 2009, at 5:35 PM, pauld wrote: > > >> Hi, >> Trying to kill a channel (hangup) when it's in CHANNEL_EXECUTE state >> as >> per mod_socket_event event information I use bgapi uuid_kill >> >> - but get response "ERR no such channel!". Same happens if I use >> unique_id or channel_name >> as an argument to uuid_kill. What I am doing wrong? >> Help would much appreciated. >> Thx. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Fri Jan 23 16:00:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jan 2009 18:00:03 -0600 Subject: [Freeswitch-users] bgapi uuid_kill question In-Reply-To: <1748763617.81232755044367.JavaMail.james@versafon31.versafon.com> References: <1223981236.71232753768369.JavaMail.james@versafon31.versafon.com> <0569E2A1-83C8-444C-A812-AB300A64864B@freeswitch.org> <1748763617.81232755044367.JavaMail.james@versafon31.versafon.com> Message-ID: <191c3a030901231600v63bdbcf1w483980eaaf333a5b@mail.gmail.com> Unique-ID On Fri, Jan 23, 2009 at 5:56 PM, pauld wrote: > Oh, > But where do I get it from? It's not seem to be available from > CHANNEL_EXECUTE event or any other channel related event. > > > Brian West wrote: > > Don't use the core_uuid.. use the session uuid. > > > > /b > > > > On Jan 23, 2009, at 5:35 PM, pauld wrote: > > > > > >> Hi, > >> Trying to kill a channel (hangup) when it's in CHANNEL_EXECUTE state > >> as > >> per mod_socket_event event information I use bgapi uuid_kill > >> > >> - but get response "ERR no such channel!". Same happens if I use > >> unique_id or channel_name > >> as an argument to uuid_kill. What I am doing wrong? > >> Help would much appreciated. > >> Thx. > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/49a99c6a/attachment.html From can_man at gmx.de Fri Jan 23 15:36:21 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Sat, 24 Jan 2009 00:36:21 +0100 Subject: [Freeswitch-users] Pylons example on the wiki Message-ID: <20090123233621.283650@gmx.net> Hello, I just want to say that I have uploaded a Pylons wiki page with examples in svn. Pylons is a python web framework, which I think can help people to get started easily with xml_curl. If there is demand to extend the examples I am happy to do at. Suggestions are always welcome. You can normally find me on irc: phm_it Cheers, Phil -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a From sicfslist at gmail.com Fri Jan 23 16:20:21 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 23 Jan 2009 18:20:21 -0600 Subject: [Freeswitch-users] auto dialing question ... In-Reply-To: <87f2f3b90901231555i6d1690a4p25ecece2340acc4e@mail.gmail.com> References: <35b355e90901231415q782b5ba9mba2bfc7fe6345fd3@mail.gmail.com> <87f2f3b90901231439u553031f4u29c1f27670fe5c78@mail.gmail.com> <191c3a030901231455g2650cd7brd3736b96490fd55b@mail.gmail.com> <35b355e90901231539s1a2b938crd24279933efbbe2@mail.gmail.com> <87f2f3b90901231555i6d1690a4p25ecece2340acc4e@mail.gmail.com> Message-ID: <35b355e90901231620t83e9320vebfe6728d9b3ebc7@mail.gmail.com> Thanks Michael. Here are some really simple examples (we limit what people do through the web ... nothing really fancy ... just some good old fashion robo calls). ultimately today it looks like this extensions_table (context, exten, priority, app, appdata): campaign,100,1,ANSWER() --> answers call campaign,100,2,WAIT(1) --> this is a pause campaign,100,3,PLAYBACK($INTROFILE) --> plays intro file campaign,100,4,BACKGROUND($MESSAGE) --> plays message campaign,100,5,WAITEXTEN(5) --> WAIT for DTMF campaign,100,6,HANGUP() Here are DTMF options campaign,1,SYSTEM($SCRIPT campaign $EXTEN) campaign,2,SYSTEM($SCRIPT campaign $EXTEN) campaign,3,DIAL,zap/g1/$ANI anothercampaign,112,1,ANSWER() anothercampaign,112,2,BACKGROUND($SOMEFILE) anothercampaign,112,3,WAITEXTEN(5) anothercampaign,112,4,HANGUP() Options: anothercampaign,1,DIAL(sip/$ANI@$IP) anothercampaign,2,PLAYBACK($GOODBYE) And they pretty much all look like that ... it's easy to return the stuff for extension 100 via XML ... but the challenge is the DTMF options (relating to the same context) ... or maybe I'm just missing something (which is a definite possibility). We don't ever do anything complicated in the IVRs (TTS or ASR) but there is just a lot of them that all get controlled, manipulated via a web interface. I can originate a call ... do all of that ... just trying to figure out a "simple" way to return the above via xml. Thanks for your help Anthony and Michael as always! Shelby On Fri, Jan 23, 2009 at 5:55 PM, Michael Collins wrote: > I see your dilemma. To keep things dynamic you definitely want to use > your XML_CURL stuff. Like you said, nothing static. > > Can you post a sample call-flow in plain English? I'm curious about > something. I don't want to say anything else. Just post a simple call > flow: > Initiate call > Wait for answer/busy/timeout/invalid > On busy/timeout/invalid: update db and move on > On answer play greeting, accept digit, route based on digit > > Something like that would help me conceptualize what you are trying to do. > > Thanks > MC > > P.S. - I've done a little bit of outbound IVR calling so hopefully I > can assist you. > > On Fri, Jan 23, 2009 at 3:39 PM, Shelby Ramsey > wrote: > > Anthony / Michael, > > Thanks for the quick responses. What I don't want to do is "drive the > call" > > (by that listen on a socket ... do this on this event ... or anything > else > > that my very limited FS foo would break) ... Just want to start it and > then > > give it instructions on where to go. > > So I guess a better question would be ... how do I give directions to FS > for > > this (and I get the 1st part ... that's obvious ... really lost on the > DTMF > > digit part) ... and please keep in mind we're talking hundreds of > extensions > > / IVR's and distributed machines so I can't have any dependancy on static > > conf files other than maybe something like what Michael mentioned where I > > point every call to something: > > [campaign] > > exten => 100,1,ANSWER() > > exten => 100,n,PLAYBACK(somefile) > > exten => 100,n,BACKGROUND(somefile) > > exten => 100,n,WAITEXTEN(4) > > exten => 100,n,HANGUP() > > but in that same context is someone triggers DTMF: > > exten => 1,1,DOSOMETHING > > exten => 2,1,DOSOMETHING > > I was imaging issuing originate via XML_RPC ... something like originate > > sofia/$ANI@$IP $SOMEEXTEN then on answer when FS tries to connect to > > $SOMEEXTEN it will ask me what to do via xml_curl ... where I would > normally > > respond with something like this: > > > type="freeswitch/xml">
> > > field="destination_number" expression=""> > data="hangup_after_bridge=true"/> > data="continue_on_fail=true"/> > data="call_timeout=180"/> > data="proxy_media=true"/> > data="pass_rfc2833=true"/> > data="accountcode=$CUSTOMER" /> > data="origination_caller_id_name=NULL" /> > data="origination_caller_id_number=$CIDNUM" /> > data="effective_caller_id_name=NULL" /> > data="effective_caller_id_number=$CIDNUM" /> > data="userfield=$BUNCHOFCRAPFORMYCDR" /> > data="sofia/external/$ANI@$PROVIDERIP" /> > > > >
> > The challenge I've got is I have no idea how to do stuff like the IVR > > mentioned above (the playback part is easy) ... but I can't grasp > > conceptually how to get the "context" with "multiple extensions" part > back > > to FS via this method (is it possible?)... > > Sorry for what is probably a very simple answer and any AST references > (but > > I've been using it in heavy production environments for about 5 years). > Just > > trying to "port" what I do today without making my brain melt out of my > ears > > (and it doesn't take much for that to happen). > > Shelby > > PS ... Really enjoy the list. I usually fall out of my chair laughing > once a > > day from your remarks Anthony. Keep it coming! > > > > On Fri, Jan 23, 2009 at 4:55 PM, Anthony Minessale > > wrote: > >> > >> Does AST mean Asterisk Open Source PBX ? > >> > >> If so, then yes I am familiar with it's archetechure as I am a former > >> developer from that project. > >> > >> You have 3 choices with FreeSWITCH > >> > >> 1) You can open a dedicated connection to mod_event_socket or XMLRPC per > >> call and issue the originate command from there: > >> This will block until you know for sure the outcome of the attempt. > >> If it's success it will give you the uuid if not it gives you the cause > >> code. > >> > >> 2) You can use a single mod_event_socket or XMLRPC connection to send > all > >> calls but use the bgapi mechanism which will do the same as above > >> only asynchronously, The command will return immediately and the > >> result will be fired as an event that you can pick up on the same or > >> different event_socket connection or > >> other event consumer such as a custom C,perl,lua etc module. > >> > >> 3) You can use mod_xml_cdr to generate detailed 1-file-per-leg call > files > >> that will tell you when where and why the calls failed or did not fail. > >> > >> > >> On Fri, Jan 23, 2009 at 4:39 PM, Michael Collins > >> wrote: > >>> > >>> On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey > >>> wrote: > >>> > Sorry for the double post ... actually hit send too early ... > >>> > OK ... Here goes another I'm doing this with AST ... but I want to > >>> > move it > >>> > to FS. Searched via google site:lists.freeswitch.org auto dialer > and > >>> > others > >>> > ... nothing useful. > >>> > Today I have a platform for auto dialing with AST (centrally managed > >>> > ... > >>> > about 10 machines) and we do this: > >>> > -- Remote machines query central DB for numbers to call based on > >>> > certain > >>> > configs > >>> > -- Use AMI to generate the call > >>> > -- If call gets answered, extension info queried via rta (central > db > >>> > again) > >>> > The nice thing about all of this is it's relatively easy to manage > >>> > (through > >>> > one central web interface we built) and it works ... the bad part is > >>> > reporting ... > >>> > So ... conceptually I'm trying to accomplish the same thing ... > >>> > Today we use FS a lot for termination of VoIP traffic ... all done > via > >>> > XML_CURL ... which is awesome (not to xml cdr ... and the "proxying" > >>> > of > >>> > media) ... > >>> > Would like to do something like: > >>> > -- originate request (looks simple enough) > >>> > -- on answer XML_CURL posts info > >>> > >>> Several choices, depending upon how much you want it handled inside > >>> the dialplan vs. handled in the scripting language. For the sake of > >>> testing you could do something like this: > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> Then have: > >>> > >>> > >>> > >>> > >>> > >>> > >>> This would have any answered call go to the "ivr-answer" extension > >>> while unanswered calls could stay in the ivr-start extension to get > >>> properly handled. (Busy, no answer, invalid/SIT, etc.) > >>> > >>> You could then have the "ivr-answer" extension do whatever is > >>> appropriate, like listen for digits, play announcement, beg for money, > >>> etc. :) > >>> > >>> -MC > >>> > >>> > But for the life of me I can't figure out how to translate this into > >>> > the xml > >>> > response ... > >>> > [campaign] > >>> > exten => 100,1,ANSWER() > >>> > exten => 100,n,WAIT(2) > >>> > exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile) > >>> > exten => 100,n,WAITEXTEN(10) > >>> > exten => 100,n,HANGUP() > >>> > exten => 1,1,PLAYBACK(goodbye) > >>> > .... and so on ... > >>> > I've looked at the ivr.conf stuff but it's all static and all of this > >>> > has to > >>> > be manageable via a web interface .... meaning dumping into a DB and > >>> > returning an XML response seems reasonable ... but trying to stick or > >>> > modify > >>> > static text files from the web interface is too much text parsing and > >>> > bad > >>> > things will happen ... > >>> > Any thoughts or pointing me in the right direction would be > >>> > appreciated. > >>> > Shelby > >>> > > >>> > > >>> > > >>> > _______________________________________________ > >>> > Freeswitch-users mailing list > >>> > Freeswitch-users at 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 at 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 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 > >> iax:guest at conference.freeswitch.org/888 > >> googletalk:conf+888 at conference.freeswitch.org > >> pstn:213-799-1400 > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/b463a30f/attachment-0001.html From brian at freeswitch.org Fri Jan 23 16:37:13 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 23 Jan 2009 18:37:13 -0600 Subject: [Freeswitch-users] Wideband/Ultrawideband Conference Message-ID: <38716ACB-BBED-4894-9FCE-543644141B84@freeswitch.org> FreeSWITCHers, We have set the 888 at conference.freeswitch.org conference to be 32k, This weekend I'm going to move it to be 48k by default. So dust off your Logitech 350 headsets ( or go buy one from your favorite retailer ). The 350 has the 16kHz pickup and the 20kHz output required to take the most advantage of the 48k CELT codec. So check it out... btw we hangout in the conference all day on Fridays so if you want to call in, ask questions or just interact with the developers of FreeSWITCH... Stop by ... say Hi... Thanks, /b PS: Don't forget to tell a friend about FreeSWITCH! From msc at freeswitch.org Fri Jan 23 16:46:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Jan 2009 16:46:09 -0800 Subject: [Freeswitch-users] Wideband/Ultrawideband Conference In-Reply-To: <38716ACB-BBED-4894-9FCE-543644141B84@freeswitch.org> References: <38716ACB-BBED-4894-9FCE-543644141B84@freeswitch.org> Message-ID: <87f2f3b90901231646t379ba9a9xb1b738dd021407a6@mail.gmail.com> On Fri, Jan 23, 2009 at 4:37 PM, Brian West wrote: > FreeSWITCHers, > We have set the 888 at conference.freeswitch.org conference to be 32k, > This weekend I'm going to move it to be 48k by default. So dust off > your Logitech 350 headsets ( or go buy one from your favorite > retailer ). > > The 350 has the 16kHz pickup and the 20kHz output required to take the > most advantage of the 48k CELT codec. So check it out... btw we > hangout in the conference all day on Fridays so if you want to call > in, ask questions or just interact with the developers of > FreeSWITCH... Stop by ... say Hi... Also, we're more likely to answer your questions if you're not running 8kHz audio! ;) -MC > > Thanks, > /b > PS: Don't forget to tell a friend about FreeSWITCH! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Fri Jan 23 16:54:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jan 2009 18:54:56 -0600 Subject: [Freeswitch-users] auto dialing question ... In-Reply-To: <35b355e90901231620t83e9320vebfe6728d9b3ebc7@mail.gmail.com> References: <35b355e90901231415q782b5ba9mba2bfc7fe6345fd3@mail.gmail.com> <87f2f3b90901231439u553031f4u29c1f27670fe5c78@mail.gmail.com> <191c3a030901231455g2650cd7brd3736b96490fd55b@mail.gmail.com> <35b355e90901231539s1a2b938crd24279933efbbe2@mail.gmail.com> <87f2f3b90901231555i6d1690a4p25ecece2340acc4e@mail.gmail.com> <35b355e90901231620t83e9320vebfe6728d9b3ebc7@mail.gmail.com> Message-ID: <191c3a030901231654w2823ed72h382dee33060acd2a@mail.gmail.com> its the same idea where i gave you that example to call 9998 call 1234 where your extension is delivered via curl when 1234 is requested use the "read" and "transfer" apps in the xml you return, you can get it working statically first in your regular dialplan. On Fri, Jan 23, 2009 at 6:20 PM, Shelby Ramsey wrote: > Thanks Michael. > Here are some really simple examples (we limit what people do through the > web ... nothing really fancy ... just some good old fashion robo calls). > ultimately today it looks like this extensions_table (context, exten, > priority, app, appdata): > > campaign,100,1,ANSWER() --> answers call > campaign,100,2,WAIT(1) --> this is a pause > campaign,100,3,PLAYBACK($INTROFILE) --> plays intro file > campaign,100,4,BACKGROUND($MESSAGE) --> plays message > campaign,100,5,WAITEXTEN(5) --> WAIT for DTMF > campaign,100,6,HANGUP() > Here are DTMF options > campaign,1,SYSTEM($SCRIPT campaign $EXTEN) > campaign,2,SYSTEM($SCRIPT campaign $EXTEN) > campaign,3,DIAL,zap/g1/$ANI > > anothercampaign,112,1,ANSWER() > anothercampaign,112,2,BACKGROUND($SOMEFILE) > anothercampaign,112,3,WAITEXTEN(5) > anothercampaign,112,4,HANGUP() > Options: > anothercampaign,1,DIAL(sip/$ANI@$IP) > anothercampaign,2,PLAYBACK($GOODBYE) > > And they pretty much all look like that ... it's easy to return the stuff > for extension 100 via XML ... but the challenge is the DTMF options > (relating to the same context) ... or maybe I'm just missing something > (which is a definite possibility). We don't ever do anything complicated in > the IVRs (TTS or ASR) but there is just a lot of them that all get > controlled, manipulated via a web interface. > > I can originate a call ... do all of that ... just trying to figure out a > "simple" way to return the above via xml. > > Thanks for your help Anthony and Michael as always! > > Shelby > > > > > > On Fri, Jan 23, 2009 at 5:55 PM, Michael Collins wrote: > >> I see your dilemma. To keep things dynamic you definitely want to use >> your XML_CURL stuff. Like you said, nothing static. >> >> Can you post a sample call-flow in plain English? I'm curious about >> something. I don't want to say anything else. Just post a simple call >> flow: >> Initiate call >> Wait for answer/busy/timeout/invalid >> On busy/timeout/invalid: update db and move on >> On answer play greeting, accept digit, route based on digit >> >> Something like that would help me conceptualize what you are trying to do. >> >> Thanks >> MC >> >> P.S. - I've done a little bit of outbound IVR calling so hopefully I >> can assist you. >> >> On Fri, Jan 23, 2009 at 3:39 PM, Shelby Ramsey >> wrote: >> > Anthony / Michael, >> > Thanks for the quick responses. What I don't want to do is "drive the >> call" >> > (by that listen on a socket ... do this on this event ... or anything >> else >> > that my very limited FS foo would break) ... Just want to start it and >> then >> > give it instructions on where to go. >> > So I guess a better question would be ... how do I give directions to FS >> for >> > this (and I get the 1st part ... that's obvious ... really lost on the >> DTMF >> > digit part) ... and please keep in mind we're talking hundreds of >> extensions >> > / IVR's and distributed machines so I can't have any dependancy on >> static >> > conf files other than maybe something like what Michael mentioned where >> I >> > point every call to something: >> > [campaign] >> > exten => 100,1,ANSWER() >> > exten => 100,n,PLAYBACK(somefile) >> > exten => 100,n,BACKGROUND(somefile) >> > exten => 100,n,WAITEXTEN(4) >> > exten => 100,n,HANGUP() >> > but in that same context is someone triggers DTMF: >> > exten => 1,1,DOSOMETHING >> > exten => 2,1,DOSOMETHING >> > I was imaging issuing originate via XML_RPC ... something like originate >> > sofia/$ANI@$IP $SOMEEXTEN then on answer when FS tries to connect to >> > $SOMEEXTEN it will ask me what to do via xml_curl ... where I would >> normally >> > respond with something like this: >> > > > type="freeswitch/xml">
>> > > > field="destination_number" expression=""> > > data="hangup_after_bridge=true"/> > > data="continue_on_fail=true"/> > > data="call_timeout=180"/> > > data="proxy_media=true"/> > > data="pass_rfc2833=true"/> > > data="accountcode=$CUSTOMER" /> > > data="origination_caller_id_name=NULL" /> > > data="origination_caller_id_number=$CIDNUM" /> > > data="effective_caller_id_name=NULL" /> > > data="effective_caller_id_number=$CIDNUM" /> > > data="userfield=$BUNCHOFCRAPFORMYCDR" /> > > data="sofia/external/$ANI@$PROVIDERIP" /> >> > >> >
>> > The challenge I've got is I have no idea how to do stuff like the IVR >> > mentioned above (the playback part is easy) ... but I can't grasp >> > conceptually how to get the "context" with "multiple extensions" part >> back >> > to FS via this method (is it possible?)... >> > Sorry for what is probably a very simple answer and any AST references >> (but >> > I've been using it in heavy production environments for about 5 years). >> Just >> > trying to "port" what I do today without making my brain melt out of my >> ears >> > (and it doesn't take much for that to happen). >> > Shelby >> > PS ... Really enjoy the list. I usually fall out of my chair laughing >> once a >> > day from your remarks Anthony. Keep it coming! >> > >> > On Fri, Jan 23, 2009 at 4:55 PM, Anthony Minessale >> > wrote: >> >> >> >> Does AST mean Asterisk Open Source PBX ? >> >> >> >> If so, then yes I am familiar with it's archetechure as I am a former >> >> developer from that project. >> >> >> >> You have 3 choices with FreeSWITCH >> >> >> >> 1) You can open a dedicated connection to mod_event_socket or XMLRPC >> per >> >> call and issue the originate command from there: >> >> This will block until you know for sure the outcome of the attempt. >> >> If it's success it will give you the uuid if not it gives you the cause >> >> code. >> >> >> >> 2) You can use a single mod_event_socket or XMLRPC connection to send >> all >> >> calls but use the bgapi mechanism which will do the same as above >> >> only asynchronously, The command will return immediately and the >> >> result will be fired as an event that you can pick up on the same or >> >> different event_socket connection or >> >> other event consumer such as a custom C,perl,lua etc module. >> >> >> >> 3) You can use mod_xml_cdr to generate detailed 1-file-per-leg call >> files >> >> that will tell you when where and why the calls failed or did not fail. >> >> >> >> >> >> On Fri, Jan 23, 2009 at 4:39 PM, Michael Collins >> >> wrote: >> >>> >> >>> On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey >> >>> wrote: >> >>> > Sorry for the double post ... actually hit send too early ... >> >>> > OK ... Here goes another I'm doing this with AST ... but I want to >> >>> > move it >> >>> > to FS. Searched via google site:lists.freeswitch.org auto dialer >> and >> >>> > others >> >>> > ... nothing useful. >> >>> > Today I have a platform for auto dialing with AST (centrally managed >> >>> > ... >> >>> > about 10 machines) and we do this: >> >>> > -- Remote machines query central DB for numbers to call based on >> >>> > certain >> >>> > configs >> >>> > -- Use AMI to generate the call >> >>> > -- If call gets answered, extension info queried via rta (central >> db >> >>> > again) >> >>> > The nice thing about all of this is it's relatively easy to manage >> >>> > (through >> >>> > one central web interface we built) and it works ... the bad part is >> >>> > reporting ... >> >>> > So ... conceptually I'm trying to accomplish the same thing ... >> >>> > Today we use FS a lot for termination of VoIP traffic ... all done >> via >> >>> > XML_CURL ... which is awesome (not to xml cdr ... and the >> "proxying" >> >>> > of >> >>> > media) ... >> >>> > Would like to do something like: >> >>> > -- originate request (looks simple enough) >> >>> > -- on answer XML_CURL posts info >> >>> >> >>> Several choices, depending upon how much you want it handled inside >> >>> the dialplan vs. handled in the scripting language. For the sake of >> >>> testing you could do something like this: >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> Then have: >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> This would have any answered call go to the "ivr-answer" extension >> >>> while unanswered calls could stay in the ivr-start extension to get >> >>> properly handled. (Busy, no answer, invalid/SIT, etc.) >> >>> >> >>> You could then have the "ivr-answer" extension do whatever is >> >>> appropriate, like listen for digits, play announcement, beg for money, >> >>> etc. :) >> >>> >> >>> -MC >> >>> >> >>> > But for the life of me I can't figure out how to translate this into >> >>> > the xml >> >>> > response ... >> >>> > [campaign] >> >>> > exten => 100,1,ANSWER() >> >>> > exten => 100,n,WAIT(2) >> >>> > exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile) >> >>> > exten => 100,n,WAITEXTEN(10) >> >>> > exten => 100,n,HANGUP() >> >>> > exten => 1,1,PLAYBACK(goodbye) >> >>> > .... and so on ... >> >>> > I've looked at the ivr.conf stuff but it's all static and all of >> this >> >>> > has to >> >>> > be manageable via a web interface .... meaning dumping into a DB and >> >>> > returning an XML response seems reasonable ... but trying to stick >> or >> >>> > modify >> >>> > static text files from the web interface is too much text parsing >> and >> >>> > bad >> >>> > things will happen ... >> >>> > Any thoughts or pointing me in the right direction would be >> >>> > appreciated. >> >>> > Shelby >> >>> > >> >>> > >> >>> > >> >>> > _______________________________________________ >> >>> > Freeswitch-users mailing list >> >>> > Freeswitch-users at 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 at 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 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 >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at 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 at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/e48bf5e4/attachment-0001.html From pauld at versafon.com Fri Jan 23 16:56:18 2009 From: pauld at versafon.com (pauld) Date: Fri, 23 Jan 2009 19:56:18 -0500 Subject: [Freeswitch-users] bgapi uuid_kill question In-Reply-To: <191c3a030901231600v63bdbcf1w483980eaaf333a5b@mail.gmail.com> References: <1223981236.71232753768369.JavaMail.james@versafon31.versafon.com> <0569E2A1-83C8-444C-A812-AB300A64864B@freeswitch.org> <1748763617.81232755044367.JavaMail.james@versafon31.versafon.com> <191c3a030901231600v63bdbcf1w483980eaaf333a5b@mail.gmail.com> Message-ID: <254306201.91232758604251.JavaMail.james@versafon31.versafon.com> Seems like this works running FS under Linux, but didn't for Windows version. Let me double check that on Monday. Thanks a lot. Anthony Minessale wrote: > Unique-ID > > On Fri, Jan 23, 2009 at 5:56 PM, pauld > wrote: > > Oh, > But where do I get it from? It's not seem to be available from > CHANNEL_EXECUTE event or any other channel related event. > > > Brian West wrote: > > Don't use the core_uuid.. use the session uuid. > > > > /b > > > > On Jan 23, 2009, at 5:35 PM, pauld wrote: > > > > > >> Hi, > >> Trying to kill a channel (hangup) when it's in CHANNEL_EXECUTE > state > >> as > >> per mod_socket_event event information I use bgapi uuid_kill > >> > >> - but get response "ERR no such channel!". Same happens if I use > >> unique_id or channel_name > >> as an argument to uuid_kill. What I am doing wrong? > >> Help would much appreciated. > >> Thx. > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Fri Jan 23 17:25:28 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 23 Jan 2009 19:25:28 -0600 Subject: [Freeswitch-users] Wideband/Ultrawideband Conference In-Reply-To: <87f2f3b90901231646t379ba9a9xb1b738dd021407a6@mail.gmail.com> References: <38716ACB-BBED-4894-9FCE-543644141B84@freeswitch.org> <87f2f3b90901231646t379ba9a9xb1b738dd021407a6@mail.gmail.com> Message-ID: <07A74559-474A-4E7A-8C7A-88C45887995F@freeswitch.org> Also if you're calling with PortAudio update ASAP... /b On Jan 23, 2009, at 6:46 PM, Michael Collins wrote: > Also, we're more likely to answer your questions if you're not running > 8kHz audio! ;) > -MC From scott.ellis at novatex.com.au Fri Jan 23 18:02:06 2009 From: scott.ellis at novatex.com.au (Scott Ellis) Date: Sat, 24 Jan 2009 13:02:06 +1100 Subject: [Freeswitch-users] api_hangup_hook not actually executing the command In-Reply-To: <04D6A331-1E76-427F-B6D1-52B9553F1D9E@freeswitch.org> References: <497605DE.8010307@novatex.com.au> <04D6A331-1E76-427F-B6D1-52B9553F1D9E@freeswitch.org> Message-ID: <497A769E.6070700@novatex.com.au> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090124/5c2eee3f/attachment.html From Laurent.Fabre at kirranet.com Fri Jan 23 22:15:53 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Sat, 24 Jan 2009 07:15:53 +0100 Subject: [Freeswitch-users] Gateway Ping Message-ID: Hi, I activated the ping option on one of my gateway and I receive that kind of message : nta: sent OPTIONS (110269468) to udp/91.121.129.17:5060 nta: received 501 Not Implemented for OPTIONS (110269468) Does that mean the ping is working ? :) Regards, -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61<+33170247461> laurent.fabre at kirranet.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090124/2e68199f/attachment.html From Laurent.Fabre at kirranet.com Fri Jan 23 22:16:55 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Sat, 24 Jan 2009 07:16:55 +0100 Subject: [Freeswitch-users] Freeswitch latest revision ODBC crashed In-Reply-To: References: Message-ID: Updated to latest and no more crash. Thanks a lot. -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61 laurent.fabre at kirranet.com -----Message d'origine----- De?: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Laurent Fabre Envoy??: mercredi 21 janvier 2009 17:08 ??: freeswitch-users at lists.freeswitch.org Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed Been running all day without crashing. Will see on the long term. -- Laurent FABRE Directeur g?n?ral 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20 Mob: +33.(0)6.75.75.02.96 Fax: +33.(0)1.70.24.74.61 laurent.fabre at kirranet.com -----Message d'origine----- De?: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Ken Rice Envoy??: mercredi 21 janvier 2009 08:22 ??: freeswitch-users at lists.freeswitch.org Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed I'm not sure that patch will work... It works for me on an unrelated thing (mod_easyroute) so hopefully it will work for you here K > From: Laurent Fabre > Reply-To: > Date: Wed, 21 Jan 2009 08:27:08 +0100 > To: "freeswitch-users at lists.freeswitch.org" > > Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > Applied the patch, compiled, restarted. > > So far so good. Everything looks OK. Registrations, subscriptions, calls are > working as expected. > > I have an issue with the presence on one of the softphone (presence.wminfo?) > but it's probably unrelated and just a glitch in Bria for Outlook. > > Keep you posted. > > -- Laurent FABRE > Directeur g?n?ral > 10, rue d'Aumale > 75009 Paris > Tel: +33.(0)1.42.81.28.20 > Mob: +33.(0)6.75.75.02.96 > Fax: +33.(0)1.70.24.74.61 > laurent.fabre at kirranet.com > > > -----Message d'origine----- > De?: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Ken Rice > Envoy??: mercredi 21 janvier 2009 07:50 > ??: freeswitch-users at lists.freeswitch.org > Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed > > Can you try that attached patch? > > K > > >> From: Laurent Fabre >> Reply-To: >> Date: Wed, 21 Jan 2009 07:56:32 +0100 >> To: "freeswitch-users at lists.freeswitch.org" >> >> Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC >> crashed >> >> Hi Ken, >> >> Thanks for looking into it. >> >> It just crashed again on the voicemails table after running like a >> charm for 1 >> hour: http://pastebin.freeswitch.org/6832 >> >> If I can be of any assistance, please let me know. >> >> Regards, >> >> -- Laurent FABRE >> Directeur g?n?ral >> 10, rue d'Aumale >> 75009 Paris >> Tel: +33.(0)1.42.81.28.20 >> Mob: +33.(0)6.75.75.02.96 >> Fax: +33.(0)1.70.24.74.61 >> laurent.fabre at kirranet.com >> >> -----Message d'origine----- >> De?: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de >> Ken Rice Envoy??: mercredi 21 janvier 2009 06:59 ??: >> freeswitch-users at lists.freeswitch.org >> Objet?: Re: [Freeswitch-users] Freeswitch latest revision ODBC crashed >> >> I think I found the problem here... Not 100%... But I did find an >> issue in switch_odbc.c... >> >> I'll have to talk to Tony about this one... There is a condition where >> we can return an error and we shouldn't >> >> >>> From: Michael Collins >>> Reply-To: >>> Date: Tue, 20 Jan 2009 21:12:30 -0800 >>> To: >>> Subject: Re: [Freeswitch-users] Freeswitch latest revision ODBC >>> crashed >>> >>> Merci beaucoup! The devs will keep an eye on this and will hopefully >>> have a resolution soon. >>> -MC >>> >>> On Tue, Jan 20, 2009 at 8:45 PM, Laurent Fabre >>> wrote: >>>> http://jira.freeswitch.org/browse/FSCORE-276 >>>> >>>> >>>> >>>> Done. >>>> >>>> >>>> >>>> -- Laurent FABRE >>>> Directeur g?n?ral >>>> 10, rue d'Aumale >>>> 75009 Paris >>>> Tel: +33.(0)1.42.81.28.20 >>>> Mob: +33.(0)6.75.75.02.96 >>>> Fax: +33.(0)1.70.24.74.61 >>>> laurent.fabre at kirranet.com >>>> >>>> >>>> >>>> De : freeswitch-users-bounces at lists.freeswitch.org >>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de >>>> Brian West Envoy? : mercredi 21 janvier 2009 04:56 ? : >>>> freeswitch-users at lists.freeswitch.org >>>> Objet : Re: [Freeswitch-users] Freeswitch latest revision ODBC >>>> crashed >>>> >>>> >>>> >>>> Please open a jira http://jira.freeswitch.org include all the >>>> requested data in detail. >>>> >>>> >>>> >>>> /b >>>> >>>> >>>> >>>> On Jan 20, 2009, at 9:57 PM, Laurent Fabre wrote: >>>> >>>> Hi, >>>> >>>> >>>> >>>> I'm experiencing a few ? random ? segfaults ever since I enabled >>>> core ODBC support. They usually happen during startup or quickly >>>> after?or not at all as long as I do not restart freeswitch. >>>> >>>> >>>> >>>> The backend is a postgresql 7.4, every module has a distinct >>>> database (LATIN9, with OIDS). >>>> >>>> >>>> >>>> I'm having a few issues with presence & call routing which tend to >>>> break from time to time. The freeswitch box is hosted and I've got >>>> several phones with STUN enabled behind a NAT firewall (namely >>>> pfSense). I'm not sure how those things might relate but maybe it's worth >>>> mentionning. >>>> >>>> >>>> >>>> I've decided to update a few hours ago but still crashing. >>>> >>>> >>>> >>>> I was about to test mod_easyroute when I decided to run gdb and look >>>> into it : http://pastebin.freeswitch.org/6830 >>>> >>>> >>>> >>>> Could someone please help me pointing the issue? J >>>> >>>> >>>> >>>> Thanks in advance, >>>> >>>> >>>> >>>> -- Laurent FABRE >>>> Directeur g?n?ral >>>> 10, rue d'Aumale >>>> 75009 Paris >>>> Tel: +33.(0)1.42.81.28.20 >>>> Mob: +33.(0)6.75.75.02.96 >>>> Fax: +33.(0)1.70.24.74.61 >>>> laurent.fabre at kirranet.com >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>>> sers >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>>> sers >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> ers >>> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> _______________________________________________ > Freeswitch-users mailing >> list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman >> /listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >> ions/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pmhshz at gmail.com Sat Jan 24 03:59:55 2009 From: pmhshz at gmail.com (shehzad p) Date: Sat, 24 Jan 2009 03:59:55 -0800 (PST) Subject: [Freeswitch-users] How to bridge without Answer? In-Reply-To: <191c3a030901220630y6401d13ene8302c6cf840d61a@mail.gmail.com> References: <21583334.post@talk.nabble.com> <191c3a030901210556n2d443179n17d8bbb9ed24b8ab@mail.gmail.com> <21602715.post@talk.nabble.com> <49786A24.8010300@3c.co.uk> <191c3a030901220630y6401d13ene8302c6cf840d61a@mail.gmail.com> Message-ID: <21639876.post@talk.nabble.com> Hi Anthony, Where I can apply such patch in code to suppress answer. Thanks for response. Anthony Minessale-2 wrote: > > Daivd, > > I think you missed part of his question. > > You can easily choose not to answer an inbound call in FS by never > explicitly answering it. > you can call pre_answer instead or if you send the call to an app that > requires media it's pre_answered automatically. > > pre_answer in FS terms is early media. > > You can run an ivr for instance completely in early_media (assuming the > telco allows dtmf during early media) > > He asked if we can *bridge* the call outward to another endpoint and not > pass the answer across when the far end answers. > > The bridge application is designed to bridge calls and the standard > behavior > when forwarding a call would be > once the far end answers, pass the answer indication down the line. > > We do not currently have provision for supressing the answer as I stated. > It would require a patch. > > > > > > On Thu, Jan 22, 2009 at 6:44 AM, David Knell wrote: > >> There's a whole bunch of reasons why you might not want to answer an >> inbound call: >> - intercept messages (e.g. "the cellphone you've called is switched off") >> - cost reduction on 1-800 calls, although you won't get a forward audio >> path from the >> caller until you do answer it >> - in one case, a company for whom I'd provided some IVR (back in the >> 1990s) >> had >> someone mail out some tens of thousands of cards with "You owe us X - you >> must call >> this (900) number now to avoid court proceedings" on; we were able to not >> answer the >> inbound leg of the call, but still play a recorded message to the caller >> informing them >> that they could just ignore it. Had we had to answer the inbound leg, >> they'd have been >> charged. >> >> --Dave >> >> Thanks Anthony, >> >> There are some toll-free numbers I need to configure such that, >> originator >> does not need to charge to its users, even though they are answered on >> terminator side. >> >> >> >> >> Anthony Minessale-2 wrote: >> >> >> 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 >> 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 >> listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthmMSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conferencesip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing >> listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> >> -- >> David Knell, Director, 3C Limited >> T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623http://www.3c.co.uk >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/How-to-bridge-without-Answer--tp21583334p21639876.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From helmut.kuper at ewetel.de Sat Jan 24 04:11:15 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Sat, 24 Jan 2009 13:11:15 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> Message-ID: <497B0563.9090701@ewetel.de> Hello, to keep you informed. The stand alone c code is fully working now. wireshark can decode and display the generated packets containing Q931 hex dump samples I copied from FS log into the c code. So now I will start to put it into openzap/mod_openzap. regards helmut From testeador01 at gmail.com Sat Jan 24 05:22:12 2009 From: testeador01 at gmail.com (Milena) Date: Sat, 24 Jan 2009 08:22:12 -0500 Subject: [Freeswitch-users] My fs_cli stopped working, can you help me please? Message-ID: Well, telnet works, what else could it be? another thing i remember doing before the fs_cli stopped working was installing TCAPI and getting it to work, but i didn't change any config files or anything other than what it says in the instalation instructions; I don't mind so much about replicating the issue and figuring out how did it get damaged, what I really want to know now is how to make it work again cause even reconfiguring/installing fs doesn't fix it and the truth is, the fs_cli is comfortable; any idea on how to fix it other than starting a fresh install of fs from a new os instalation? Thank you! My telnet output: -bash-3.2# telnet localhost 8021 Trying 127.0.0.1... Connected to localhost.localdomain (127.0.0.1). Escape character is '^]'. Content-Type: auth/request auth Content-Type: command/reply Reply-Text: +OK accepted exit Content-Type: command/reply Reply-Text: +OK bye Content-Type: text/disconnect-notice Content-Length: 70 Disconnected, goodbye! See you at ClueCon http://www.cluecon.com/ !!! Connection closed by foreign host. ------- *Michael Collins wrote:* just for kicks, can you try a raw telnet session, just to make sure the the event socket is working properly? telnet localhost 8021 auth ClueCon (press enter twice) If you get something like this: h-3.2# telnet localhost 8021 Trying ::1... telnet: connect to address ::1: Connection refused Trying fe80::1... telnet: connect to address fe80::1: Connection refused Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Content-Type: auth/request auth ClueCon Content-Type: command/reply Reply-Text: +OK accepted Then the event socket stuff is working. To exit out, type "exit" then enter twice. Let us know what happens. -MC On Fri, Jan 23, 2009 at 2:14 PM, Milena > wrote: >* *>* I use the stable version of fs 1.0.2 testing out different options of *>* configuration; *>* Last thing I was doing was trying to fix some issues with dingaling-google *>* talk cause i hear no audio at all from an external ip, and i still didnt get *>* it to work; *>* *>* so i tried deleting the sofia_* files on the folder db/ as suggested on *>* another thread about nat related issues (oops?), and then restarting fs. *>* After this, trying to execute bin/fs_cli doesn't work anymore; it tries to *>* connect cause i get the output on the box: *>* 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1791 listener_run() *>* Connection Open from 127.0.0.1:33055 *>* 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1914 listener_run() Session *>* complete, waiting for children *>* 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1938 listener_run() *>* Connection Closed *>* *>* but on the console where im trying to run the cli nothing happens, *>* I tried rebooting the machine; i tried with freeswitch make current, it *>* didn't fix it; i tried all the way from configure to make and make install *>* and it didn't fix my problem either. *>* *>* What should i run or change on the configurations to fix this? *>* *>* Thank you very much. *>* *>* _______________________________________________ *>* Freeswitch-users mailing list *>* Freeswitch-users at lists.freeswitch.org *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users *>* http://www.freeswitch.org *>* *>* * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090124/fbfc35f9/attachment.html From pmhshz at gmail.com Sat Jan 24 05:47:19 2009 From: pmhshz at gmail.com (shehzad p) Date: Sat, 24 Jan 2009 05:47:19 -0800 (PST) Subject: [Freeswitch-users] ATA-answering machine question/recommendation In-Reply-To: <1461348542-1232566409-cardhu_decombobulator_blackberry.rim.net-569820458-@bxe161.bisx.prod.on.blackberry> References: <207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45@mail.gmail.com> <1461348542-1232566409-cardhu_decombobulator_blackberry.rim.net-569820458-@bxe161.bisx.prod.on.blackberry> Message-ID: <21640813.post@talk.nabble.com> Hi all, On my existing Freeswitch 1.0.2, I installed and configured mod_vmd as below: make mod_vmd-install Then configured it as on wiki page: http://wiki.freeswitch.org/wiki/Mod_vmd After that my dialplan terminates call to another system, where it is just answered and wait for some time there. So that there should be a variable called vmd_detect must be created as shown in dialplan http://wiki.freeswitch.org/wiki/Mod_vmd#From_Dialplan. But eventhough no variable named 'vmd_detect' is created after that.!!! Is there i am missing something? Is there another way of using mod_vmd? Thanks in advance. msp Lucas Cornelisse wrote: > > Hi Jonathan, > > Mod_vmd (voicemail detection) should do the trick. > > Just search the wiki for mod_vmd, there are a number of ways of using it. > > > Sent from my BlackBerry device on the Rogers Wireless Network > > -----Original Message----- > From: jonathan augenstine > > Date: Wed, 21 Jan 2009 06:53:41 > To: > Subject: [Freeswitch-users] ATA-answering machine question/recommendation > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/ATA-answering-machine-question-recommendation-tp21584795p21640813.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Sat Jan 24 06:06:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 24 Jan 2009 08:06:02 -0600 Subject: [Freeswitch-users] How to bridge without Answer? In-Reply-To: <21639876.post@talk.nabble.com> References: <21583334.post@talk.nabble.com> <191c3a030901210556n2d443179n17d8bbb9ed24b8ab@mail.gmail.com> <21602715.post@talk.nabble.com> <49786A24.8010300@3c.co.uk> <191c3a030901220630y6401d13ene8302c6cf840d61a@mail.gmail.com> <21639876.post@talk.nabble.com> Message-ID: <191c3a030901240606v1d329596y68928365a8c2f3b1@mail.gmail.com> off the top of my head I would say in switch_ivr_originate.c and switch_ivr_bridge.c On Sat, Jan 24, 2009 at 5:59 AM, shehzad p wrote: > > Hi Anthony, > > Where I can apply such patch in code to suppress answer. > > Thanks for response. > > > Anthony Minessale-2 wrote: > > > > Daivd, > > > > I think you missed part of his question. > > > > You can easily choose not to answer an inbound call in FS by never > > explicitly answering it. > > you can call pre_answer instead or if you send the call to an app that > > requires media it's pre_answered automatically. > > > > pre_answer in FS terms is early media. > > > > You can run an ivr for instance completely in early_media (assuming the > > telco allows dtmf during early media) > > > > He asked if we can *bridge* the call outward to another endpoint and not > > pass the answer across when the far end answers. > > > > The bridge application is designed to bridge calls and the standard > > behavior > > when forwarding a call would be > > once the far end answers, pass the answer indication down the line. > > > > We do not currently have provision for supressing the answer as I stated. > > It would require a patch. > > > > > > > > > > > > On Thu, Jan 22, 2009 at 6:44 AM, David Knell wrote: > > > >> There's a whole bunch of reasons why you might not want to answer an > >> inbound call: > >> - intercept messages (e.g. "the cellphone you've called is switched > off") > >> - cost reduction on 1-800 calls, although you won't get a forward audio > >> path from the > >> caller until you do answer it > >> - in one case, a company for whom I'd provided some IVR (back in the > >> 1990s) > >> had > >> someone mail out some tens of thousands of cards with "You owe us X - > you > >> must call > >> this (900) number now to avoid court proceedings" on; we were able to > not > >> answer the > >> inbound leg of the call, but still play a recorded message to the caller > >> informing them > >> that they could just ignore it. Had we had to answer the inbound leg, > >> they'd have been > >> charged. > >> > >> --Dave > >> > >> Thanks Anthony, > >> > >> There are some toll-free numbers I need to configure such that, > >> originator > >> does not need to charge to its users, even though they are answered on > >> terminator side. > >> > >> > >> > >> > >> Anthony Minessale-2 wrote: > >> > >> > >> 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 > >> 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 > >> listFreeswitch-users at lists.freeswitch.orghttp:// > lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> > >> AIM: anthmMSN:anthony_minessale at hotmail.com > >> > > > >> > >GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >> > > > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conferencesip:888 at conference.freeswitch.org > >> > > > >> > > > iax:guest at conference.freeswitch.org/888googletalk:conf+888 at conference.freeswitch.org > > > > >> > > > >> pstn:213-799-1400 > >> > >> _______________________________________________ > >> Freeswitch-users mailing > >> listFreeswitch-users at lists.freeswitch.orghttp:// > lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> David Knell, Director, 3C Limited > >> T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623http://www.3c.co.uk > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at hotmail.com < > MSN%3Aanthony_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 < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/How-to-bridge-without-Answer--tp21583334p21639876.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090124/0b1ec885/attachment-0001.html From anthony.minessale at gmail.com Sat Jan 24 06:10:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 24 Jan 2009 08:10:22 -0600 Subject: [Freeswitch-users] My fs_cli stopped working, can you help me please? In-Reply-To: References: Message-ID: <191c3a030901240610x343b7dadsbbd5c3411b79ec40@mail.gmail.com> try rebuilding fs_cli cd libs/esl make clean make try to run ./fs_cli if it works copy that into your path On Sat, Jan 24, 2009 at 7:22 AM, Milena wrote: > Well, telnet works, what else could it be? > > another thing i remember doing before the fs_cli stopped working was > installing TCAPI and getting it to work, but i didn't change any config > files or anything other than what it says in the instalation instructions; > > I don't mind so much about replicating the issue and figuring out how did > it get damaged, what I really want to know now is how to make it work again > cause even reconfiguring/installing fs doesn't fix it and the truth is, the > fs_cli is comfortable; any idea on how to fix it other than starting a fresh > install of fs from a new os instalation? > > Thank you! > > My telnet output: > -bash-3.2# telnet localhost 8021 > Trying 127.0.0.1... > Connected to localhost.localdomain (127.0.0.1). > Escape character is '^]'. > Content-Type: auth/request > > auth > > Content-Type: command/reply > Reply-Text: +OK accepted > > exit > > Content-Type: command/reply > Reply-Text: +OK bye > > Content-Type: text/disconnect-notice > Content-Length: 70 > > Disconnected, goodbye! > See you at ClueCon http://www.cluecon.com/ !!! > Connection closed by foreign host. > > ------- > > *Michael Collins wrote:* > > just for kicks, can you try a raw telnet session, just to make sure > > the the event socket is working properly? > telnet localhost 8021 > auth ClueCon (press enter twice) > > If you get something like this: > h-3.2# telnet localhost 8021 > Trying ::1... > telnet: connect to address ::1: Connection refused > > Trying fe80::1... > telnet: connect to address fe80::1: Connection refused > Trying 127.0.0.1... > Connected to localhost. > Escape character is '^]'. > Content-Type: auth/request > > auth ClueCon > > Content-Type: command/reply > > Reply-Text: +OK accepted > > Then the event socket stuff is working. To exit out, type "exit" then > enter twice. Let us know what happens. > -MC > > On Fri, Jan 23, 2009 at 2:14 PM, Milena > wrote: > > >* > *>* I use the stable version of fs 1.0.2 testing out different options of > *>* configuration; > *>* Last thing I was doing was trying to fix some issues with dingaling-google > *>* talk cause i hear no audio at all from an external ip, and i still didnt get > *>* it to work; > *>* > *>* so i tried deleting the sofia_* files on the folder db/ as suggested on > *>* another thread about nat related issues (oops?), and then restarting fs. > *>* After this, trying to execute bin/fs_cli doesn't work anymore; it tries to > *>* connect cause i get the output on the box: > *>* 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1791 listener_run() > *>* Connection Open from 127.0.0.1:33055 > *>* 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1914 listener_run() Session > *>* complete, waiting for children > *>* 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1938 listener_run() > *>* Connection Closed > *>* > *>* but on the console where im trying to run the cli nothing happens, > *>* I tried rebooting the machine; i tried with freeswitch make current, it > *>* didn't fix it; i tried all the way from configure to make and make install > *>* and it didn't fix my problem either. > *>* > *>* What should i run or change on the configurations to fix this? > *>* > *>* Thank you very much. > *>* > *>* _______________________________________________ > *>* Freeswitch-users mailing list > * > >* Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090124/2ba48207/attachment.html From testeador01 at gmail.com Sat Jan 24 07:16:41 2009 From: testeador01 at gmail.com (Milena) Date: Sat, 24 Jan 2009 10:16:41 -0500 Subject: [Freeswitch-users] My fs_cli stopped working, can you help me please? Message-ID: Hi, thanks for the reply, I tried it both with fs down and it didn't work, also when it was turned on, no results either. Any other ideas please? i really wanna fix it without having to start from zero. thank you very much On the Freeswitch console i keep getting the message: 2009-01-24 10:03:23 [DEBUG] mod_event_socket.c:1791 listener_run() Connection Open from 127.0.0.1:54741 2009-01-24 10:03:23 [DEBUG] mod_event_socket.c:1914 listener_run() Session complete, waiting for children 2009-01-24 10:03:23 [DEBUG] mod_event_socket.c:1938 listener_run() Connection Closed This is what i did in the other console: -bash-3.2# cd /usr/src/freeswitch-1.0.2/libs/esl/ -bash-3.2# make clean rm -f *.o src/*.o testclient testserver fs_cli libesl.a *~ src/*~ src/include/*~ -bash-3.2# make cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o src/esl_event.o cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o src/esl_threadmutex.o cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o src/esl_config.o ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o src/esl_config.o ranlib libesl.a cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L. -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient -L. -lncurses -lpthread -lesl cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver -L. -lncurses -lpthread -lesl -bash-3.2# ./fs_cli -bash-3.2# -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090124/c49ba5e3/attachment.html From anthony.minessale at gmail.com Sat Jan 24 09:24:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 24 Jan 2009 11:24:19 -0600 Subject: [Freeswitch-users] My fs_cli stopped working, can you help me please? In-Reply-To: References: Message-ID: <191c3a030901240924q3c5329fdg6beb156bbfa6b851@mail.gmail.com> Try running fs_cli -d 7 Maybe the debug log will shed some light. On Sat, Jan 24, 2009 at 9:16 AM, Milena wrote: > Hi, thanks for the reply, > > I tried it both with fs down and it didn't work, also when it was turned > on, no results either. > Any other ideas please? i really wanna fix it without having to start from > zero. > > thank you very much > > On the Freeswitch console i keep getting the message: > 2009-01-24 10:03:23 [DEBUG] mod_event_socket.c:1791 listener_run() > Connection Open from 127.0.0.1:54741 > 2009-01-24 10:03:23 [DEBUG] mod_event_socket.c:1914 listener_run() Session > complete, waiting for children > 2009-01-24 10:03:23 [DEBUG] mod_event_socket.c:1938 listener_run() > Connection Closed > > This is what i did in the other console: > -bash-3.2# cd /usr/src/freeswitch-1.0.2/libs/esl/ > -bash-3.2# make clean > rm -f *.o src/*.o testclient testserver fs_cli libesl.a *~ src/*~ > src/include/*~ > -bash-3.2# make > cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings > -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o > cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings > -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o > src/esl_event.o > cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings > -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o > src/esl_threadmutex.o > cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings > -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o > src/esl_config.o > ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o > src/esl_config.o > ranlib libesl.a > cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings > -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L. > -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit > cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings > -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient -L. > -lncurses -lpthread -lesl > cc -Isrc/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings > -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver -L. > -lncurses -lpthread -lesl > -bash-3.2# ./fs_cli > -bash-3.2# > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090124/b8b66029/attachment-0001.html From asannucci at gmail.com Sat Jan 24 08:55:44 2009 From: asannucci at gmail.com (Andrea) Date: Sat, 24 Jan 2009 11:55:44 -0500 Subject: [Freeswitch-users] Problem with NAT References: <35b355e90901231404lfb937b5x1152b4703ce88130@mail.gmail.com> Message-ID: <8744BCB0A856494DBAE8E45C6D6A8195@quos> A very simple (maybe) question. I have freeswitch installed behind a firewall with needs ports open I have users internal (connected to local lan address) and users external (connected to my public internet IP address) All users are configured in the directory/default. When I call from a lan user a external user all work fine When I call from a external user a lan user the call drop after 30-32 seconds When I call from a external user other external user the call drop after 30-32 seconds When I call from a lan user other lan user all work fine I thing is a NAT problem but I don't know how solve it Regards - Andrea - -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090124/04bf1cdf/attachment.html From mike at jerris.com Sat Jan 24 12:57:16 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 24 Jan 2009 15:57:16 -0500 Subject: [Freeswitch-users] Gateway Ping In-Reply-To: References: Message-ID: Yes. Mike On Jan 24, 2009, at 1:15 AM, Laurent Fabre wrote: > Hi, > > > > I activated the ping option on one of my gateway and I receive that > kind of message : > > > > nta: sent OPTIONS (110269468) to udp/91.121.129.17:5060 > > nta: received 501 Not Implemented for OPTIONS (110269468) > > > > Does that mean the ping is working ? J > > > > Regards, > > > > -- Laurent FABRE > Directeur g?n?ral > 10, rue d'Aumale > 75009 Paris > Tel: +33.(0)1.42.81.28.20 > Mob: +33.(0)6.75.75.02.96 > Fax: +33.(0)1.70.24.74.61 > laurent.fabre at kirranet.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090124/f14b84a5/attachment.html From jmesquita at gmail.com Sat Jan 24 18:59:26 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 25 Jan 2009 00:59:26 -0200 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org> <4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br> <87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> Message-ID: We have very high taxes on hardware imports in Brazil, so software is not a problem on the equation. Cisco's routers for example are 3 times more expensive here then in the US (even the cheap ones). If the price of the equipment is compared to the price of the G729 license, the latter becomes quite insignificant. And bw has poor quality and high prices. The same rule applies to all Latin America. Hope I have illustrated the picture correctly. JMesquita On Jan 23, 2009, at 6:29 PM, Michael Collins wrote: > On Fri, Jan 23, 2009 at 12:16 PM, Rodrigo P. Telles > wrote: >> Hi Dave, >> >> Down here in Brazil, the bandwidth costs is very high (around U$ >> 400.00/Mb) so it should be valid only for a "non" third >> world country. >> G729 and G723.1 is almost a law here, if you don't play at least >> with G729 your ITSP is out of mark share! >> >> My 2 cents from a third world country. > > What is the patent and licensing situation in Brazil? Those are also > factors. $10/port might be cheap in the US but in Brazil it could be > much more? (I'm asking...) > -MC > >> >> Regards, >> Rodrigo Telles >> >> Em 23-01-2009 03:09, David Knell escreveu: >>> Steve Underwood wrote: >>>> Depends what you are after. Speex offers the quality of G.729 at >>>> around >>>> the same processing load. However, nobody seems to want to pay >>>> for the >>>> processing load of G.729. Almost everything uses G.729A. Half the >>>> processing load, but significantly poorer quality. >>>> >>>> VoIP is mostly a race to the bottom, and people wonder why it >>>> makes no >>>> money for provides. :-\ >>>> >>> And, at the wholesale level, it makes no sense whatsoever to >>> compress calls >>> any more: bandwidth is so cheap (and has been for a while) that >>> the loss in >>> call quality - especially from tandem compressions - and the >>> increased >>> processing requirements and other bits of expense do not stack >>> up. Case in >>> point: we moved a route from G.711 to G.729, and saw the ACD drop >>> from >>> over 10 to under 7 minutes. It was a route to mobiles, so the >>> audio was >>> being >>> recompressed with the GSM codec on its way to the handsets. >>> Economically, >>> had we carried on using G.729, we'd have lost about 30% of our >>> margin on >>> that route. >>> >>> --Dave >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mszlazak at aol.com Sun Jan 25 00:31:43 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sun, 25 Jan 2009 03:31:43 -0500 Subject: [Freeswitch-users] OBDC.numRows() always returning -1 when records returned from db. Message-ID: <8CB4CDE7CE0D5B2-11BC-24BD@webmail-me03.sysops.aol.com> I haven't seen obdc.numRows() come up in this list but I can't get anything but -1 even though I get records from the database. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090125/d2e56d3b/attachment.html From jason at jasonjgw.net Sun Jan 25 01:24:23 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 25 Jan 2009 20:24:23 +1100 Subject: [Freeswitch-users] PortAudio not closing audio device properly after hangup? Message-ID: <20090125092423.GA22684@jdc.jasonjgw.net> Under revision 11484, if I make a call using PortAudio: pa call then hang up the call: pa hangup and after that, try to access the same audio device with other software, I get a "device or resource busy" error. This is using Alsa under Debian Sid, kernel 2.6.26. Shutting down FreeSWITCH fixes this. This is also a regression compared with the earlier version of FreeSWITCH that I was running before doing the latest build. It seems to me that the device isn't being closed as it should be when the last call hangs up. From brian at freeswitch.org Sun Jan 25 01:54:29 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 25 Jan 2009 03:54:29 -0600 Subject: [Freeswitch-users] PortAudio not closing audio device properly after hangup? In-Reply-To: <20090125092423.GA22684@jdc.jasonjgw.net> References: <20090125092423.GA22684@jdc.jasonjgw.net> Message-ID: <8BD2BD9C-6075-48C8-A620-89B213176BCF@freeswitch.org> Thats really odd since we never close the audio device anymore. Can you open a jira with the proper debug info so I can take a look at it? Please assign it to "brian" when reporting. http://jira.freeswitch.org Include debug log, double check you're on svn trunk... and then collect the info and attach it to the jira. Please don't paste logs inline its harder to read ;) /b On Jan 25, 2009, at 3:24 AM, Jason White wrote: > Under revision 11484, if I make a call using PortAudio: > pa call > then hang up the call: > pa hangup > and after that, try to access the same audio device with other > software, I get > a "device or resource busy" error. > > This is using Alsa under Debian Sid, kernel 2.6.26. > > Shutting down FreeSWITCH fixes this. > > This is also a regression compared with the earlier version of > FreeSWITCH that > I was running before doing the latest build. > > It seems to me that the device isn't being closed as it should be > when the > last call hangs up. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090125/376982cf/attachment.html From jason at jasonjgw.net Sun Jan 25 02:02:44 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 25 Jan 2009 21:02:44 +1100 Subject: [Freeswitch-users] PortAudio not closing audio device properly after hangup? In-Reply-To: <8BD2BD9C-6075-48C8-A620-89B213176BCF@freeswitch.org> References: <20090125092423.GA22684@jdc.jasonjgw.net> <8BD2BD9C-6075-48C8-A620-89B213176BCF@freeswitch.org> Message-ID: <20090125100244.GA23456@jdc.jasonjgw.net> Brian West wrote: > Thats really odd since we never close the audio device anymore. Maybe we're misunderstanding each other here. Not closing the audio device anymore is the bug, because trying to access that device with another audio program (e.g., a sound file player outside FreeSWITCH) fails, even after all FreeSWITCH calls have hung up, because FreeSWITCH is still holding the audio device open. Would it be possible to have a configuration parameter that says, in effect, close the audio device when it isn't being used? From brian at freeswitch.org Sun Jan 25 02:10:18 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 25 Jan 2009 04:10:18 -0600 Subject: [Freeswitch-users] PortAudio not closing audio device properly after hangup? In-Reply-To: <20090125100244.GA23456@jdc.jasonjgw.net> References: <20090125092423.GA22684@jdc.jasonjgw.net> <8BD2BD9C-6075-48C8-A620-89B213176BCF@freeswitch.org> <20090125100244.GA23456@jdc.jasonjgw.net> Message-ID: <4346C468-8A78-43CD-AB07-81ABCFC23802@freeswitch.org> Oh so its a problem with your linux distro and how it setups up the sound card... aren't there things for Linux that allow multiple access to the sound hardware? ESD? /b On Jan 25, 2009, at 4:02 AM, Jason White wrote: > Maybe we're misunderstanding each other here. Not closing the audio > device > anymore is the bug, because trying to access that device with > another audio > program (e.g., a sound file player outside FreeSWITCH) fails, even > after all > FreeSWITCH calls have hung up, because FreeSWITCH is still holding > the audio > device open. > > Would it be possible to have a configuration parameter that says, in > effect, > close the audio device when it isn't being used? From jason at jasonjgw.net Sun Jan 25 02:19:13 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 25 Jan 2009 21:19:13 +1100 Subject: [Freeswitch-users] PortAudio not closing audio device properly after hangup? In-Reply-To: <4346C468-8A78-43CD-AB07-81ABCFC23802@freeswitch.org> References: <20090125092423.GA22684@jdc.jasonjgw.net> <8BD2BD9C-6075-48C8-A620-89B213176BCF@freeswitch.org> <20090125100244.GA23456@jdc.jasonjgw.net> <4346C468-8A78-43CD-AB07-81ABCFC23802@freeswitch.org> Message-ID: <20090125101913.GA23605@jdc.jasonjgw.net> Brian West wrote: > Oh so its a problem with your linux distro and how it setups up the > sound card... aren't there things for Linux that allow multiple access > to the sound hardware? ESD? It's supposed to work by default in recent versions of Alsa. I'll investigate why it isn't working in this case. From ronmccar at gmail.com Sun Jan 25 10:00:57 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Sun, 25 Jan 2009 11:00:57 -0700 Subject: [Freeswitch-users] Can I have FS send a 503? Message-ID: <3885f4fe0901251000p608273adgbd6d12df0b029ff8@mail.gmail.com> Hi, Is it possible to have FS send a 503 / no route found error back? I have looked and have not seen a place where you can add a custom sip response code like this. I want to return a 503 if they can't call that destination, instead of a 404 basically. Any help would be great, thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090125/327fee47/attachment.html From anthony.minessale at gmail.com Sun Jan 25 10:13:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 25 Jan 2009 12:13:45 -0600 Subject: [Freeswitch-users] Can I have FS send a 503? In-Reply-To: <191c3a030901251010x52b0f4b8k8b5d07ea6a083dfc@mail.gmail.com> References: <3885f4fe0901251000p608273adgbd6d12df0b029ff8@mail.gmail.com> <191c3a030901251010x52b0f4b8k8b5d07ea6a083dfc@mail.gmail.com> Message-ID: <191c3a030901251013i785b759al992f49a02efce59@mail.gmail.com> We use the standard rfc q.850 to sip cause code map so you can set continue_on_fail=true and hangup with the q.850 that maps to 503 (I forgot what it is atm) On Jan 25, 2009 12:09 PM, "Ron McCarthy" wrote: Hi, Is it possible to have FS send a 503 / no route found error back? I have looked and have not seen a place where you can add a custom sip response code like this. I want to return a 503 if they can't call that destination, instead of a 404 basically. Any help would be great, thanks! _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090125/fe5091c2/attachment.html From ronmccar at gmail.com Sun Jan 25 11:46:29 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Sun, 25 Jan 2009 12:46:29 -0700 Subject: [Freeswitch-users] Can I have FS send a 503? In-Reply-To: <191c3a030901251013i785b759al992f49a02efce59@mail.gmail.com> References: <3885f4fe0901251000p608273adgbd6d12df0b029ff8@mail.gmail.com> <191c3a030901251010x52b0f4b8k8b5d07ea6a083dfc@mail.gmail.com> <191c3a030901251013i785b759al992f49a02efce59@mail.gmail.com> Message-ID: <3885f4fe0901251146k78eb71e6u28cba840b8d2028b@mail.gmail.com> I actaully just used respond with a 503 and that sends back a "Service unavailable". If I do: I get a busy back on the other side, it looks like application="respond" actaully works better in this case. Thanks for the help! On Sun, Jan 25, 2009 at 11:13 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > We use the standard rfc q.850 to sip cause code map so you can set > continue_on_fail=true and hangup with the q.850 that maps to 503 (I forgot > what it is atm) > > On Jan 25, 2009 12:09 PM, "Ron McCarthy" wrote: > > Hi, > > Is it possible to have FS send a 503 / no route found error back? I have > looked and have not seen a place where you can add a custom sip response > code like this. I want to return a 503 if they can't call that destination, > instead of a 404 basically. > > Any help would be great, thanks! > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090125/0092f8b9/attachment.html From anthony.minessale at gmail.com Sun Jan 25 13:15:49 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 25 Jan 2009 15:15:49 -0600 Subject: [Freeswitch-users] PortAudio not closing audio device properly after hangup? In-Reply-To: <20090125101913.GA23605@jdc.jasonjgw.net> References: <20090125092423.GA22684@jdc.jasonjgw.net> <8BD2BD9C-6075-48C8-A620-89B213176BCF@freeswitch.org> <20090125100244.GA23456@jdc.jasonjgw.net> <4346C468-8A78-43CD-AB07-81ABCFC23802@freeswitch.org> <20090125101913.GA23605@jdc.jasonjgw.net> Message-ID: <191c3a030901251315t308887adgc3f774c23f635ade@mail.gmail.com> I'll try another approach. try latest trunk and see how that works? On Sun, Jan 25, 2009 at 4:19 AM, Jason White wrote: > Brian West wrote: > > Oh so its a problem with your linux distro and how it setups up the > > sound card... aren't there things for Linux that allow multiple access > > to the sound hardware? ESD? > > It's supposed to work by default in recent versions of Alsa. > > I'll investigate why it isn't working in this case. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090125/3a9f2bd5/attachment.html From gcd at i.ph Sun Jan 25 00:21:45 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 25 Jan 2009 16:21:45 +0800 Subject: [Freeswitch-users] TDM400 FXO can dialout only once Message-ID: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> hi everybody, i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working using IP phones, softphones and digium FXS port. but there's a problem in dialing out to PSTN using digium tdm400 fxo - it works fine on the first attempt (after starting FS) but it fails on the subsequent attempts. i tested to call using the FXS port and IP phone. same problem. before i place any call, i checked >oz dump 2 1 (show current state = DOWN, last state = DOWN) in the first call, there's this message: [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for channel 2:1 but then i hangup. checked >oz dump 21 (show current state=DOWN, last state=HANGUP) in the 2nd (and subsequent) attempts, the fxo just goes off-hook but doesn't send the dtmf tones. >oz dump 2 1 (shows current state = DIALING, last state = DOWN) has anyone encountered this problem before? i appreciate for any help to correct this problem. tks, nandy Environment: ================== kernel 2.6.18-92.1.22.el5 FS 1.0.2 zaptel 1.4.11 oslec digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) zaptel.conf ======== loadzone = us defaultzone=us channels=1-2 alaw=1-4 fxsks=2 fxoks=1 openzap.conf.xml: =============== openzap.conf ========== [span zt] name => OpenZAP FXS number => 1 fxs-channel => 1 [span zt] name => OpenZAP FXO number => 2 fxo-channel => 2 tones.conf (the dialtone and ring tone is set to Philipping tones) ======== [us] generate-dial => v=-7;%(1000,0,425) detect-dial => 425 generate-ring => v=-7;%(1000,4000,425,480) detect-ring => 425,480 generate-busy => v=-7;%(500,500,480,620) detect-busy => 480,620 generate-attn => v=0;%(200,300,1400,1800) detect-attn => 1400,1800 generate-callwaiting-sas => v=0;%(300,10000,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 776.7 LOG OF FIRST CALL (OK) ==================== 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute bridge(openzap/2/1/3400534) 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU 20ms 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 channel_outgoing_channel() Connect outbound channel OpenZAP/2:1/3400534 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/2:1/3400534 [e5f12114-ea88-11dd-9f5c-290fb4a527a4] 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 channel_outgoing_channel() (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call() Changing state on 2:1 from DOWN to DIALING 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() ANALOG CHANNEL thread starting. 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_INIT 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for DIALING 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (OpenZAP/2:1/3400534) State INIT 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to sleep 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_ROUTING 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() OpenZAP/2:1/3400534 CHANNEL ROUTING 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to sleep 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_CONSUME_MEDIA 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 zap_analog_channel_run() Detected tone DIAL on 2:1 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig 2:1 [TONE_DETECTED] 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for channel 2:1 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 zchan_activate_dtmf_buffer() Created DTMF Buffer! 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 GENERATE DTMF [3400534] 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 zap_analog_channel_run() Changing state on 2:1 from DIALING to UP 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for UP 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig 2:1 [UP] 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 switch_channel_perform_mark_answered() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() Channel [OpenZAP/2:1/3400534] has been answered 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message [AUDIO_SYNC] 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 switch_channel_perform_answer() OpenZAP/1:1/93400534 receive message [ANSWER] 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has been answered 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message [AUDIO_SYNC] 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 switch_core_session_perform_receive_message() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 switch_ivr_originate() Originate Resulted in Success: [OpenZAP/2:1/3400534] 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message [AUDIO_SYNC] 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message [AUDIO_SYNC] 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive message [BRIDGE] 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive message [BRIDGE] 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 switch_core_session_perform_receive_message() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 1:1 for UP 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig [UP] 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA going to sleep 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_EXCHANGE_MEDIA 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() EVENT [ONHOOK][1:1] STATE [UP] 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() Changing state on 1:1 from UP to DOWN 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 1:1 for DOWN 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig [STOP] 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() ANALOG CHANNEL 1:1 thread ended. 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 audio_bridge_thread() OpenZAP/1:1/93400534 ending bridge by request from read function 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 audio_bridge_thread() BRIDGE THREAD DONE [OpenZAP/1:1/93400534] 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 audio_bridge_thread() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 audio_bridge_thread() OpenZAP/1:1/93400534 ending bridge by request from write function 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 audio_bridge_thread() OpenZAP/2:1/3400534 receive message [UNBRIDGE] 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 audio_bridge_thread() BRIDGE THREAD DONE [OpenZAP/2:1/3400534] 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 audio_bridge_thread() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA going to sleep 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_HANGUP 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() Changing state on 2:1 from UP to HANGUP 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() OpenZAP/2:1/3400534 CHANNEL HANGUP 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, cause: NORMAL_CLEARING 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to sleep 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Locked, Waiting on external entities 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to sleep 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 [CS_HANGUP] 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change CS_HANGUP 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() OpenZAP/1:1/93400534 CHANNEL HANGUP 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, cause: NORMAL_CLEARING 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to sleep 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Locked, Waiting on external entities 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 [CS_HANGUP] 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for HANGUP 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() Changing state on 2:1 from HANGUP to DOWN 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for DOWN 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig 2:1 [STOP] 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 2:1 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() ANALOG CHANNEL 2:1 thread ended. LOG OF FAILED CALLS ================== 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute bridge(openzap/2/1/3400534) 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU 20ms 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 channel_outgoing_channel() Connect outbound channel OpenZAP/2:1/3400534 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/2:1/3400534 [079f5420-ea89-11dd-9f5c-290fb4a527a4] 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 channel_outgoing_channel() (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call() Changing state on 2:1 from DOWN to DIALING 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() ANALOG CHANNEL thread starting. 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_INIT 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for DIALING 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (OpenZAP/2:1/3400534) State INIT 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to sleep 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_ROUTING 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() OpenZAP/2:1/3400534 CHANNEL ROUTING 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to sleep 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_CONSUME_MEDIA 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() EVENT [ONHOOK][1:1] STATE [IDLE] 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() Changing state on 1:1 from IDLE to DOWN 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 1:1 for DOWN 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig [STOP] 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() ANALOG CHANNEL 1:1 thread ended. 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 switch_ivr_originate() Hangup OpenZAP/2:1/3400534 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 switch_ivr_originate() Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to sleep 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change CS_HANGUP 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() OpenZAP/1:1/93400534 CHANNEL HANGUP 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, cause: NORMAL_CLEARING 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to sleep 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Locked, Waiting on external entities 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA going to sleep 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 [CS_HANGUP] 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_HANGUP 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() Changing state on 2:1 from DIALING to HANGUP 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() OpenZAP/2:1/3400534 CHANNEL HANGUP 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to sleep 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Locked, Waiting on external entities 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 [CS_HANGUP] 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for HANGUP 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() Changing state on 2:1 from HANGUP to DOWN 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for DOWN 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig 2:1 [STOP] 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 2:1 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() ANALOG CHANNEL 2:1 thread ended. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090125/42bdb64a/attachment-0001.html From jim at archer.net Sun Jan 25 15:49:55 2009 From: jim at archer.net (Jim Archer) Date: Sun, 25 Jan 2009 18:49:55 -0500 Subject: [Freeswitch-users] Phone configuration In-Reply-To: <3885f4fe0901251146k78eb71e6u28cba840b8d2028b@mail.gmail.com> References: <3885f4fe0901251000p608273adgbd6d12df0b029ff8@mail.gmail.com> <191c3a030901251010x52b0f4b8k8b5d07ea6a083dfc@mail.gmail.com> <191c3a030901251013i785b759al992f49a02efce59@mail.gmail.com> <3885f4fe0901251146k78eb71e6u28cba840b8d2028b@mail.gmail.com> Message-ID: <497CFAA3.8080801@archer.net> Hi All... I looked through the wiki but I can't find instructions or information about how to configure phones to work with FreeSwitch. I have some Polycom 501 phones, and they have a bunch of fields. Is there some info about getting these or oher phones to work? I did see that they are reported to work. Thanks... From jason at jasonjgw.net Sun Jan 25 16:21:15 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 26 Jan 2009 11:21:15 +1100 Subject: [Freeswitch-users] PortAudio not closing audio device properly after hangup? In-Reply-To: <191c3a030901251315t308887adgc3f774c23f635ade@mail.gmail.com> References: <20090125092423.GA22684@jdc.jasonjgw.net> <8BD2BD9C-6075-48C8-A620-89B213176BCF@freeswitch.org> <20090125100244.GA23456@jdc.jasonjgw.net> <4346C468-8A78-43CD-AB07-81ABCFC23802@freeswitch.org> <20090125101913.GA23605@jdc.jasonjgw.net> <191c3a030901251315t308887adgc3f774c23f635ade@mail.gmail.com> Message-ID: <20090126002115.GA30037@jdc.jasonjgw.net> Anthony Minessale wrote: > I'll try another approach. I've tested it once and, so far, this solves the problem. Thanks for the very quick response to my report. From brian at freeswitch.org Sun Jan 25 16:24:02 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 25 Jan 2009 18:24:02 -0600 Subject: [Freeswitch-users] Phone configuration In-Reply-To: <497CFAA3.8080801@archer.net> References: <3885f4fe0901251000p608273adgbd6d12df0b029ff8@mail.gmail.com> <191c3a030901251010x52b0f4b8k8b5d07ea6a083dfc@mail.gmail.com> <191c3a030901251013i785b759al992f49a02efce59@mail.gmail.com> <3885f4fe0901251146k78eb71e6u28cba840b8d2028b@mail.gmail.com> <497CFAA3.8080801@archer.net> Message-ID: Fill out the Display Name, Address, Label, userID, Password via the web interface or the user interface on the phone itself. Also when emailing the list please try not to click reply on new posts. If you click reply, change the subject and delete the body you hijack the current thread which can result in someone ignoring your email if the lists continues to grow. Click "new message" then input freeswitch-users at freeswitch.org and that will start a new thread. Thanks, Brian On Jan 25, 2009, at 5:49 PM, Jim Archer wrote: > Hi All... > > I looked through the wiki but I can't find instructions or information > about how to configure phones to work with FreeSwitch. I have some > Polycom 501 phones, and they have a bunch of fields. Is there some > info > about getting these or oher phones to work? I did see that they are > reported to work. > > Thanks... > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Sun Jan 25 18:18:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 25 Jan 2009 20:18:54 -0600 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> Message-ID: <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> Its not detecting a dial tone on the failure case. Before dialing it waits until it picks up dialtone. Try the svn trunk version to see if it works any better or verify there is a dialtone on the line. On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: hi everybody, i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working using IP phones, softphones and digium FXS port. but there's a problem in dialing out to PSTN using digium tdm400 fxo - it works fine on the first attempt (after starting FS) but it fails on the subsequent attempts. i tested to call using the FXS port and IP phone. same problem. before i place any call, i checked >oz dump 2 1 (show current state = DOWN, last state = DOWN) in the first call, there's this message: [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for channel 2:1 but then i hangup. checked >oz dump 21 (show current state=DOWN, last state=HANGUP) in the 2nd (and subsequent) attempts, the fxo just goes off-hook but doesn't send the dtmf tones. >oz dump 2 1 (shows current state = DIALING, last state = DOWN) has anyone encountered this problem before? i appreciate for any help to correct this problem. tks, nandy Environment: ================== kernel 2.6.18-92.1.22.el5 FS 1.0.2 zaptel 1.4.11 oslec digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) zaptel.conf ======== loadzone = us defaultzone=us channels=1-2 alaw=1-4 fxsks=2 fxoks=1 openzap.conf.xml: =============== openzap.conf ========== [span zt] name => OpenZAP FXS number => 1 fxs-channel => 1 [span zt] name => OpenZAP FXO number => 2 fxo-channel => 2 tones.conf (the dialtone and ring tone is set to Philipping tones) ======== [us] generate-dial => v=-7;%(1000,0,425) detect-dial => 425 generate-ring => v=-7;%(1000,4000,425,480) detect-ring => 425,480 generate-busy => v=-7;%(500,500,480,620) detect-busy => 480,620 generate-attn => v=0;%(200,300,1400,1800) detect-attn => 1400,1800 generate-callwaiting-sas => v=0;%(300,10000,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 776.7 LOG OF FIRST CALL (OK) ==================== 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute bridge(openzap/2/1/3400534) 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU 20ms 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 channel_outgoing_channel() Connect outbound channel OpenZAP/2:1/3400534 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/2:1/3400534 [e5f12114-ea88-11dd-9f5c-290fb4a527a4] 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 channel_outgoing_channel() (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call() Changing state on 2:1 from DOWN to DIALING 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() ANALOG CHANNEL thread starting. 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_INIT 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for DIALING 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (OpenZAP/2:1/3400534) State INIT 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to sleep 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_ROUTING 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() OpenZAP/2:1/3400534 CHANNEL ROUTING 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to sleep 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_CONSUME_MEDIA 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 zap_analog_channel_run() Detected tone DIAL on 2:1 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig 2:1 [TONE_DETECTED] 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for channel 2:1 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 zchan_activate_dtmf_buffer() Created DTMF Buffer! 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 GENERATE DTMF [3400534] 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 zap_analog_channel_run() Changing state on 2:1 from DIALING to UP 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for UP 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig 2:1 [UP] 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 switch_channel_perform_mark_answered() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() Channel [OpenZAP/2:1/3400534] has been answered 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message [AUDIO_SYNC] 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 switch_channel_perform_answer() OpenZAP/1:1/93400534 receive message [ANSWER] 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has been answered 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message [AUDIO_SYNC] 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 switch_core_session_perform_receive_message() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 switch_ivr_originate() Originate Resulted in Success: [OpenZAP/2:1/3400534] 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message [AUDIO_SYNC] 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message [AUDIO_SYNC] 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive message [BRIDGE] 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive message [BRIDGE] 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 switch_core_session_perform_receive_message() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 1:1 for UP 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig [UP] 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA going to sleep 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_EXCHANGE_MEDIA 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 channel_on_exchange_media() CHANNEL EXCHANGE_MEDIA 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() EVENT [ONHOOK][1:1] STATE [UP] 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() Changing state on 1:1 from UP to DOWN 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 1:1 for DOWN 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig [STOP] 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() ANALOG CHANNEL 1:1 thread ended. 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 audio_bridge_thread() OpenZAP/1:1/93400534 ending bridge by request from read function 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 audio_bridge_thread() BRIDGE THREAD DONE [OpenZAP/1:1/93400534] 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 audio_bridge_thread() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 audio_bridge_thread() OpenZAP/1:1/93400534 ending bridge by request from write function 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 audio_bridge_thread() OpenZAP/2:1/3400534 receive message [UNBRIDGE] 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 audio_bridge_thread() BRIDGE THREAD DONE [OpenZAP/2:1/3400534] 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 audio_bridge_thread() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA going to sleep 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_HANGUP 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() Changing state on 2:1 from UP to HANGUP 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() OpenZAP/2:1/3400534 CHANNEL HANGUP 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, cause: NORMAL_CLEARING 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to sleep 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Locked, Waiting on external entities 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to sleep 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 [CS_HANGUP] 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change CS_HANGUP 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() OpenZAP/1:1/93400534 CHANNEL HANGUP 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, cause: NORMAL_CLEARING 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to sleep 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Locked, Waiting on external entities 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 [CS_HANGUP] 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for HANGUP 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() Changing state on 2:1 from HANGUP to DOWN 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for DOWN 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig 2:1 [STOP] 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 2:1 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() ANALOG CHANNEL 2:1 thread ended. LOG OF FAILED CALLS ================== 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute bridge(openzap/2/1/3400534) 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU 20ms 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 channel_outgoing_channel() Connect outbound channel OpenZAP/2:1/3400534 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/2:1/3400534 [079f5420-ea89-11dd-9f5c-290fb4a527a4] 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 channel_outgoing_channel() (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call() Changing state on 2:1 from DOWN to DIALING 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() ANALOG CHANNEL thread starting. 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_INIT 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for DIALING 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (OpenZAP/2:1/3400534) State INIT 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to sleep 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_ROUTING 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() OpenZAP/2:1/3400534 CHANNEL ROUTING 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to sleep 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_CONSUME_MEDIA 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() EVENT [ONHOOK][1:1] STATE [IDLE] 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() Changing state on 1:1 from IDLE to DOWN 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 1:1 for DOWN 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig [STOP] 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/1:1/93400534 [BREAK] 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 1:1 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() ANALOG CHANNEL 1:1 thread ended. 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 switch_ivr_originate() Hangup OpenZAP/2:1/3400534 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 [BREAK] 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 switch_ivr_originate() Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to sleep 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change CS_HANGUP 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() OpenZAP/1:1/93400534 CHANNEL HANGUP 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, cause: NORMAL_CLEARING 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to sleep 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Locked, Waiting on external entities 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA going to sleep 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 [CS_HANGUP] 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_HANGUP 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() Changing state on 2:1 from DIALING to HANGUP 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() OpenZAP/2:1/3400534 CHANNEL HANGUP 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, cause: ORIGINATOR_CANCEL 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to sleep 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Locked, Waiting on external entities 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 [CS_HANGUP] 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for HANGUP 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() Changing state on 2:1 from HANGUP to DOWN 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for DOWN 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig 2:1 [STOP] 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel done 2:1 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() ANALOG CHANNEL 2:1 thread ended. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090125/f284c069/attachment-0001.html From jim at archer.net Sun Jan 25 20:23:16 2009 From: jim at archer.net (Jim Archer) Date: Sun, 25 Jan 2009 23:23:16 -0500 Subject: [Freeswitch-users] Phone configuration In-Reply-To: References: <3885f4fe0901251000p608273adgbd6d12df0b029ff8@mail.gmail.com> <191c3a030901251010x52b0f4b8k8b5d07ea6a083dfc@mail.gmail.com> <191c3a030901251013i785b759al992f49a02efce59@mail.gmail.com> <3885f4fe0901251146k78eb71e6u28cba840b8d2028b@mail.gmail.com> <497CFAA3.8080801@archer.net> Message-ID: <497D3AB4.4060404@archer.net> Hi Brian and thanks... On the new message, I could have sworn I did exactly that, reply but deleted body and subject line. But I'll make a new message from now on. Thanks again... Jim Brian West wrote: > Fill out the Display Name, Address, Label, userID, Password via the > web interface or the user interface on the phone itself. > > Also when emailing the list please try not to click reply on new > posts. If you click reply, change the subject and delete the body you > hijack the current thread which can result in someone ignoring your > email if the lists continues to grow. Click "new message" then input freeswitch-users at freeswitch.org > and that will start a new thread. > > Thanks, > Brian > > > > On Jan 25, 2009, at 5:49 PM, Jim Archer wrote: > >> Hi All... >> >> I looked through the wiki but I can't find instructions or information >> about how to configure phones to work with FreeSwitch. I have some >> Polycom 501 phones, and they have a bunch of fields. Is there some >> info >> about getting these or oher phones to work? I did see that they are >> reported to work. >> >> Thanks... >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Sun Jan 25 20:29:55 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 25 Jan 2009 22:29:55 -0600 Subject: [Freeswitch-users] Phone configuration In-Reply-To: <497D3AB4.4060404@archer.net> References: <3885f4fe0901251000p608273adgbd6d12df0b029ff8@mail.gmail.com> <191c3a030901251010x52b0f4b8k8b5d07ea6a083dfc@mail.gmail.com> <191c3a030901251013i785b759al992f49a02efce59@mail.gmail.com> <3885f4fe0901251146k78eb71e6u28cba840b8d2028b@mail.gmail.com> <497CFAA3.8080801@archer.net> <497D3AB4.4060404@archer.net> Message-ID: <94824852-ED77-40BE-8E5F-8F7082578679@freeswitch.org> Yah doing this causes the thread to be hijacked because it includes a reference header back to the list server in the reply.. changing the subject or body doesn't start a new thread ;) /b On Jan 25, 2009, at 10:23 PM, Jim Archer wrote: > Hi Brian and thanks... > > On the new message, I could have sworn I did exactly that, reply but > deleted body and subject line. But I'll make a new message from now > on. > > Thanks again... > > Jim From mrene_lists at avgs.ca Sun Jan 25 20:53:09 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 25 Jan 2009 23:53:09 -0500 Subject: [Freeswitch-users] Can I have FS send a 503? In-Reply-To: <3885f4fe0901251146k78eb71e6u28cba840b8d2028b@mail.gmail.com> References: <3885f4fe0901251000p608273adgbd6d12df0b029ff8@mail.gmail.com> <191c3a030901251010x52b0f4b8k8b5d07ea6a083dfc@mail.gmail.com> <191c3a030901251013i785b759al992f49a02efce59@mail.gmail.com> <3885f4fe0901251146k78eb71e6u28cba840b8d2028b@mail.gmail.com> Message-ID: NORMAL_CIRCUIT_CONGESTION would've been the cause you were looking for. Mathieu On Sun, Jan 25, 2009 at 2:46 PM, Ron McCarthy wrote: > I actaully just used respond with a 503 and that sends back a "Service > unavailable". > > If I do: I > get a busy back on the other side, it looks like application="respond" > actaully works better in this case. > > Thanks for the help! > > > > On Sun, Jan 25, 2009 at 11:13 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> We use the standard rfc q.850 to sip cause code map so you can set >> continue_on_fail=true and hangup with the q.850 that maps to 503 (I forgot >> what it is atm) >> >> On Jan 25, 2009 12:09 PM, "Ron McCarthy" wrote: >> >> Hi, >> >> Is it possible to have FS send a 503 / no route found error back? I have >> looked and have not seen a place where you can add a custom sip response >> code like this. I want to return a 503 if they can't call that destination, >> instead of a 404 basically. >> >> Any help would be great, thanks! >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090125/f44d58b5/attachment.html From krice at freeswitch.org Sun Jan 25 21:22:39 2009 From: krice at freeswitch.org (Ken Rice) Date: Sun, 25 Jan 2009 23:22:39 -0600 Subject: [Freeswitch-users] Can I have FS send a 503? In-Reply-To: Message-ID: You can make the respond app reply with any sip message you want... Example K From: Mathieu Rene Reply-To: Date: Sun, 25 Jan 2009 23:53:09 -0500 To: Subject: Re: [Freeswitch-users] Can I have FS send a 503? NORMAL_CIRCUIT_CONGESTION would've been the cause you were looking for. Mathieu On Sun, Jan 25, 2009 at 2:46 PM, Ron McCarthy wrote: > I actaully just used respond with a 503 and that sends back a "Service > unavailable". > > If I do: I > get a busy back on the other side, it looks like application="respond" > actaully works better in this case. > > Thanks for the help! > > > > On Sun, Jan 25, 2009 at 11:13 AM, Anthony Minessale > wrote: >> >> We use the standard rfc q.850 to sip cause code map so you can set >> continue_on_fail=true and hangup with the q.850 that maps to 503 (I forgot >> what it is atm) >> >>> On Jan 25, 2009 12:09 PM, "Ron McCarthy" wrote: >>> >>> Hi, >>> >>> Is it possible to have FS send a 503 / no route found error back? I have >>> looked and have not seen a place where you can add a custom sip response >>> code like this. I want to return a 503 if they can't call that destination, >>> instead of a 404 basically. >>> >>> Any help would be great, thanks! >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090125/9e04e8ca/attachment.html From sias at cpdata.co.za Sun Jan 25 22:55:18 2009 From: sias at cpdata.co.za (Sias Mey) Date: Mon, 26 Jan 2009 08:55:18 +0200 Subject: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches In-Reply-To: <191c3a030901230613t6335d8ic48e3ce22786803f@mail.gmail.com> References: <20090122202406.GA21067@cpdata.co.za> <191c3a030901221425p379ad74j7f866eee4596a6a9@mail.gmail.com> <20090123095036.GA6568@cpdata.co.za> <20090123100326.GA6830@cpdata.co.za> <191c3a030901230613t6335d8ic48e3ce22786803f@mail.gmail.com> Message-ID: <20090126065518.GA6742@cpdata.co.za> Hmmm no it didnt... but at least now I know what to fix when it doesent work whe I update again. Thank you very much for your help. Sias On Fri, Jan 23, 2009 at 08:13:16AM -0600, Anthony Minessale wrote: > That was the change i checked into trunk to allow app::arg as well as > apparg > that doesn't work for you? When i said update it was down to the > minute i sent the email that the change was added. > > On Fri, Jan 23, 2009 at 4:03 AM, Sias Mey <[1]sias at cpdata.co.za> wrote: > > Woot greater win. > Thanks you so much for that pointer. > although i did have to change the dialplan line to > > data="{api_hangup_hook=jsapi > foo.js}sofia/default/[1][2]user at dest.com"/> > > (space between jsapi and foo.js instead of ::) > and im not sure if the api.js file actually made any difference.. > but it > did point me in the right direction. > > On Fri, Jan 23, 2009 at 11:50:36AM +0200, Sias Mey wrote: > > Wait sory ignore my previous reply... > > > > I only just realized you were actually routing through the javascript > > xml_rpc module. and I didnt actually have the api.js file in my > scripts > > dir. > > > > let me see what this does before you worry about it any more ;-) > > On Thu, Jan 22, 2009 at 04:25:54PM -0600, Anthony Minessale wrote: > > > Try this (update to svn trunk first) > > > > > > data="{api_hangup_hook=jsapi::foo.js}sofia/default/[1][3]user at dest.com" > /> > > > then place your call as usual > > > then in foo.js > > > // dumps the event to text/plain > > > env = request.dumpENV("text"); > > > // dumps the event to text/xml > > > xmlenv = request.dumpENV("xml"); > > > // makes an XML obj from the xml text > > > xinfo = new XML("" + xmlenv + ""); > > > // dump the plain text event data > > > consoleLog("info", env + "\n"); > > > // dump the xml event data > > > consoleLog("info", xmlenv + "\n"); > > > // Get a header from the event object > > > consoleLog("warning", "media ip was [" + > > > request.getHeader("local_media_ip") + "]\n"); > > > // Get the same header from the xml object > > > consoleLog("warning", "media ip was [" + > > > xinfo.event.headers.local_media_ip + "]\n"); > > > > > > On Thu, Jan 22, 2009 at 2:24 PM, Sias Mey > <[2][4]sias at cpdata.co.za> wrote: > > > > > > Hi, > > > Im trying to originate calls from a conference and use > javascript to > > > watch out for hangup events so I can use the data in the > session to > > > flesh out some database info. However it seems that Im having > some > > > strangeness. It might just be my code. So I include that. > > > I run FreeSwitch Version 1.0.trunk (11226) > > > Dialplan: > > > > > > > > expression="^confout-(10\d{2})$"> > > > > > > > > > > > > confout.js: > > > is attached > > > I use API calls to pull one user into a conference. Then I use > more > > > api calls to do a conference dial to loopback/confout-1001 > > > This should run the js and then bridge extension 1001 into the > same > > > conference. > > > (I have hardcoded the additional extension for testing). I > dont know > > > if there is another way to get a conference dial to run a > javascript > > > file for information logging, but I am open to enlightenment. > > > Oh im using conference dial because that provides clear > audible > > > progress to the other conference memebers as to what is > actually > > > happening with the new call. > > > Any help would be greatly apreciated, Thanks in advance. > > > Sias > > > _______________________________________________ > > > Freeswitch-users mailing list > > > [3][5]Freeswitch-users at lists.freeswitch.org > > > > [4][6]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:[5][7]http://lists.freeswitch.org/mailman/options/freeswitc > > > h-users > > > [6][8]http://www.freeswitch.org > > > > > > -- > > > Anthony Minessale II > > > FreeSWITCH [7][9]http://www.freeswitch.org/ > > > ClueCon [8][10]http://www.cluecon.com/ > > > AIM: anthm > > > [9][11]MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/[10][12]PAYPAL:anthony.minessale at gmail.com > > > IRC: [11][13]irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > > > [12][14]sip:888 at conference.freeswitch.org > > > [13][15]iax:guest at conference.freeswitch.org/888 > > > [14][16]googletalk:conf+888 at conference.freeswitch.org > > > pstn:213-799-1400 > > > > > > References > > > > > > 1. mailto:[17]user at dest.com > > > 2. mailto:[18]sias at cpdata.co.za > > > 3. mailto:[19]Freeswitch-users at lists.freeswitch.org > > > 4. > [20]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > 5. > [21]http://lists.freeswitch.org/mailman/options/freeswitch-users > > > 6. [22]http://www.freeswitch.org/ > > > 7. [23]http://www.freeswitch.org/ > > > 8. [24]http://www.cluecon.com/ > > > 9. mailto:[25]MSN%3Aanthony_minessale at hotmail.com > > > 10. mailto:[26]PAYPAL%3Aanthony.minessale at gmail.com > > > 11. [27]http://irc.freenode.net/ > > > 12. mailto:[28]sip%3A888 at conference.freeswitch.org > > > 13. [29]http://iax:guest at conference.freeswitch.org/888 > > > 14. mailto:[30]googletalk%3Aconf%2B888 at conference.freeswitch.org > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > [31]Freeswitch-users at lists.freeswitch.org > > > [32]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:[33]http://lists.freeswitch.org/mailman/options/freeswitch- > users > > > [34]http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > [35]Freeswitch-users at lists.freeswitch.org > > [36]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:[37]http://lists.freeswitch.org/mailman/options/freeswitch- > users > > [38]http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > [39]Freeswitch-users at lists.freeswitch.org > [40]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[41]http://lists.freeswitch.org/mailman/options/freeswitch- > users > [42]http://www.freeswitch.org > > -- > Anthony Minessale II > FreeSWITCH [43]http://www.freeswitch.org/ > ClueCon [44]http://www.cluecon.com/ > AIM: anthm > [45]MSN:anthony_minessale at hotmail.com > GTALK/JABBER/[46]PAYPAL:anthony.minessale at gmail.com > IRC: [47]irc.freenode.net #freeswitch > FreeSWITCH Developer Conference > [48]sip:888 at conference.freeswitch.org > [49]iax:guest at conference.freeswitch.org/888 > [50]googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > References > > 1. mailto:sias at cpdata.co.za > 2. mailto:user at dest.com > 3. mailto:user at dest.com > 4. mailto:sias at cpdata.co.za > 5. mailto:Freeswitch-users at lists.freeswitch.org > 6. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 7. http://lists.freeswitch.org/mailman/options/freeswitc > 8. http://www.freeswitch.org/ > 9. http://www.freeswitch.org/ > 10. http://www.cluecon.com/ > 11. mailto:MSN%3Aanthony_minessale at hotmail.com > 12. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 13. http://irc.freenode.net/ > 14. mailto:sip%3A888 at conference.freeswitch.org > 15. http://iax:guest at conference.freeswitch.org/888 > 16. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > 17. mailto:user at dest.com > 18. mailto:sias at cpdata.co.za > 19. mailto:Freeswitch-users at lists.freeswitch.org > 20. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 21. http://lists.freeswitch.org/mailman/options/freeswitch-users > 22. http://www.freeswitch.org/ > 23. http://www.freeswitch.org/ > 24. http://www.cluecon.com/ > 25. mailto:MSN%253Aanthony_minessale at hotmail.com > 26. mailto:PAYPAL%253Aanthony.minessale at gmail.com > 27. http://irc.freenode.net/ > 28. mailto:sip%253A888 at conference.freeswitch.org > 29. http://iax:guest at conference.freeswitch.org/888 > 30. mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org > 31. mailto:Freeswitch-users at lists.freeswitch.org > 32. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 33. http://lists.freeswitch.org/mailman/options/freeswitch-users > 34. http://www.freeswitch.org/ > 35. mailto:Freeswitch-users at lists.freeswitch.org > 36. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 37. http://lists.freeswitch.org/mailman/options/freeswitch-users > 38. http://www.freeswitch.org/ > 39. mailto:Freeswitch-users at lists.freeswitch.org > 40. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 41. http://lists.freeswitch.org/mailman/options/freeswitch-users > 42. http://www.freeswitch.org/ > 43. http://www.freeswitch.org/ > 44. http://www.cluecon.com/ > 45. mailto:MSN%3Aanthony_minessale at hotmail.com > 46. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 47. http://irc.freenode.net/ > 48. mailto:sip%3A888 at conference.freeswitch.org > 49. http://iax:guest at conference.freeswitch.org/888 > 50. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gcd at i.ph Sun Jan 25 23:05:36 2009 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 26 Jan 2009 15:05:36 +0800 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> Message-ID: <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> i monitored the line using another phone. there's indeed dialtone in all attempts. i see TONE_DETECTED in the first call but i wonder there's a WARNING message immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for channel 2:1. the dialtone freq should be okay since it's detected in the first call.could the WARNING message gives us a hint of a possible problem other than the dialtone freq? okay, i'll try the SVN version next. On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Its not detecting a dial tone on the failure case. > Before dialing it waits until it picks up dialtone. > Try the svn trunk version to see if it works any better or verify there is > a dialtone on the line. > > On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: > > hi everybody, > > i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working using > IP phones, softphones and digium FXS port. but there's a problem in dialing > out to PSTN using digium tdm400 fxo - it works fine on the first attempt > (after starting FS) but it fails on the subsequent attempts. i tested to > call using the FXS port and IP phone. same problem. > > before i place any call, i checked >oz dump 2 1 (show current state = > DOWN, last state = DOWN) > > in the first call, there's this message: > [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for channel 2:1 > but > > then i hangup. checked >oz dump 21 (show current state=DOWN, last > state=HANGUP) > > in the 2nd (and subsequent) attempts, the fxo just goes off-hook but > doesn't send the dtmf tones. > >oz dump 2 1 (shows current state = DIALING, last state = DOWN) > > has anyone encountered this problem before? i appreciate for any help to > correct this problem. > > tks, > nandy > > > Environment: > ================== > kernel 2.6.18-92.1.22.el5 > FS 1.0.2 > zaptel 1.4.11 > oslec > digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) > > zaptel.conf > ======== > loadzone = us > defaultzone=us > channels=1-2 > alaw=1-4 > fxsks=2 > fxoks=1 > > > openzap.conf.xml: > =============== > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > openzap.conf > ========== > [span zt] > name => OpenZAP FXS > number => 1 > fxs-channel => 1 > > [span zt] > name => OpenZAP FXO > number => 2 > fxo-channel => 2 > > tones.conf (the dialtone and ring tone is set to Philipping tones) > ======== > [us] > generate-dial => v=-7;%(1000,0,425) > detect-dial => 425 > > generate-ring => v=-7;%(1000,4000,425,480) > detect-ring => 425,480 > > generate-busy => v=-7;%(500,500,480,620) > detect-busy => 480,620 > > generate-attn => v=0;%(200,300,1400,1800) > detect-attn => 1400,1800 > > generate-callwaiting-sas => v=0;%(300,10000,440) > detect-callwaiting-sas => 440 > > generate-callwaiting-cas => v=0;%(80,0,2750,2130) > detect-callwaiting-cas => 2750,2130 > > detect-fail1 => 913.8 > detect-fail2 => 1370.6 > detect-fail3 => 776.7 > > LOG OF FIRST CALL (OK) > ==================== > 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute > bridge(openzap/2/1/3400534) > 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU > 20ms > 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 channel_outgoing_channel() > Connect outbound channel OpenZAP/2:1/3400534 > 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 switch_channel_set_name() > New Channel OpenZAP/2:1/3400534 [e5f12114-ea88-11dd-9f5c-290fb4a527a4] > 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 channel_outgoing_channel() > (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT > 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 > [BREAK] > 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call() > Changing state on 2:1 from DOWN to DIALING > 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() > ANALOG CHANNEL thread starting. > 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_INIT > 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 2:1 for DIALING > 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (OpenZAP/2:1/3400534) State INIT > 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() > (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING > 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 > [BREAK] > 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to sleep > 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > CS_ROUTING > 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING > 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() > OpenZAP/2:1/3400534 CHANNEL ROUTING > 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 > originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 > [BREAK] > 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to sleep > 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > CS_CONSUME_MEDIA > 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA > 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 zap_analog_channel_run() > Detected tone DIAL on 2:1 > 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig > 2:1 [TONE_DETECTED] > 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled > type for channel 2:1 > 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 zchan_activate_dtmf_buffer() > Created DTMF Buffer! > 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 GENERATE DTMF > [3400534] > 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 zap_analog_channel_run() > Changing state on 2:1 from DIALING to UP > 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 2:1 for UP > 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig > 2:1 [UP] > 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 > switch_channel_perform_mark_answered() Send signal OpenZAP/1:1/93400534 > [BREAK] > 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() Channel > [OpenZAP/2:1/3400534] has been answered > 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message [AUDIO_SYNC] > 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 > switch_channel_perform_answer() OpenZAP/1:1/93400534 receive message > [ANSWER] > 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 channel_receive_message_fxs() > Changing state on 1:1 from IDLE to UP > 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 > channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has been > answered > 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message > [AUDIO_SYNC] > 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 > switch_core_session_perform_receive_message() Send signal > OpenZAP/1:1/93400534 [BREAK] > 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 > switch_ivr_originate() Originate Resulted in Success: [OpenZAP/2:1/3400534] > 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message [AUDIO_SYNC] > 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message > [AUDIO_SYNC] > 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 > switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive message > [BRIDGE] > 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/3400534 [BREAK] > 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 > switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive message > [BRIDGE] > 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 > switch_core_session_perform_receive_message() Send signal > OpenZAP/1:1/93400534 [BREAK] > 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 > switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State Change > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 > [BREAK] > 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 1:1 for UP > 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig > [UP] > 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA going to > sleep > 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > CS_EXCHANGE_MEDIA > 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 > switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA > 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 channel_on_exchange_media() > CHANNEL EXCHANGE_MEDIA > 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [ONHOOK][1:1] STATE [UP] > 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() Changing > state on 1:1 from UP to DOWN > 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 1:1 for DOWN > 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig > [STOP] > 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup > OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 > switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] > 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal OpenZAP/1:1/93400534 > [BREAK] > 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() channel done > 1:1 > 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() > ANALOG CHANNEL 1:1 thread ended. > 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 audio_bridge_thread() > OpenZAP/1:1/93400534 ending bridge by request from read function > 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 audio_bridge_thread() > BRIDGE THREAD DONE [OpenZAP/1:1/93400534] > 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 audio_bridge_thread() > Send signal OpenZAP/2:1/3400534 [BREAK] > 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 audio_bridge_thread() > OpenZAP/1:1/93400534 ending bridge by request from write function > 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 audio_bridge_thread() > OpenZAP/2:1/3400534 receive message [UNBRIDGE] > 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 > switch_core_session_perform_receive_message() Send signal > OpenZAP/2:1/3400534 [BREAK] > 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 audio_bridge_thread() > BRIDGE THREAD DONE [OpenZAP/2:1/3400534] > 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 audio_bridge_thread() > Send signal OpenZAP/1:1/93400534 [BREAK] > 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 > audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 > switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] > 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 > [BREAK] > 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 > switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA going > to sleep > 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > CS_HANGUP > 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP > 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() Changing > state on 2:1 from UP to HANGUP > 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() > OpenZAP/2:1/3400534 CHANNEL HANGUP > 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, cause: > NORMAL_CLEARING > 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to sleep > 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 > switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Locked, Waiting > on external entities > 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to > sleep > 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 > switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended > 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 > switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 [CS_HANGUP] > 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change > CS_HANGUP > 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP > 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() > OpenZAP/1:1/93400534 CHANNEL HANGUP > 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, > cause: NORMAL_CLEARING > 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to sleep > 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 > switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Locked, > Waiting on external entities > 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 > switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended > 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 > switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 [CS_HANGUP] > 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 2:1 for HANGUP > 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() > Changing state on 2:1 from HANGUP to DOWN > 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 2:1 for DOWN > 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig > 2:1 [STOP] > 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel done > 2:1 > 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() > ANALOG CHANNEL 2:1 thread ended. > > LOG OF FAILED CALLS > ================== > 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute > bridge(openzap/2/1/3400534) > 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU > 20ms > 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 channel_outgoing_channel() > Connect outbound channel OpenZAP/2:1/3400534 > 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 switch_channel_set_name() > New Channel OpenZAP/2:1/3400534 [079f5420-ea89-11dd-9f5c-290fb4a527a4] > 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 channel_outgoing_channel() > (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT > 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 > [BREAK] > 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call() > Changing state on 2:1 from DOWN to DIALING > 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() > ANALOG CHANNEL thread starting. > 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_INIT > 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 2:1 for DIALING > 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (OpenZAP/2:1/3400534) State INIT > 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() > (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING > 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 > [BREAK] > 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to sleep > 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > CS_ROUTING > 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING > 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() > OpenZAP/2:1/3400534 CHANNEL ROUTING > 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 > originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 > [BREAK] > 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to sleep > 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > CS_CONSUME_MEDIA > 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA > 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() EVENT > [ONHOOK][1:1] STATE [IDLE] > 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() Changing > state on 1:1 from IDLE to DOWN > 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 1:1 for DOWN > 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig > [STOP] > 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup > OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 > switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] > 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal OpenZAP/1:1/93400534 > [BREAK] > 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel done > 1:1 > 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() > ANALOG CHANNEL 1:1 thread ended. > 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 > switch_ivr_originate() Hangup OpenZAP/2:1/3400534 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 > switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] > 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 > [BREAK] > 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 > switch_ivr_originate() Originate Cancelled by originator termination Cause: > 487 [ORIGINATOR_CANCEL] > 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 audio_bridge_function() > Originate Failed. Cause: ORIGINATOR_CANCEL > 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to > sleep > 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change > CS_HANGUP > 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP > 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() > OpenZAP/1:1/93400534 CHANNEL HANGUP > 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, > cause: NORMAL_CLEARING > 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to sleep > 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 > switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Locked, > Waiting on external entities > 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 > switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended > 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA going to > sleep > 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 > switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 [CS_HANGUP] > 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > CS_HANGUP > 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP > 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() Changing > state on 2:1 from DIALING to HANGUP > 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() > OpenZAP/2:1/3400534 CHANNEL HANGUP > 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to sleep > 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 > switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Locked, Waiting > on external entities > 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 > switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended > 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 > switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 [CS_HANGUP] > 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 2:1 for HANGUP > 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() > Changing state on 2:1 from HANGUP to DOWN > 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > Executing state handler on 2:1 for DOWN > 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig > 2:1 [STOP] > 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel done > 2:1 > 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() > ANALOG CHANNEL 2:1 thread ended. > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/e7a66331/attachment-0001.html From sicfslist at gmail.com Mon Jan 26 00:45:10 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Mon, 26 Jan 2009 02:45:10 -0600 Subject: [Freeswitch-users] auto dialing question ... In-Reply-To: <191c3a030901231654w2823ed72h382dee33060acd2a@mail.gmail.com> References: <35b355e90901231415q782b5ba9mba2bfc7fe6345fd3@mail.gmail.com> <87f2f3b90901231439u553031f4u29c1f27670fe5c78@mail.gmail.com> <191c3a030901231455g2650cd7brd3736b96490fd55b@mail.gmail.com> <35b355e90901231539s1a2b938crd24279933efbbe2@mail.gmail.com> <87f2f3b90901231555i6d1690a4p25ecece2340acc4e@mail.gmail.com> <35b355e90901231620t83e9320vebfe6728d9b3ebc7@mail.gmail.com> <191c3a030901231654w2823ed72h382dee33060acd2a@mail.gmail.com> Message-ID: <35b355e90901260045y311f3b20u9334e9153faa418b@mail.gmail.com> MC / Anthony, Muchas gracias ... looks like I'm going to be on my way to being * free :) Ran a small production test today (about 20,000 dials) and going to get after it tomorrow with a real campaign. You guys kick ass. Thanks again for the assistance fine sirs. Shelby On Fri, Jan 23, 2009 at 6:54 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > its the same idea where i gave you that example to call 9998 call 1234 > where your extension is delivered via curl when 1234 is requested > use the "read" and "transfer" apps in the xml you return, you can get it > working statically first in your regular dialplan. > > > > On Fri, Jan 23, 2009 at 6:20 PM, Shelby Ramsey wrote: > >> Thanks Michael. >> Here are some really simple examples (we limit what people do through the >> web ... nothing really fancy ... just some good old fashion robo calls). >> ultimately today it looks like this extensions_table (context, exten, >> priority, app, appdata): >> >> campaign,100,1,ANSWER() --> answers call >> campaign,100,2,WAIT(1) --> this is a pause >> campaign,100,3,PLAYBACK($INTROFILE) --> plays intro file >> campaign,100,4,BACKGROUND($MESSAGE) --> plays message >> campaign,100,5,WAITEXTEN(5) --> WAIT for DTMF >> campaign,100,6,HANGUP() >> Here are DTMF options >> campaign,1,SYSTEM($SCRIPT campaign $EXTEN) >> campaign,2,SYSTEM($SCRIPT campaign $EXTEN) >> campaign,3,DIAL,zap/g1/$ANI >> >> anothercampaign,112,1,ANSWER() >> anothercampaign,112,2,BACKGROUND($SOMEFILE) >> anothercampaign,112,3,WAITEXTEN(5) >> anothercampaign,112,4,HANGUP() >> Options: >> anothercampaign,1,DIAL(sip/$ANI@$IP) >> anothercampaign,2,PLAYBACK($GOODBYE) >> >> And they pretty much all look like that ... it's easy to return the stuff >> for extension 100 via XML ... but the challenge is the DTMF options >> (relating to the same context) ... or maybe I'm just missing something >> (which is a definite possibility). We don't ever do anything complicated in >> the IVRs (TTS or ASR) but there is just a lot of them that all get >> controlled, manipulated via a web interface. >> >> I can originate a call ... do all of that ... just trying to figure out a >> "simple" way to return the above via xml. >> >> Thanks for your help Anthony and Michael as always! >> >> Shelby >> >> >> >> >> >> On Fri, Jan 23, 2009 at 5:55 PM, Michael Collins wrote: >> >>> I see your dilemma. To keep things dynamic you definitely want to use >>> your XML_CURL stuff. Like you said, nothing static. >>> >>> Can you post a sample call-flow in plain English? I'm curious about >>> something. I don't want to say anything else. Just post a simple call >>> flow: >>> Initiate call >>> Wait for answer/busy/timeout/invalid >>> On busy/timeout/invalid: update db and move on >>> On answer play greeting, accept digit, route based on digit >>> >>> Something like that would help me conceptualize what you are trying to >>> do. >>> >>> Thanks >>> MC >>> >>> P.S. - I've done a little bit of outbound IVR calling so hopefully I >>> can assist you. >>> >>> On Fri, Jan 23, 2009 at 3:39 PM, Shelby Ramsey >>> wrote: >>> > Anthony / Michael, >>> > Thanks for the quick responses. What I don't want to do is "drive the >>> call" >>> > (by that listen on a socket ... do this on this event ... or anything >>> else >>> > that my very limited FS foo would break) ... Just want to start it and >>> then >>> > give it instructions on where to go. >>> > So I guess a better question would be ... how do I give directions to >>> FS for >>> > this (and I get the 1st part ... that's obvious ... really lost on the >>> DTMF >>> > digit part) ... and please keep in mind we're talking hundreds of >>> extensions >>> > / IVR's and distributed machines so I can't have any dependancy on >>> static >>> > conf files other than maybe something like what Michael mentioned where >>> I >>> > point every call to something: >>> > [campaign] >>> > exten => 100,1,ANSWER() >>> > exten => 100,n,PLAYBACK(somefile) >>> > exten => 100,n,BACKGROUND(somefile) >>> > exten => 100,n,WAITEXTEN(4) >>> > exten => 100,n,HANGUP() >>> > but in that same context is someone triggers DTMF: >>> > exten => 1,1,DOSOMETHING >>> > exten => 2,1,DOSOMETHING >>> > I was imaging issuing originate via XML_RPC ... something like >>> originate >>> > sofia/$ANI@$IP $SOMEEXTEN then on answer when FS tries to connect to >>> > $SOMEEXTEN it will ask me what to do via xml_curl ... where I would >>> normally >>> > respond with something like this: >>> > >> > type="freeswitch/xml">
>>> > >> > field="destination_number" expression=""> >> > data="hangup_after_bridge=true"/> >> > data="continue_on_fail=true"/> >> > data="call_timeout=180"/> >> > data="proxy_media=true"/> >> > data="pass_rfc2833=true"/> >> > data="accountcode=$CUSTOMER" /> >> > data="origination_caller_id_name=NULL" /> >> > data="origination_caller_id_number=$CIDNUM" /> >> application="set" >>> > data="effective_caller_id_name=NULL" /> >> > data="effective_caller_id_number=$CIDNUM" /> >> > data="userfield=$BUNCHOFCRAPFORMYCDR" /> >> > data="sofia/external/$ANI@$PROVIDERIP" /> >>> > >>> >
>>> > The challenge I've got is I have no idea how to do stuff like the IVR >>> > mentioned above (the playback part is easy) ... but I can't grasp >>> > conceptually how to get the "context" with "multiple extensions" part >>> back >>> > to FS via this method (is it possible?)... >>> > Sorry for what is probably a very simple answer and any AST references >>> (but >>> > I've been using it in heavy production environments for about 5 years). >>> Just >>> > trying to "port" what I do today without making my brain melt out of my >>> ears >>> > (and it doesn't take much for that to happen). >>> > Shelby >>> > PS ... Really enjoy the list. I usually fall out of my chair laughing >>> once a >>> > day from your remarks Anthony. Keep it coming! >>> > >>> > On Fri, Jan 23, 2009 at 4:55 PM, Anthony Minessale >>> > wrote: >>> >> >>> >> Does AST mean Asterisk Open Source PBX ? >>> >> >>> >> If so, then yes I am familiar with it's archetechure as I am a former >>> >> developer from that project. >>> >> >>> >> You have 3 choices with FreeSWITCH >>> >> >>> >> 1) You can open a dedicated connection to mod_event_socket or XMLRPC >>> per >>> >> call and issue the originate command from there: >>> >> This will block until you know for sure the outcome of the >>> attempt. >>> >> If it's success it will give you the uuid if not it gives you the >>> cause >>> >> code. >>> >> >>> >> 2) You can use a single mod_event_socket or XMLRPC connection to send >>> all >>> >> calls but use the bgapi mechanism which will do the same as above >>> >> only asynchronously, The command will return immediately and the >>> >> result will be fired as an event that you can pick up on the same or >>> >> different event_socket connection or >>> >> other event consumer such as a custom C,perl,lua etc module. >>> >> >>> >> 3) You can use mod_xml_cdr to generate detailed 1-file-per-leg call >>> files >>> >> that will tell you when where and why the calls failed or did not >>> fail. >>> >> >>> >> >>> >> On Fri, Jan 23, 2009 at 4:39 PM, Michael Collins >>> >> wrote: >>> >>> >>> >>> On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey >>> >>> wrote: >>> >>> > Sorry for the double post ... actually hit send too early ... >>> >>> > OK ... Here goes another I'm doing this with AST ... but I want to >>> >>> > move it >>> >>> > to FS. Searched via google site:lists.freeswitch.org auto dialer >>> and >>> >>> > others >>> >>> > ... nothing useful. >>> >>> > Today I have a platform for auto dialing with AST (centrally >>> managed >>> >>> > ... >>> >>> > about 10 machines) and we do this: >>> >>> > -- Remote machines query central DB for numbers to call based on >>> >>> > certain >>> >>> > configs >>> >>> > -- Use AMI to generate the call >>> >>> > -- If call gets answered, extension info queried via rta (central >>> db >>> >>> > again) >>> >>> > The nice thing about all of this is it's relatively easy to manage >>> >>> > (through >>> >>> > one central web interface we built) and it works ... the bad part >>> is >>> >>> > reporting ... >>> >>> > So ... conceptually I'm trying to accomplish the same thing ... >>> >>> > Today we use FS a lot for termination of VoIP traffic ... all done >>> via >>> >>> > XML_CURL ... which is awesome (not to xml cdr ... and the >>> "proxying" >>> >>> > of >>> >>> > media) ... >>> >>> > Would like to do something like: >>> >>> > -- originate request (looks simple enough) >>> >>> > -- on answer XML_CURL posts info >>> >>> >>> >>> Several choices, depending upon how much you want it handled inside >>> >>> the dialplan vs. handled in the scripting language. For the sake of >>> >>> testing you could do something like this: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Then have: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> This would have any answered call go to the "ivr-answer" extension >>> >>> while unanswered calls could stay in the ivr-start extension to get >>> >>> properly handled. (Busy, no answer, invalid/SIT, etc.) >>> >>> >>> >>> You could then have the "ivr-answer" extension do whatever is >>> >>> appropriate, like listen for digits, play announcement, beg for >>> money, >>> >>> etc. :) >>> >>> >>> >>> -MC >>> >>> >>> >>> > But for the life of me I can't figure out how to translate this >>> into >>> >>> > the xml >>> >>> > response ... >>> >>> > [campaign] >>> >>> > exten => 100,1,ANSWER() >>> >>> > exten => 100,n,WAIT(2) >>> >>> > exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile) >>> >>> > exten => 100,n,WAITEXTEN(10) >>> >>> > exten => 100,n,HANGUP() >>> >>> > exten => 1,1,PLAYBACK(goodbye) >>> >>> > .... and so on ... >>> >>> > I've looked at the ivr.conf stuff but it's all static and all of >>> this >>> >>> > has to >>> >>> > be manageable via a web interface .... meaning dumping into a DB >>> and >>> >>> > returning an XML response seems reasonable ... but trying to stick >>> or >>> >>> > modify >>> >>> > static text files from the web interface is too much text parsing >>> and >>> >>> > bad >>> >>> > things will happen ... >>> >>> > Any thoughts or pointing me in the right direction would be >>> >>> > appreciated. >>> >>> > Shelby >>> >>> > >>> >>> > >>> >>> > >>> >>> > _______________________________________________ >>> >>> > Freeswitch-users mailing list >>> >>> > Freeswitch-users at 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 at 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 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 >>> >> iax:guest at conference.freeswitch.org/888 >>> >> googletalk:conf+888 at conference.freeswitch.org >>> >> pstn:213-799-1400 >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at 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 at 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 at 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 at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/aedcafb2/attachment-0001.html From cstomi.levlist at gmail.com Mon Jan 26 01:50:58 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Mon, 26 Jan 2009 10:50:58 +0100 Subject: [Freeswitch-users] record session in fifo In-Reply-To: <191c3a030901231045m784f0cd7s6994b763d02b3e26@mail.gmail.com> References: <4979F41F.70605@gmail.com> <191c3a030901231045m784f0cd7s6994b763d02b3e26@mail.gmail.com> Message-ID: <497D8782.2070603@gmail.com> Hello, Thank you your help. I tested with r11489, but moh is still recorded in fifo. I quess you I should test the CF_PAUSE_BUGS in r11466. But I didn't find where you check this flag. Is it maybe possible you forget to commit something? Thanks, Tamas I didn't find where you Anthony Minessale ?rta: > please test latest trunk. > Patch added to pause media bugs while not in a bridge which should pause > recordings and cut out the moh. > > > On Fri, Jan 23, 2009 at 10:45 AM, Tamas Cseke wrote: > > >> Hello, >> >> we would like to distribute calls with fifo and record these sessions >> but we'd like to skip the recording while the caller is waiting. >> (we don't need to record the hold music, just the speech with the fifo >> consumer.) >> >> I tried >> >> >> >> >> but it doesn't work because the channel is answered immediately when the >> caller is pushed into the fifo. >> (I don't know if there exists any other channel flag that could be use >> here) >> >> I also tried fifo_record_template. >> but it records the session from the point of view of the consumer's >> session, and after the bridge the recording is stopped. >> we would like to record the whole session into a single file even after >> calltransfers >> >> moreover we'd like to use some kind of predcitive dialing >> which >> 1, originate a loopback channel via event socket >> 2, loopback-b channel is hunting the dialplan, wich decide routing, >> caller_id, the need for recordings and so forth, and bridge a sofia call >> 3. the record_session is running on the sofia channel with >> bridge_pre_execute magic vars >> 4 loopback-a channel is pushed into the fifo >> 5 a script get the fifo::info via event socket >> 6 originate a call to the consumer with the proper strategy with &fifo >> out application >> 7 sofia channel is bridged to the consumer >> 8 loopback channels die >> >> after transfers everything is recorded into one file. >> but the problem here is again the unwanted recording in the fifo while >> the caller is waiting >> >> Could you please advise me any solution, if there is? >> >> >> Thank you, >> Tamas >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon Jan 26 02:01:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Jan 2009 02:01:26 -0800 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> Message-ID: <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> I have a TDM400 clone and I will see if I can reproduce these symptoms. BTW, are you in the Philippines? Is there any difference in the dial tone there than in the US? -MC On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon wrote: > i monitored the line using another phone. there's indeed dialtone in all > attempts. > i see TONE_DETECTED in the first call but i wonder there's a WARNING message > immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled > type for channel 2:1. > the dialtone freq should be okay since it's detected in the first call.could > the WARNING message gives us a hint of a possible problem other than the > dialtone freq? > > okay, i'll try the SVN version next. > > > On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale > wrote: >> >> Its not detecting a dial tone on the failure case. >> Before dialing it waits until it picks up dialtone. >> Try the svn trunk version to see if it works any better or verify there is >> a dialtone on the line. >> >> On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: >> >> hi everybody, >> >> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working using >> IP phones, softphones and digium FXS port. but there's a problem in dialing >> out to PSTN using digium tdm400 fxo - it works fine on the first attempt >> (after starting FS) but it fails on the subsequent attempts. i tested to >> call using the FXS port and IP phone. same problem. >> >> before i place any call, i checked >oz dump 2 1 (show current state = >> DOWN, last state = DOWN) >> >> in the first call, there's this message: >> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for channel >> 2:1 >> but >> >> then i hangup. checked >oz dump 21 (show current state=DOWN, last >> state=HANGUP) >> >> in the 2nd (and subsequent) attempts, the fxo just goes off-hook but >> doesn't send the dtmf tones. >> >oz dump 2 1 (shows current state = DIALING, last state = DOWN) >> >> has anyone encountered this problem before? i appreciate for any help to >> correct this problem. >> >> tks, >> nandy >> >> >> Environment: >> ================== >> kernel 2.6.18-92.1.22.el5 >> FS 1.0.2 >> zaptel 1.4.11 >> oslec >> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) >> >> zaptel.conf >> ======== >> loadzone = us >> defaultzone=us >> channels=1-2 >> alaw=1-4 >> fxsks=2 >> fxoks=1 >> >> >> openzap.conf.xml: >> =============== >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> openzap.conf >> ========== >> [span zt] >> name => OpenZAP FXS >> number => 1 >> fxs-channel => 1 >> >> [span zt] >> name => OpenZAP FXO >> number => 2 >> fxo-channel => 2 >> >> tones.conf (the dialtone and ring tone is set to Philipping tones) >> ======== >> [us] >> generate-dial => v=-7;%(1000,0,425) >> detect-dial => 425 >> >> generate-ring => v=-7;%(1000,4000,425,480) >> detect-ring => 425,480 >> >> generate-busy => v=-7;%(500,500,480,620) >> detect-busy => 480,620 >> >> generate-attn => v=0;%(200,300,1400,1800) >> detect-attn => 1400,1800 >> >> generate-callwaiting-sas => v=0;%(300,10000,440) >> detect-callwaiting-sas => 440 >> >> generate-callwaiting-cas => v=0;%(80,0,2750,2130) >> detect-callwaiting-cas => 2750,2130 >> >> detect-fail1 => 913.8 >> detect-fail2 => 1370.6 >> detect-fail3 => 776.7 >> >> LOG OF FIRST CALL (OK) >> ==================== >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >> bridge(openzap/2/1/3400534) >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU >> 20ms >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 channel_outgoing_channel() >> Connect outbound channel OpenZAP/2:1/3400534 >> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >> [e5f12114-ea88-11dd-9f5c-290fb4a527a4] >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 channel_outgoing_channel() >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 >> [BREAK] >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call() >> Changing state on 2:1 from DOWN to DIALING >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() >> ANALOG CHANNEL thread starting. >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_INIT >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> Executing state handler on 2:1 for DIALING >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 >> [BREAK] >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to sleep >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> CS_ROUTING >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() >> OpenZAP/2:1/3400534 CHANNEL ROUTING >> 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> >> CS_CONSUME_MEDIA >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 >> [BREAK] >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to sleep >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> CS_CONSUME_MEDIA >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >> 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 zap_analog_channel_run() >> Detected tone DIAL on 2:1 >> 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig >> 2:1 [TONE_DETECTED] >> 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled >> type for channel 2:1 >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 zchan_activate_dtmf_buffer() >> Created DTMF Buffer! >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 GENERATE DTMF >> [3400534] >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 zap_analog_channel_run() >> Changing state on 2:1 from DIALING to UP >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> Executing state handler on 2:1 for UP >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig >> 2:1 [UP] >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 >> switch_channel_perform_mark_answered() Send signal OpenZAP/1:1/93400534 >> [BREAK] >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() Channel >> [OpenZAP/2:1/3400534] has been answered >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message [AUDIO_SYNC] >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 >> switch_channel_perform_answer() OpenZAP/1:1/93400534 receive message >> [ANSWER] >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 >> channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 >> channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has been >> answered >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >> [AUDIO_SYNC] >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >> switch_core_session_perform_receive_message() Send signal >> OpenZAP/1:1/93400534 [BREAK] >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 >> switch_ivr_originate() Originate Resulted in Success: [OpenZAP/2:1/3400534] >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message [AUDIO_SYNC] >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >> [AUDIO_SYNC] >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 >> switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive message >> [BRIDGE] >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >> switch_core_session_perform_receive_message() Send signal >> OpenZAP/2:1/3400534 [BREAK] >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 >> switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive message >> [BRIDGE] >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >> switch_core_session_perform_receive_message() Send signal >> OpenZAP/1:1/93400534 [BREAK] >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 >> switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State Change >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 >> [BREAK] >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> Executing state handler on 1:1 for UP >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig >> [UP] >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA going to >> sleep >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> CS_EXCHANGE_MEDIA >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 channel_on_exchange_media() >> CHANNEL EXCHANGE_MEDIA >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() EVENT >> [ONHOOK][1:1] STATE [UP] >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() Changing >> state on 1:1 from UP to DOWN >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> Executing state handler on 1:1 for DOWN >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig >> [STOP] >> 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal OpenZAP/1:1/93400534 >> [BREAK] >> 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() channel done >> 1:1 >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() >> ANALOG CHANNEL 1:1 thread ended. >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 audio_bridge_thread() >> OpenZAP/1:1/93400534 ending bridge by request from read function >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 audio_bridge_thread() >> BRIDGE THREAD DONE [OpenZAP/1:1/93400534] >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 audio_bridge_thread() >> Send signal OpenZAP/2:1/3400534 [BREAK] >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 audio_bridge_thread() >> OpenZAP/1:1/93400534 ending bridge by request from write function >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 audio_bridge_thread() >> OpenZAP/2:1/3400534 receive message [UNBRIDGE] >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 >> switch_core_session_perform_receive_message() Send signal >> OpenZAP/2:1/3400534 [BREAK] >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 audio_bridge_thread() >> BRIDGE THREAD DONE [OpenZAP/2:1/3400534] >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 audio_bridge_thread() >> Send signal OpenZAP/1:1/93400534 [BREAK] >> 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 >> audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 >> [BREAK] >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA going >> to sleep >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> CS_HANGUP >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() Changing >> state on 2:1 from UP to HANGUP >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >> OpenZAP/2:1/3400534 CHANNEL HANGUP >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, cause: >> NORMAL_CLEARING >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to sleep >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Locked, Waiting >> on external entities >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to >> sleep >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 [CS_HANGUP] >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change >> CS_HANGUP >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >> OpenZAP/1:1/93400534 CHANNEL HANGUP >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, >> cause: NORMAL_CLEARING >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to sleep >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Locked, >> Waiting on external entities >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 [CS_HANGUP] >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> Executing state handler on 2:1 for HANGUP >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() >> Changing state on 2:1 from HANGUP to DOWN >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> Executing state handler on 2:1 for DOWN >> 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig >> 2:1 [STOP] >> 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel done >> 2:1 >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() >> ANALOG CHANNEL 2:1 thread ended. >> >> LOG OF FAILED CALLS >> ================== >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >> bridge(openzap/2/1/3400534) >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU >> 20ms >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 channel_outgoing_channel() >> Connect outbound channel OpenZAP/2:1/3400534 >> 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >> [079f5420-ea89-11dd-9f5c-290fb4a527a4] >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 channel_outgoing_channel() >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 >> [BREAK] >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call() >> Changing state on 2:1 from DOWN to DIALING >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() >> ANALOG CHANNEL thread starting. >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change CS_INIT >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> Executing state handler on 2:1 for DIALING >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 >> [BREAK] >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to sleep >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> CS_ROUTING >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() >> OpenZAP/2:1/3400534 CHANNEL ROUTING >> 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> >> CS_CONSUME_MEDIA >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 >> [BREAK] >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to sleep >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> CS_CONSUME_MEDIA >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() EVENT >> [ONHOOK][1:1] STATE [IDLE] >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() Changing >> state on 1:1 from IDLE to DOWN >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> Executing state handler on 1:1 for DOWN >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS sig >> [STOP] >> 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal OpenZAP/1:1/93400534 >> [BREAK] >> 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel done >> 1:1 >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() >> ANALOG CHANNEL 1:1 thread ended. >> 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 >> switch_ivr_originate() Hangup OpenZAP/2:1/3400534 [CS_CONSUME_MEDIA] >> [ORIGINATOR_CANCEL] >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >> switch_core_session_signal_state_change() Send signal OpenZAP/2:1/3400534 >> [BREAK] >> 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 >> switch_ivr_originate() Originate Cancelled by originator termination Cause: >> 487 [ORIGINATOR_CANCEL] >> 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 audio_bridge_function() >> Originate Failed. Cause: ORIGINATOR_CANCEL >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to >> sleep >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change >> CS_HANGUP >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >> OpenZAP/1:1/93400534 CHANNEL HANGUP >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, >> cause: NORMAL_CLEARING >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to sleep >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Locked, >> Waiting on external entities >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA going to >> sleep >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 [CS_HANGUP] >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> CS_HANGUP >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() Changing >> state on 2:1 from DIALING to HANGUP >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >> OpenZAP/2:1/3400534 CHANNEL HANGUP >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, cause: >> ORIGINATOR_CANCEL >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to sleep >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Locked, Waiting >> on external entities >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 [CS_HANGUP] >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> Executing state handler on 2:1 for HANGUP >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() >> Changing state on 2:1 from HANGUP to DOWN >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> Executing state handler on 2:1 for DOWN >> 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO sig >> 2:1 [STOP] >> 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel done >> 2:1 >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() >> ANALOG CHANNEL 2:1 thread ended. >> >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Mon Jan 26 02:03:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Jan 2009 02:03:24 -0800 Subject: [Freeswitch-users] ATA-answering machine question/recommendation In-Reply-To: <21640813.post@talk.nabble.com> References: <207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45@mail.gmail.com> <1461348542-1232566409-cardhu_decombobulator_blackberry.rim.net-569820458-@bxe161.bisx.prod.on.blackberry> <21640813.post@talk.nabble.com> Message-ID: <87f2f3b90901260203xd447883xe373244207c2fc45@mail.gmail.com> Did you ever post your dialplan and a debug trace of a call to the pastebin? If not, please do so and we will check it out. -MC On Sat, Jan 24, 2009 at 5:47 AM, shehzad p wrote: > > Hi all, > > On my existing Freeswitch 1.0.2, I installed and configured mod_vmd as > below: > make mod_vmd-install > > Then configured it as on wiki page: > http://wiki.freeswitch.org/wiki/Mod_vmd > > > After that my dialplan terminates call to another system, where it is just > answered and wait for some time there. > So that there should be a variable called vmd_detect must be created as > shown in dialplan http://wiki.freeswitch.org/wiki/Mod_vmd#From_Dialplan. > > But eventhough no variable named 'vmd_detect' is created after that.!!! > Is there i am missing something? Is there another way of using mod_vmd? > > Thanks in advance. > msp > > > > > > > Lucas Cornelisse wrote: >> >> Hi Jonathan, >> >> Mod_vmd (voicemail detection) should do the trick. >> >> Just search the wiki for mod_vmd, there are a number of ways of using it. >> >> >> Sent from my BlackBerry device on the Rogers Wireless Network >> >> -----Original Message----- >> From: jonathan augenstine >> >> Date: Wed, 21 Jan 2009 06:53:41 >> To: >> Subject: [Freeswitch-users] ATA-answering machine question/recommendation >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/ATA-answering-machine-question-recommendation-tp21584795p21640813.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From krivushinme at rn-inform.tomsk.ru Mon Jan 26 02:16:03 2009 From: krivushinme at rn-inform.tomsk.ru (=?utf-8?b?0JrRgNC40LLRg9GI0LjQvSDQnNC40YXQsNC40Ls=?=) Date: Mon, 26 Jan 2009 16:16:03 +0600 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <4979F75C.12740.BA74FA2@rehan.supertec.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <200901231249.19207.krivushinme@rn-inform.tomsk.ru> <4979F75C.12740.BA74FA2@rehan.supertec.com> Message-ID: <200901261616.03238.krivushinme@rn-inform.tomsk.ru> On Saturday 24 January 2009 05:59:08 Rehan Allah Wala wrote: > Spacibah Balshoi > > When are you making g723 for the Russians? > I'm so sorry, but g729 is only one we need. But you can do it yourself from free asterisk codec - it's not so hard, just see my code and compare it with mod_g723 and asterisk code. Work for one day. And I'm not programer. In fact I can do it, but not now. Time time time... -- ? ?????????, ???????? ?????? ??????? ?????????? ?????? ????????????????, ??? "??-??????" ?????? ? ?.??????, ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru mail: KrivushinME at rn-inform.tomsk.ru From sias at cpdata.co.za Mon Jan 26 02:25:46 2009 From: sias at cpdata.co.za (Sias Mey) Date: Mon, 26 Jan 2009 12:25:46 +0200 Subject: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches In-Reply-To: <20090126065518.GA6742@cpdata.co.za> References: <20090122202406.GA21067@cpdata.co.za> <191c3a030901221425p379ad74j7f866eee4596a6a9@mail.gmail.com> <20090123095036.GA6568@cpdata.co.za> <20090123100326.GA6830@cpdata.co.za> <191c3a030901230613t6335d8ic48e3ce22786803f@mail.gmail.com> <20090126065518.GA6742@cpdata.co.za> Message-ID: <20090126102546.GA9766@cpdata.co.za> On a similar note is it possible to use api commands from the dialplan. I would like a execute_on_answer to run a script in the same fasion, but I cant seem to get it to execute as a api command. From rehan at supertec.com Mon Jan 26 16:14:49 2009 From: rehan at supertec.com (Rehan Allah Wala) Date: Mon, 26 Jan 2009 17:14:49 -0700 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <200901261616.03238.krivushinme@rn-inform.tomsk.ru> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru>, <4979F75C.12740.BA74FA2@rehan.supertec.com>, <200901261616.03238.krivushinme@rn-inform.tomsk.ru> Message-ID: <497DEF89.6195.1B28F045@rehan.supertec.com> Great, well I am not a programmer and do not understand the code at all :) My Apology Rehan > On Saturday 24 January 2009 05:59:08 Rehan Allah Wala wrote: > > Spacibah Balshoi > > > > When are you making g723 for the Russians? > > > I'm so sorry, but g729 is only one we need. But you can do it yourself from > free asterisk codec - it's not so hard, just see my code and compare it with > mod_g723 and asterisk code. Work for one day. And I'm not programer. > > In fact I can do it, but not now. Time time time... > > -- > , > , > "-" ., > . . +7 913 865 78 66 > icq: 218 744 127 > xmpp: KrivushinME at jabber.ru > mail: KrivushinME at rn-inform.tomsk.ru > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Rehan Ahmed AllahWala Msn/Yahoo/GoogleTalk/Email: Rehan at Rehan.com http://www.supertec.com/ - Internet Telephony Solutions Http://www.DIDX.net - DID Number Market Place. Don't Remember Me ? Visit http://www.Rehan.com ~~~~~~~~~~~~~~~~~~~ "First they ignore you, then they laugh at you, then they fight you, then you win." By Gandhi. "Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi From nicholsonster at gmail.com Mon Jan 26 05:01:30 2009 From: nicholsonster at gmail.com (John Nicholson) Date: Mon, 26 Jan 2009 07:01:30 -0600 Subject: [Freeswitch-users] VMWare voice quality In-Reply-To: <496F9B3B.20509@shaw.ca> References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl> <87f2f3b90901150932r2d7bf58cxd4d63b30e464e48f@mail.gmail.com> <496F9B3B.20509@shaw.ca> Message-ID: <6275d5b40901260501l741444b3hc575d9cc1da98957@mail.gmail.com> I think the problem with changing the CPU's is off your going from a single to multi, or multi to single CPU and not adjusting the kernel from SMP to normal or vise versa? I've got a nice ESXi farm at the moment (32 cores, 48 gigs of ram) thats running at under 5% usage. Haven't noticed any timing issues yet with FS, but haven't put it under any type of load yet. Thanks for the release, I'll see if i can get you some feedback. On Thu, Jan 15, 2009 at 2:23 PM, Chav Paskov wrote: > Michael Collins wrote: > > If anyone figures this out please post it to this thread. I am working > > on a wiki page for the VMWare appliance and I would like to be able to > > inform people on how to handle this situation. > > > > Also, IIUC, those running VMWare Fusion on Macs are not experiencing > > this, correct? What about those using a hypervisor like ESXi? Any > > known issues? > > > > Thanks, > > MC > > > > On Thu, Jan 15, 2009 at 9:15 AM, Ken Rice wrote: > > > >> On 1/15/09 11:01 AM, "Remko Kloosterman" wrote: > >> > >> > >>> Hello Ken, hello all, > >>> > >>> I just read about the FreeSWITCH VMware applicance. I'm curious about > >>> your experiences with the audio quality on VMWare, so here's a new > >>> thread. > >>> > >>> I've installed freeswitch on VMware Server for Windows. The IVR audio > >>> always plays choppy, while the server itself has no performance issues. > >>> The same poor voice quality also goes for Asterisk or Yate, even on a > >>> very fast VMware ESX system. > >>> > >>> Did you experience the same and/or do you have pointers on how to > >>> troubleshoot and fix this? > >>> > >> There is a high resolution timer you need to enable on vmware... I'm not > >> familiar enuff with all the versions of vmware to advise there that > switch > >> is, but they have a couple of articles on it in their knowledge base > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > Hi All, > I'm using freeswitch in production environment running on ESXi . I have > no issues with voice /probably because simply i leave the media to flow > between endpoints/ . Performance is amazing and i'd recommend this setup > to everybody. > it is important though when you set your VM on ESXi to set in advance > the number of CPUs. Changing # of CPUs later might affect your > performance. My recommendation is NOT to use VMWARE server on top of > other OS. ESXi as hipervisor is linux in its core that provides you > with enough access to the HW and nothing more so the overhead is as > minimal as possible /while this is not the case fro VMware server - it > needs underlaying OS and so on/. > I hope this info helps. > If anybody is interested i'd be glad to share me experience on his > matter. > Best Regards > Chav > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/8916df78/attachment.html From gcd at i.ph Mon Jan 26 05:32:16 2009 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 26 Jan 2009 21:32:16 +0800 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> Message-ID: <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> that's great. yes, i'm in the philippines. there's a difference in dialtone - it's 425 Hz. -nandy On Mon, Jan 26, 2009 at 6:01 PM, Michael Collins wrote: > I have a TDM400 clone and I will see if I can reproduce these > symptoms. BTW, are you in the Philippines? Is there any difference in > the dial tone there than in the US? > -MC > > On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon wrote: > > i monitored the line using another phone. there's indeed dialtone in all > > attempts. > > i see TONE_DETECTED in the first call but i wonder there's a WARNING > message > > immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() > Unhandled > > type for channel 2:1. > > the dialtone freq should be okay since it's detected in the first > call.could > > the WARNING message gives us a hint of a possible problem other than the > > dialtone freq? > > > > okay, i'll try the SVN version next. > > > > > > On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale > > wrote: > >> > >> Its not detecting a dial tone on the failure case. > >> Before dialing it waits until it picks up dialtone. > >> Try the svn trunk version to see if it works any better or verify there > is > >> a dialtone on the line. > >> > >> On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: > >> > >> hi everybody, > >> > >> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working > using > >> IP phones, softphones and digium FXS port. but there's a problem in > dialing > >> out to PSTN using digium tdm400 fxo - it works fine on the first attempt > >> (after starting FS) but it fails on the subsequent attempts. i tested to > >> call using the FXS port and IP phone. same problem. > >> > >> before i place any call, i checked >oz dump 2 1 (show current state = > >> DOWN, last state = DOWN) > >> > >> in the first call, there's this message: > >> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for channel > >> 2:1 > >> but > >> > >> then i hangup. checked >oz dump 21 (show current state=DOWN, last > >> state=HANGUP) > >> > >> in the 2nd (and subsequent) attempts, the fxo just goes off-hook but > >> doesn't send the dtmf tones. > >> >oz dump 2 1 (shows current state = DIALING, last state = DOWN) > >> > >> has anyone encountered this problem before? i appreciate for any help to > >> correct this problem. > >> > >> tks, > >> nandy > >> > >> > >> Environment: > >> ================== > >> kernel 2.6.18-92.1.22.el5 > >> FS 1.0.2 > >> zaptel 1.4.11 > >> oslec > >> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) > >> > >> zaptel.conf > >> ======== > >> loadzone = us > >> defaultzone=us > >> channels=1-2 > >> alaw=1-4 > >> fxsks=2 > >> fxoks=1 > >> > >> > >> openzap.conf.xml: > >> =============== > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> openzap.conf > >> ========== > >> [span zt] > >> name => OpenZAP FXS > >> number => 1 > >> fxs-channel => 1 > >> > >> [span zt] > >> name => OpenZAP FXO > >> number => 2 > >> fxo-channel => 2 > >> > >> tones.conf (the dialtone and ring tone is set to Philipping tones) > >> ======== > >> [us] > >> generate-dial => v=-7;%(1000,0,425) > >> detect-dial => 425 > >> > >> generate-ring => v=-7;%(1000,4000,425,480) > >> detect-ring => 425,480 > >> > >> generate-busy => v=-7;%(500,500,480,620) > >> detect-busy => 480,620 > >> > >> generate-attn => v=0;%(200,300,1400,1800) > >> detect-attn => 1400,1800 > >> > >> generate-callwaiting-sas => v=0;%(300,10000,440) > >> detect-callwaiting-sas => 440 > >> > >> generate-callwaiting-cas => v=0;%(80,0,2750,2130) > >> detect-callwaiting-cas => 2750,2130 > >> > >> detect-fail1 => 913.8 > >> detect-fail2 => 1370.6 > >> detect-fail3 => 776.7 > >> > >> LOG OF FIRST CALL (OK) > >> ==================== > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 > >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute > >> bridge(openzap/2/1/3400534) > >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU > >> 20ms > >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 > channel_outgoing_channel() > >> Connect outbound channel OpenZAP/2:1/3400534 > >> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 > >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 > >> [e5f12114-ea88-11dd-9f5c-290fb4a527a4] > >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 > channel_outgoing_channel() > >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT > >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call() > >> Changing state on 2:1 from DOWN to DIALING > >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() > >> ANALOG CHANNEL thread starting. > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > CS_INIT > >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > >> Executing state handler on 2:1 for DIALING > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT > >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() > >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING > >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to > sleep > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > >> CS_ROUTING > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING > >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() > >> OpenZAP/2:1/3400534 CHANNEL ROUTING > >> 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 > >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> > >> CS_CONSUME_MEDIA > >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to > sleep > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > >> CS_CONSUME_MEDIA > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA > >> 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 zap_analog_channel_run() > >> Detected tone DIAL on 2:1 > >> 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO > sig > >> 2:1 [TONE_DETECTED] > >> 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() > Unhandled > >> type for channel 2:1 > >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 zchan_activate_dtmf_buffer() > >> Created DTMF Buffer! > >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 GENERATE > DTMF > >> [3400534] > >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 zap_analog_channel_run() > >> Changing state on 2:1 from DIALING to UP > >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > >> Executing state handler on 2:1 for UP > >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO > sig > >> 2:1 [UP] > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 > >> switch_channel_perform_mark_answered() Send signal OpenZAP/1:1/93400534 > >> [BREAK] > >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() Channel > >> [OpenZAP/2:1/3400534] has been answered > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message > [AUDIO_SYNC] > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 > >> switch_channel_perform_answer() OpenZAP/1:1/93400534 receive message > >> [ANSWER] > >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 > >> channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP > >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 > >> channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has been > >> answered > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message > >> [AUDIO_SYNC] > >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 > >> switch_core_session_perform_receive_message() Send signal > >> OpenZAP/1:1/93400534 [BREAK] > >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 > >> switch_ivr_originate() Originate Resulted in Success: > [OpenZAP/2:1/3400534] > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message > [AUDIO_SYNC] > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message > >> [AUDIO_SYNC] > >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 > >> switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive message > >> [BRIDGE] > >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 > >> switch_core_session_perform_receive_message() Send signal > >> OpenZAP/2:1/3400534 [BREAK] > >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 > >> switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive message > >> [BRIDGE] > >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 > >> switch_core_session_perform_receive_message() Send signal > >> OpenZAP/1:1/93400534 [BREAK] > >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 > >> switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State Change > >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > >> Executing state handler on 1:1 for UP > >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS > sig > >> [UP] > >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA > going to > >> sleep > >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > >> CS_EXCHANGE_MEDIA > >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA > >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 > channel_on_exchange_media() > >> CHANNEL EXCHANGE_MEDIA > >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() EVENT > >> [ONHOOK][1:1] STATE [UP] > >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() Changing > >> state on 1:1 from UP to DOWN > >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > >> Executing state handler on 1:1 for DOWN > >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS > sig > >> [STOP] > >> 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup > >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] > >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 > >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] > >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/1:1/93400534 > >> [BREAK] > >> 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done > >> 1:1 > >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() > >> ANALOG CHANNEL 1:1 thread ended. > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 > audio_bridge_thread() > >> OpenZAP/1:1/93400534 ending bridge by request from read function > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 > audio_bridge_thread() > >> BRIDGE THREAD DONE [OpenZAP/1:1/93400534] > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 > audio_bridge_thread() > >> Send signal OpenZAP/2:1/3400534 [BREAK] > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 > audio_bridge_thread() > >> OpenZAP/1:1/93400534 ending bridge by request from write function > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 > audio_bridge_thread() > >> OpenZAP/2:1/3400534 receive message [UNBRIDGE] > >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 > >> switch_core_session_perform_receive_message() Send signal > >> OpenZAP/2:1/3400534 [BREAK] > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 > audio_bridge_thread() > >> BRIDGE THREAD DONE [OpenZAP/2:1/3400534] > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 > audio_bridge_thread() > >> Send signal OpenZAP/1:1/93400534 [BREAK] > >> 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 > >> audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 > >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 > >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] > >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA > going > >> to sleep > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > >> CS_HANGUP > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP > >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() > Changing > >> state on 2:1 from UP to HANGUP > >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() > >> OpenZAP/2:1/3400534 CHANNEL HANGUP > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 > >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, > cause: > >> NORMAL_CLEARING > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to > sleep > >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 > >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Locked, > Waiting > >> on external entities > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to > >> sleep > >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 > >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended > >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 > >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 > [CS_HANGUP] > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change > >> CS_HANGUP > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP > >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() > >> OpenZAP/1:1/93400534 CHANNEL HANGUP > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 > >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, > >> cause: NORMAL_CLEARING > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to > sleep > >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 > >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Locked, > >> Waiting on external entities > >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 > >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended > >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 > >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 > [CS_HANGUP] > >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > >> Executing state handler on 2:1 for HANGUP > >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() > >> Changing state on 2:1 from HANGUP to DOWN > >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > >> Executing state handler on 2:1 for DOWN > >> 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO > sig > >> 2:1 [STOP] > >> 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done > >> 2:1 > >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() > >> ANALOG CHANNEL 2:1 thread ended. > >> > >> LOG OF FAILED CALLS > >> ================== > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 > >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute > >> bridge(openzap/2/1/3400534) > >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU > >> 20ms > >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 > channel_outgoing_channel() > >> Connect outbound channel OpenZAP/2:1/3400534 > >> 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 > >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 > >> [079f5420-ea89-11dd-9f5c-290fb4a527a4] > >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 > channel_outgoing_channel() > >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT > >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 analog_fxo_outgoing_call() > >> Changing state on 2:1 from DOWN to DIALING > >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() > >> ANALOG CHANNEL thread starting. > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > CS_INIT > >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > >> Executing state handler on 2:1 for DIALING > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT > >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() > >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING > >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to > sleep > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > >> CS_ROUTING > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING > >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() > >> OpenZAP/2:1/3400534 CHANNEL ROUTING > >> 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 > >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> > >> CS_CONSUME_MEDIA > >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to > sleep > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > >> CS_CONSUME_MEDIA > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA > >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() EVENT > >> [ONHOOK][1:1] STATE [IDLE] > >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() Changing > >> state on 1:1 from IDLE to DOWN > >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > >> Executing state handler on 1:1 for DOWN > >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS > sig > >> [STOP] > >> 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup > >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] > >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 > >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] > >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/1:1/93400534 > >> [BREAK] > >> 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done > >> 1:1 > >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() > >> ANALOG CHANNEL 1:1 thread ended. > >> 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 > >> switch_ivr_originate() Hangup OpenZAP/2:1/3400534 [CS_CONSUME_MEDIA] > >> [ORIGINATOR_CANCEL] > >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 > >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] > >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 > >> switch_ivr_originate() Originate Cancelled by originator termination > Cause: > >> 487 [ORIGINATOR_CANCEL] > >> 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 audio_bridge_function() > >> Originate Failed. Cause: ORIGINATOR_CANCEL > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to > >> sleep > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change > >> CS_HANGUP > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP > >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() > >> OpenZAP/1:1/93400534 CHANNEL HANGUP > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 > >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, > >> cause: NORMAL_CLEARING > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to > sleep > >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 > >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Locked, > >> Waiting on external entities > >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 > >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA > going to > >> sleep > >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 > >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 > [CS_HANGUP] > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change > >> CS_HANGUP > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP > >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() > Changing > >> state on 2:1 from DIALING to HANGUP > >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() > >> OpenZAP/2:1/3400534 CHANNEL HANGUP > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 > >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, > cause: > >> ORIGINATOR_CANCEL > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to > sleep > >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 > >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Locked, > Waiting > >> on external entities > >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 > >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended > >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 > >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 > [CS_HANGUP] > >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > >> Executing state handler on 2:1 for HANGUP > >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() > >> Changing state on 2:1 from HANGUP to DOWN > >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() > >> Executing state handler on 2:1 for DOWN > >> 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO > sig > >> 2:1 [STOP] > >> 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel > done > >> 2:1 > >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() > >> ANALOG CHANNEL 2:1 thread ended. > >> > >> > >> > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/107b9d86/attachment-0001.html From testeador01 at gmail.com Mon Jan 26 05:33:33 2009 From: testeador01 at gmail.com (Milena) Date: Mon, 26 Jan 2009 08:33:33 -0500 Subject: [Freeswitch-users] My fs_cli stopped working, can you help me please? Message-ID: Good morning Ok, here is what i get from the console, do you know what can i do to fix it? thank you very much -bash-3.2# /usr/local/freeswitch/bin/fs_cli -d 7 [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration file is /root/.fs_cli_conf. [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration file is /etc/fs_cli.conf. [DEBUG] libs/esl/fs_cli.c:573 main() profile default does not exist using builtin profile [DEBUG] libs/esl/fs_cli.c:597 main() Using profile internal [127.0.0.1] [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Content-Type] = [auth/request] [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE Content-Type: [auth/request] [DEBUG] libs/esl/src/esl.c:853 esl_send() SEND auth ClueCon [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Reply-Text] = [-ERR invalid] [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE Content-Type: [command/reply] Reply-Text: [-ERR invalid] [ERROR] libs/esl/fs_cli.c:610 main() Error Connecting [Connection Error] -bash-3.2# Anthony Minessale wrote: > > Try running > > fs_cli -d 7 > > Maybe the debug log will shed some light. > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/eec6a105/attachment.html From cstomi.levlist at gmail.com Mon Jan 26 08:11:53 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Mon, 26 Jan 2009 17:11:53 +0100 Subject: [Freeswitch-users] record session in fifo In-Reply-To: <497D8782.2070603@gmail.com> References: <4979F41F.70605@gmail.com> <191c3a030901231045m784f0cd7s6994b763d02b3e26@mail.gmail.com> <497D8782.2070603@gmail.com> Message-ID: <497DE0C9.5090005@gmail.com> Hello, I tested with the attached patch. It is working fine in a normal case. I have only problems with the automatic calls, because in this case the loopback channel is in the fifo, but the record_session is running on the sofia channel. Maybe it could be sort out with putting the bug pause/resume functions into api function, what I should turn on and off on demand? Anyway, I quess this is a bit extreme circumstance, and it isn't so important to us now. Thanks, Tamas Tamas Cseke ?rta: > Hello, > > Thank you your help. > > I tested with r11489, but moh is still recorded in fifo. > > I quess you I should test the CF_PAUSE_BUGS in r11466. > But I didn't find where you check this flag. > Is it maybe possible you forget to commit something? > > Thanks, > Tamas > > > I didn't find where you > Anthony Minessale ?rta: > >> please test latest trunk. >> Patch added to pause media bugs while not in a bridge which should pause >> recordings and cut out the moh. >> >> >> On Fri, Jan 23, 2009 at 10:45 AM, Tamas Cseke wrote: >> >> >> >>> Hello, >>> >>> we would like to distribute calls with fifo and record these sessions >>> but we'd like to skip the recording while the caller is waiting. >>> (we don't need to record the hold music, just the speech with the fifo >>> consumer.) >>> >>> I tried >>> >>> >>> >>> >>> but it doesn't work because the channel is answered immediately when the >>> caller is pushed into the fifo. >>> (I don't know if there exists any other channel flag that could be use >>> here) >>> >>> I also tried fifo_record_template. >>> but it records the session from the point of view of the consumer's >>> session, and after the bridge the recording is stopped. >>> we would like to record the whole session into a single file even after >>> calltransfers >>> >>> moreover we'd like to use some kind of predcitive dialing >>> which >>> 1, originate a loopback channel via event socket >>> 2, loopback-b channel is hunting the dialplan, wich decide routing, >>> caller_id, the need for recordings and so forth, and bridge a sofia call >>> 3. the record_session is running on the sofia channel with >>> bridge_pre_execute magic vars >>> 4 loopback-a channel is pushed into the fifo >>> 5 a script get the fifo::info via event socket >>> 6 originate a call to the consumer with the proper strategy with &fifo >>> out application >>> 7 sofia channel is bridged to the consumer >>> 8 loopback channels die >>> >>> after transfers everything is recorded into one file. >>> but the problem here is again the unwanted recording in the fifo while >>> the caller is waiting >>> >>> Could you please advise me any solution, if there is? >>> >>> >>> Thank you, >>> Tamas >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- A non-text attachment was scrubbed... Name: pause_bugs.patch Type: text/x-patch Size: 473 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/bb7063b8/attachment.bin From msc at freeswitch.org Mon Jan 26 08:32:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Jan 2009 08:32:56 -0800 Subject: [Freeswitch-users] Conference javascript and hanuphooks giving me headaches In-Reply-To: <20090126102546.GA9766@cpdata.co.za> References: <20090122202406.GA21067@cpdata.co.za> <191c3a030901221425p379ad74j7f866eee4596a6a9@mail.gmail.com> <20090123095036.GA6568@cpdata.co.za> <20090123100326.GA6830@cpdata.co.za> <191c3a030901230613t6335d8ic48e3ce22786803f@mail.gmail.com> <20090126065518.GA6742@cpdata.co.za> <20090126102546.GA9766@cpdata.co.za> Message-ID: <87f2f3b90901260832s6808ec3aw54bdf259a664b15d@mail.gmail.com> What have you tried? -MC On Mon, Jan 26, 2009 at 2:25 AM, Sias Mey wrote: > On a similar note is it possible to use api commands from the dialplan. > > I would like a execute_on_answer to run a script in the same fasion, but > I cant seem to get it to execute as a api command. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Mon Jan 26 08:33:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 26 Jan 2009 10:33:51 -0600 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> Message-ID: <191c3a030901260833l2e07bfffkca9400d33a5483fe@mail.gmail.com> The unhanded type message just means that mod_openzap does not do anything with the TONE_DETECTED event that was passed up from the ozmod_analog. On Mon, Jan 26, 2009 at 7:32 AM, Nandy Dagondon wrote: > that's great. yes, i'm in the philippines. there's a difference in dialtone > - it's 425 Hz. > -nandy > > > > On Mon, Jan 26, 2009 at 6:01 PM, Michael Collins wrote: > >> I have a TDM400 clone and I will see if I can reproduce these >> symptoms. BTW, are you in the Philippines? Is there any difference in >> the dial tone there than in the US? >> -MC >> >> On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon wrote: >> > i monitored the line using another phone. there's indeed dialtone in all >> > attempts. >> > i see TONE_DETECTED in the first call but i wonder there's a WARNING >> message >> > immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() >> Unhandled >> > type for channel 2:1. >> > the dialtone freq should be okay since it's detected in the first >> call.could >> > the WARNING message gives us a hint of a possible problem other than the >> > dialtone freq? >> > >> > okay, i'll try the SVN version next. >> > >> > >> > On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale >> > wrote: >> >> >> >> Its not detecting a dial tone on the failure case. >> >> Before dialing it waits until it picks up dialtone. >> >> Try the svn trunk version to see if it works any better or verify there >> is >> >> a dialtone on the line. >> >> >> >> On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: >> >> >> >> hi everybody, >> >> >> >> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working >> using >> >> IP phones, softphones and digium FXS port. but there's a problem in >> dialing >> >> out to PSTN using digium tdm400 fxo - it works fine on the first >> attempt >> >> (after starting FS) but it fails on the subsequent attempts. i tested >> to >> >> call using the FXS port and IP phone. same problem. >> >> >> >> before i place any call, i checked >oz dump 2 1 (show current state = >> >> DOWN, last state = DOWN) >> >> >> >> in the first call, there's this message: >> >> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for channel >> >> 2:1 >> >> but >> >> >> >> then i hangup. checked >oz dump 21 (show current state=DOWN, last >> >> state=HANGUP) >> >> >> >> in the 2nd (and subsequent) attempts, the fxo just goes off-hook but >> >> doesn't send the dtmf tones. >> >> >oz dump 2 1 (shows current state = DIALING, last state = DOWN) >> >> >> >> has anyone encountered this problem before? i appreciate for any help >> to >> >> correct this problem. >> >> >> >> tks, >> >> nandy >> >> >> >> >> >> Environment: >> >> ================== >> >> kernel 2.6.18-92.1.22.el5 >> >> FS 1.0.2 >> >> zaptel 1.4.11 >> >> oslec >> >> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) >> >> >> >> zaptel.conf >> >> ======== >> >> loadzone = us >> >> defaultzone=us >> >> channels=1-2 >> >> alaw=1-4 >> >> fxsks=2 >> >> fxoks=1 >> >> >> >> >> >> openzap.conf.xml: >> >> =============== >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> openzap.conf >> >> ========== >> >> [span zt] >> >> name => OpenZAP FXS >> >> number => 1 >> >> fxs-channel => 1 >> >> >> >> [span zt] >> >> name => OpenZAP FXO >> >> number => 2 >> >> fxo-channel => 2 >> >> >> >> tones.conf (the dialtone and ring tone is set to Philipping tones) >> >> ======== >> >> [us] >> >> generate-dial => v=-7;%(1000,0,425) >> >> detect-dial => 425 >> >> >> >> generate-ring => v=-7;%(1000,4000,425,480) >> >> detect-ring => 425,480 >> >> >> >> generate-busy => v=-7;%(500,500,480,620) >> >> detect-busy => 480,620 >> >> >> >> generate-attn => v=0;%(200,300,1400,1800) >> >> detect-attn => 1400,1800 >> >> >> >> generate-callwaiting-sas => v=0;%(300,10000,440) >> >> detect-callwaiting-sas => 440 >> >> >> >> generate-callwaiting-cas => v=0;%(80,0,2750,2130) >> >> detect-callwaiting-cas => 2750,2130 >> >> >> >> detect-fail1 => 913.8 >> >> detect-fail2 => 1370.6 >> >> detect-fail3 => 776.7 >> >> >> >> LOG OF FIRST CALL (OK) >> >> ==================== >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 >> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >> >> bridge(openzap/2/1/3400534) >> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec >> PCMU >> >> 20ms >> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 >> channel_outgoing_channel() >> >> Connect outbound channel OpenZAP/2:1/3400534 >> >> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 >> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >> >> [e5f12114-ea88-11dd-9f5c-290fb4a527a4] >> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 >> channel_outgoing_channel() >> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >> >> switch_core_session_signal_state_change() Send signal >> OpenZAP/2:1/3400534 >> >> [BREAK] >> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 >> analog_fxo_outgoing_call() >> >> Changing state on 2:1 from DOWN to DIALING >> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() >> >> ANALOG CHANNEL thread starting. >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> CS_INIT >> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> >> Executing state handler on 2:1 for DIALING >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() >> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >> >> switch_core_session_signal_state_change() Send signal >> OpenZAP/2:1/3400534 >> >> [BREAK] >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >> sleep >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> >> CS_ROUTING >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() >> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >> >> 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 >> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> >> >> CS_CONSUME_MEDIA >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >> >> switch_core_session_signal_state_change() Send signal >> OpenZAP/2:1/3400534 >> >> [BREAK] >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to >> sleep >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> >> CS_CONSUME_MEDIA >> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >> >> 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 zap_analog_channel_run() >> >> Detected tone DIAL on 2:1 >> >> 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO >> sig >> >> 2:1 [TONE_DETECTED] >> >> 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() >> Unhandled >> >> type for channel 2:1 >> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 zchan_activate_dtmf_buffer() >> >> Created DTMF Buffer! >> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 GENERATE >> DTMF >> >> [3400534] >> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 zap_analog_channel_run() >> >> Changing state on 2:1 from DIALING to UP >> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> >> Executing state handler on 2:1 for UP >> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO >> sig >> >> 2:1 [UP] >> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 >> >> switch_channel_perform_mark_answered() Send signal OpenZAP/1:1/93400534 >> >> [BREAK] >> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() Channel >> >> [OpenZAP/2:1/3400534] has been answered >> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >> [AUDIO_SYNC] >> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 >> >> switch_channel_perform_answer() OpenZAP/1:1/93400534 receive message >> >> [ANSWER] >> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 >> >> channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP >> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 >> >> channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has been >> >> answered >> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >> >> [AUDIO_SYNC] >> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >> >> switch_core_session_perform_receive_message() Send signal >> >> OpenZAP/1:1/93400534 [BREAK] >> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 >> >> switch_ivr_originate() Originate Resulted in Success: >> [OpenZAP/2:1/3400534] >> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >> [AUDIO_SYNC] >> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >> >> [AUDIO_SYNC] >> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 >> >> switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive message >> >> [BRIDGE] >> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >> >> switch_core_session_perform_receive_message() Send signal >> >> OpenZAP/2:1/3400534 [BREAK] >> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 >> >> switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive message >> >> [BRIDGE] >> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >> >> switch_core_session_perform_receive_message() Send signal >> >> OpenZAP/1:1/93400534 [BREAK] >> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 >> >> switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State Change >> >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 >> >> switch_core_session_signal_state_change() Send signal >> OpenZAP/2:1/3400534 >> >> [BREAK] >> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> >> Executing state handler on 1:1 for UP >> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS >> sig >> >> [UP] >> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >> going to >> >> sleep >> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> >> CS_EXCHANGE_MEDIA >> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 >> channel_on_exchange_media() >> >> CHANNEL EXCHANGE_MEDIA >> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() EVENT >> >> [ONHOOK][1:1] STATE [UP] >> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() Changing >> >> state on 1:1 from UP to DOWN >> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> >> Executing state handler on 1:1 for DOWN >> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS >> sig >> >> [STOP] >> >> 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup >> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >> >> switch_core_session_signal_state_change() Send signal >> OpenZAP/1:1/93400534 >> >> [BREAK] >> >> 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done >> >> 1:1 >> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() >> >> ANALOG CHANNEL 1:1 thread ended. >> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 >> audio_bridge_thread() >> >> OpenZAP/1:1/93400534 ending bridge by request from read function >> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >> audio_bridge_thread() >> >> BRIDGE THREAD DONE [OpenZAP/1:1/93400534] >> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >> audio_bridge_thread() >> >> Send signal OpenZAP/2:1/3400534 [BREAK] >> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 >> audio_bridge_thread() >> >> OpenZAP/1:1/93400534 ending bridge by request from write function >> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 >> audio_bridge_thread() >> >> OpenZAP/2:1/3400534 receive message [UNBRIDGE] >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 >> >> switch_core_session_perform_receive_message() Send signal >> >> OpenZAP/2:1/3400534 [BREAK] >> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >> audio_bridge_thread() >> >> BRIDGE THREAD DONE [OpenZAP/2:1/3400534] >> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >> audio_bridge_thread() >> >> Send signal OpenZAP/1:1/93400534 [BREAK] >> >> 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 >> >> audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 >> >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >> >> switch_core_session_signal_state_change() Send signal >> OpenZAP/2:1/3400534 >> >> [BREAK] >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >> going >> >> to sleep >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> >> CS_HANGUP >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() >> Changing >> >> state on 2:1 from UP to HANGUP >> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, >> cause: >> >> NORMAL_CLEARING >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to >> sleep >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Locked, >> Waiting >> >> on external entities >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 >> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to >> >> sleep >> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended >> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >> [CS_HANGUP] >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change >> >> CS_HANGUP >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, >> >> cause: NORMAL_CLEARING >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to >> sleep >> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Locked, >> >> Waiting on external entities >> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended >> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >> [CS_HANGUP] >> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> >> Executing state handler on 2:1 for HANGUP >> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() >> >> Changing state on 2:1 from HANGUP to DOWN >> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> >> Executing state handler on 2:1 for DOWN >> >> 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO >> sig >> >> 2:1 [STOP] >> >> 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done >> >> 2:1 >> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() >> >> ANALOG CHANNEL 2:1 thread ended. >> >> >> >> LOG OF FAILED CALLS >> >> ================== >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 >> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >> >> bridge(openzap/2/1/3400534) >> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec >> PCMU >> >> 20ms >> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 >> channel_outgoing_channel() >> >> Connect outbound channel OpenZAP/2:1/3400534 >> >> 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 >> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >> >> [079f5420-ea89-11dd-9f5c-290fb4a527a4] >> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 >> channel_outgoing_channel() >> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >> >> switch_core_session_signal_state_change() Send signal >> OpenZAP/2:1/3400534 >> >> [BREAK] >> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 >> analog_fxo_outgoing_call() >> >> Changing state on 2:1 from DOWN to DIALING >> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() >> >> ANALOG CHANNEL thread starting. >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> CS_INIT >> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> >> Executing state handler on 2:1 for DIALING >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() >> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >> >> switch_core_session_signal_state_change() Send signal >> OpenZAP/2:1/3400534 >> >> [BREAK] >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >> sleep >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> >> CS_ROUTING >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() >> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >> >> 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 >> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING -> >> >> CS_CONSUME_MEDIA >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >> >> switch_core_session_signal_state_change() Send signal >> OpenZAP/2:1/3400534 >> >> [BREAK] >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to >> sleep >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> >> CS_CONSUME_MEDIA >> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() EVENT >> >> [ONHOOK][1:1] STATE [IDLE] >> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() Changing >> >> state on 1:1 from IDLE to DOWN >> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> >> Executing state handler on 1:1 for DOWN >> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS >> sig >> >> [STOP] >> >> 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup >> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 [KILL] >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >> >> switch_core_session_signal_state_change() Send signal >> OpenZAP/1:1/93400534 >> >> [BREAK] >> >> 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done >> >> 1:1 >> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() >> >> ANALOG CHANNEL 1:1 thread ended. >> >> 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 >> >> switch_ivr_originate() Hangup OpenZAP/2:1/3400534 [CS_CONSUME_MEDIA] >> >> [ORIGINATOR_CANCEL] >> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >> >> switch_core_session_signal_state_change() Send signal >> OpenZAP/2:1/3400534 >> >> [BREAK] >> >> 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 >> >> switch_ivr_originate() Originate Cancelled by originator termination >> Cause: >> >> 487 [ORIGINATOR_CANCEL] >> >> 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 audio_bridge_function() >> >> Originate Failed. Cause: ORIGINATOR_CANCEL >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 >> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going to >> >> sleep >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change >> >> CS_HANGUP >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, >> >> cause: NORMAL_CLEARING >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to >> sleep >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Locked, >> >> Waiting on external entities >> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >> going to >> >> sleep >> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >> [CS_HANGUP] >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >> >> CS_HANGUP >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() >> Changing >> >> state on 2:1 from DIALING to HANGUP >> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, >> cause: >> >> ORIGINATOR_CANCEL >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to >> sleep >> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Locked, >> Waiting >> >> on external entities >> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended >> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >> [CS_HANGUP] >> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> >> Executing state handler on 2:1 for HANGUP >> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() >> >> Changing state on 2:1 from HANGUP to DOWN >> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() >> >> Executing state handler on 2:1 for DOWN >> >> 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO >> sig >> >> 2:1 [STOP] >> >> 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel >> done >> >> 2:1 >> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() >> >> ANALOG CHANNEL 2:1 thread ended. >> >> >> >> >> >> >> >> >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at 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 at 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 at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/5376149d/attachment-0001.html From mrene_lists at avgs.ca Mon Jan 26 08:39:23 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 26 Jan 2009 11:39:23 -0500 Subject: [Freeswitch-users] My fs_cli stopped working, can you help me please? In-Reply-To: References: Message-ID: Loos like the wrong password to me, look in /usr/local/freeswitch/conf/autoload_configs/event_socket.conf.xml and use fs_cli -p [pass] Mathieu On Mon, Jan 26, 2009 at 8:33 AM, Milena wrote: > Good morning > Ok, here is what i get from the console, do you know what can i do to fix > it? thank you very much > > -bash-3.2# /usr/local/freeswitch/bin/fs_cli -d 7 > [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration > file is /root/.fs_cli_conf. > [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration > file is /etc/fs_cli.conf. > [DEBUG] libs/esl/fs_cli.c:573 main() profile default does not exist using > builtin profile > [DEBUG] libs/esl/fs_cli.c:597 main() Using profile internal [127.0.0.1] > [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Content-Type] > = [auth/request] > [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE > Content-Type: [auth/request] > > > [DEBUG] libs/esl/src/esl.c:853 esl_send() SEND > auth ClueCon > > > [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Content-Type] > = [command/reply] > [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Reply-Text] = > [-ERR invalid] > [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE > Content-Type: [command/reply] > Reply-Text: [-ERR invalid] > > > [ERROR] libs/esl/fs_cli.c:610 main() Error Connecting [Connection Error] > -bash-3.2# > > > > > Anthony Minessale wrote: > >> >> Try running >> >> fs_cli -d 7 >> >> Maybe the debug log will shed some light. >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/a392df5c/attachment.html From testeador01 at gmail.com Mon Jan 26 08:46:26 2009 From: testeador01 at gmail.com (Milena) Date: Mon, 26 Jan 2009 11:46:26 -0500 Subject: [Freeswitch-users] My fs_cli stopped working, can you help me please? In-Reply-To: References: Message-ID: thanks ^^ 2009/1/26 Mathieu Rene > Loos like the wrong password to me, look in > /usr/local/freeswitch/conf/autoload_configs/event_socket.conf.xml and use > fs_cli -p [pass] > > > Mathieu > > On Mon, Jan 26, 2009 at 8:33 AM, Milena wrote: > >> Good morning >> Ok, here is what i get from the console, do you know what can i do to fix >> it? thank you very much >> >> -bash-3.2# /usr/local/freeswitch/bin/fs_cli -d 7 >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration >> file is /root/.fs_cli_conf. >> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration >> file is /etc/fs_cli.conf. >> [DEBUG] libs/esl/fs_cli.c:573 main() profile default does not exist using >> builtin profile >> [DEBUG] libs/esl/fs_cli.c:597 main() Using profile internal [127.0.0.1] >> [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Content-Type] >> = [auth/request] >> [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE >> Content-Type: [auth/request] >> >> >> [DEBUG] libs/esl/src/esl.c:853 esl_send() SEND >> auth ClueCon >> >> >> [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Content-Type] >> = [command/reply] >> [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Reply-Text] = >> [-ERR invalid] >> [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE >> Content-Type: [command/reply] >> Reply-Text: [-ERR invalid] >> >> >> [ERROR] libs/esl/fs_cli.c:610 main() Error Connecting [Connection Error] >> -bash-3.2# >> >> >> >> >> Anthony Minessale wrote: >> >>> >>> Try running >>> >>> fs_cli -d 7 >>> >>> Maybe the debug log will shed some light. >>> >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/f76b21b4/attachment.html From anthony.minessale at gmail.com Mon Jan 26 08:47:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 26 Jan 2009 10:47:15 -0600 Subject: [Freeswitch-users] record session in fifo In-Reply-To: <497DE0C9.5090005@gmail.com> References: <4979F41F.70605@gmail.com> <191c3a030901231045m784f0cd7s6994b763d02b3e26@mail.gmail.com> <497D8782.2070603@gmail.com> <497DE0C9.5090005@gmail.com> Message-ID: <191c3a030901260847j660bb791w27657a08bdc232d0@mail.gmail.com> yes some code was missing for some reason, try again On Mon, Jan 26, 2009 at 10:11 AM, Tamas Cseke wrote: > Hello, > > I tested with the attached patch. > It is working fine in a normal case. > > I have only problems with the automatic calls, because in this case the > loopback channel is in the fifo, but the record_session is running on the > sofia channel. > Maybe it could be sort out with putting the bug pause/resume functions into > api function, what I should turn on and off on demand? > Anyway, I quess this is a bit extreme circumstance, and it isn't so > important to us now. > > Thanks, > Tamas > > Tamas Cseke ?rta: > > Hello, >> >> Thank you your help. >> >> I tested with r11489, but moh is still recorded in fifo. >> >> I quess you I should test the CF_PAUSE_BUGS in r11466. But I didn't find >> where you check this flag. >> Is it maybe possible you forget to commit something? >> >> Thanks, >> Tamas >> >> >> I didn't find where you >> Anthony Minessale ?rta: >> >> >>> please test latest trunk. >>> Patch added to pause media bugs while not in a bridge which should pause >>> recordings and cut out the moh. >>> >>> >>> On Fri, Jan 23, 2009 at 10:45 AM, Tamas Cseke >> >wrote: >>> >>> >>> >>>> Hello, >>>> >>>> we would like to distribute calls with fifo and record these sessions >>>> but we'd like to skip the recording while the caller is waiting. >>>> (we don't need to record the hold music, just the speech with the fifo >>>> consumer.) >>>> >>>> I tried >>>> >>>> >>>> >>>> >>>> but it doesn't work because the channel is answered immediately when the >>>> caller is pushed into the fifo. >>>> (I don't know if there exists any other channel flag that could be use >>>> here) >>>> >>>> I also tried fifo_record_template. >>>> but it records the session from the point of view of the consumer's >>>> session, and after the bridge the recording is stopped. >>>> we would like to record the whole session into a single file even after >>>> calltransfers >>>> >>>> moreover we'd like to use some kind of predcitive dialing >>>> which >>>> 1, originate a loopback channel via event socket >>>> 2, loopback-b channel is hunting the dialplan, wich decide routing, >>>> caller_id, the need for recordings and so forth, and bridge a sofia call >>>> 3. the record_session is running on the sofia channel with >>>> bridge_pre_execute magic vars >>>> 4 loopback-a channel is pushed into the fifo >>>> 5 a script get the fifo::info via event socket >>>> 6 originate a call to the consumer with the proper strategy with &fifo >>>> out application >>>> 7 sofia channel is bridged to the consumer >>>> 8 loopback channels die >>>> >>>> after transfers everything is recorded into one file. >>>> but the problem here is again the unwanted recording in the fifo while >>>> the caller is waiting >>>> >>>> Could you please advise me any solution, if there is? >>>> >>>> >>>> Thank you, >>>> Tamas >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/b3354082/attachment.html From Claudio.Cavalera at italtel.it Mon Jan 26 09:00:57 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Mon, 26 Jan 2009 18:00:57 +0100 Subject: [Freeswitch-users] is there an event for SIP MESSAGE? Message-ID: Hello freeswitchers, I'm experimenting with sip clients registered to fs and Instant Messaging. I've seen SIP Messages are properly routed by the sofia sip stack in fs. However it seems no event is ever generated on the event socket. Would it be possible to make fs reporting the SIP Messaging? :-) Best Regards, Claudio 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. ----------------------------------------------------------------------------------------------------- From testeador01 at gmail.com Mon Jan 26 09:41:00 2009 From: testeador01 at gmail.com (Milena) Date: Mon, 26 Jan 2009 12:41:00 -0500 Subject: [Freeswitch-users] My fs_cli stopped working, can you help me please? In-Reply-To: References: Message-ID: On a second thought... it would be nice if the console showed a message when there is a wrong password, something more descriptive than just doing nothing, also it is not clear to me why the console didn't require me to put the password before i deleted that folder's content even knowing that i changed my password a long time ago. well, thank you all :) 1/26 Milena > thanks ^^ > > 2009/1/26 Mathieu Rene > > Loos like the wrong password to me, look in >> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf.xml and use >> fs_cli -p [pass] >> >> >> Mathieu >> >> On Mon, Jan 26, 2009 at 8:33 AM, Milena wrote: >> >>> Good morning >>> Ok, here is what i get from the console, do you know what can i do to fix >>> it? thank you very much >>> >>> -bash-3.2# /usr/local/freeswitch/bin/fs_cli -d 7 >>> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration >>> file is /root/.fs_cli_conf. >>> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration >>> file is /etc/fs_cli.conf. >>> [DEBUG] libs/esl/fs_cli.c:573 main() profile default does not exist using >>> builtin profile >>> [DEBUG] libs/esl/fs_cli.c:597 main() Using profile internal [127.0.0.1] >>> [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER >>> [Content-Type] = [auth/request] >>> [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE >>> Content-Type: [auth/request] >>> >>> >>> [DEBUG] libs/esl/src/esl.c:853 esl_send() SEND >>> auth ClueCon >>> >>> >>> [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER >>> [Content-Type] = [command/reply] >>> [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Reply-Text] >>> = [-ERR invalid] >>> [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE >>> Content-Type: [command/reply] >>> Reply-Text: [-ERR invalid] >>> >>> >>> [ERROR] libs/esl/fs_cli.c:610 main() Error Connecting [Connection Error] >>> -bash-3.2# >>> >>> >>> >>> >>> Anthony Minessale wrote: >>> >>>> >>>> Try running >>>> >>>> fs_cli -d 7 >>>> >>>> Maybe the debug log will shed some light. >>>> >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/7e8fcb47/attachment.html From anthony.minessale at gmail.com Mon Jan 26 10:10:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 26 Jan 2009 12:10:45 -0600 Subject: [Freeswitch-users] My fs_cli stopped working, can you help me please? In-Reply-To: References: Message-ID: <191c3a030901261010o17d8d1dbi7a6963b12639c82f@mail.gmail.com> you're right, its fixed int tree to explain the issue now. On Mon, Jan 26, 2009 at 11:41 AM, Milena wrote: > On a second thought... > > it would be nice if the console showed a message when there is a wrong > password, something more descriptive than just doing nothing, > also it is not clear to me why the console didn't require me to put the > password before i deleted that folder's content even knowing that i changed > my password a long time ago. > > well, thank you all :) > > 1/26 Milena > > thanks ^^ >> >> 2009/1/26 Mathieu Rene >> >> Loos like the wrong password to me, look in >>> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf.xml and use >>> fs_cli -p [pass] >>> >>> >>> Mathieu >>> >>> On Mon, Jan 26, 2009 at 8:33 AM, Milena wrote: >>> >>>> Good morning >>>> Ok, here is what i get from the console, do you know what can i do to >>>> fix it? thank you very much >>>> >>>> -bash-3.2# /usr/local/freeswitch/bin/fs_cli -d 7 >>>> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() >>>> Configuration file is /root/.fs_cli_conf. >>>> [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() >>>> Configuration file is /etc/fs_cli.conf. >>>> [DEBUG] libs/esl/fs_cli.c:573 main() profile default does not exist >>>> using builtin profile >>>> [DEBUG] libs/esl/fs_cli.c:597 main() Using profile internal [127.0.0.1] >>>> [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER >>>> [Content-Type] = [auth/request] >>>> [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE >>>> Content-Type: [auth/request] >>>> >>>> >>>> [DEBUG] libs/esl/src/esl.c:853 esl_send() SEND >>>> auth ClueCon >>>> >>>> >>>> [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER >>>> [Content-Type] = [command/reply] >>>> [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Reply-Text] >>>> = [-ERR invalid] >>>> [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE >>>> Content-Type: [command/reply] >>>> Reply-Text: [-ERR invalid] >>>> >>>> >>>> [ERROR] libs/esl/fs_cli.c:610 main() Error Connecting [Connection Error] >>>> -bash-3.2# >>>> >>>> >>>> >>>> >>>> Anthony Minessale wrote: >>>> >>>>> >>>>> Try running >>>>> >>>>> fs_cli -d 7 >>>>> >>>>> Maybe the debug log will shed some light. >>>>> >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/7253a9f2/attachment.html From raul at etellicom.com Mon Jan 26 10:16:09 2009 From: raul at etellicom.com (Raul Fragoso) Date: Mon, 26 Jan 2009 16:16:09 -0200 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> Message-ID: <1232993769.8161.3.camel@stargate> You may want to define the conf/tones.conf file with the proper tone definitions for your country, then specify that tone group (tonegroup=XX where XX is the 2 letters code for your country). For example, this is the definition for my country (Brazil) and I also have tonegroup=br in my openzap configuration file: [br] generate-busy => v=-7;%(250,250,425) generate-dial => v=-7;%(1000,0,425) generate-ring => v=-7;%(1000,4000,425) generate-callwaiting-sas => v=0;%(300,10000,425) detect-busy => 425 detect-dial => 425 detect-ring => 425 detect-callwaiting-sas => 425 Regards, Raul Fragoso On Mon, 2009-01-26 at 21:32 +0800, Nandy Dagondon wrote: > that's great. yes, i'm in the philippines. there's a difference in > dialtone - it's 425 Hz. > -nandy > > > On Mon, Jan 26, 2009 at 6:01 PM, Michael Collins > wrote: > I have a TDM400 clone and I will see if I can reproduce these > symptoms. BTW, are you in the Philippines? Is there any > difference in > the dial tone there than in the US? > -MC > > > On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon > wrote: > > i monitored the line using another phone. there's indeed > dialtone in all > > attempts. > > i see TONE_DETECTED in the first call but i wonder there's a > WARNING message > > immediately following [WARNING] mod_openzap.c:1196 > on_fxo_signal() Unhandled > > type for channel 2:1. > > the dialtone freq should be okay since it's detected in the > first call.could > > the WARNING message gives us a hint of a possible problem > other than the > > dialtone freq? > > > > okay, i'll try the SVN version next. > > > > > > On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale > > wrote: > >> > >> Its not detecting a dial tone on the failure case. > >> Before dialing it waits until it picks up dialtone. > >> Try the svn trunk version to see if it works any better or > verify there is > >> a dialtone on the line. > >> > >> On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: > >> > >> hi everybody, > >> > >> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). > it's working using > >> IP phones, softphones and digium FXS port. but there's a > problem in dialing > >> out to PSTN using digium tdm400 fxo - it works fine on the > first attempt > >> (after starting FS) but it fails on the subsequent > attempts. i tested to > >> call using the FXS port and IP phone. same problem. > >> > >> before i place any call, i checked >oz dump 2 1 (show > current state = > >> DOWN, last state = DOWN) > >> > >> in the first call, there's this message: > >> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type > for channel > >> 2:1 > >> but > >> > >> then i hangup. checked >oz dump 21 (show current > state=DOWN, last > >> state=HANGUP) > >> > >> in the 2nd (and subsequent) attempts, the fxo just goes > off-hook but > >> doesn't send the dtmf tones. > >> >oz dump 2 1 (shows current state = DIALING, last state = > DOWN) > >> > >> has anyone encountered this problem before? i appreciate > for any help to > >> correct this problem. > >> > >> tks, > >> nandy > >> > >> > >> Environment: > >> ================== > >> kernel 2.6.18-92.1.22.el5 > >> FS 1.0.2 > >> zaptel 1.4.11 > >> oslec > >> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) > >> > >> zaptel.conf > >> ======== > >> loadzone = us > >> defaultzone=us > >> channels=1-2 > >> alaw=1-4 > >> fxsks=2 > >> fxoks=1 > >> > >> > >> openzap.conf.xml: > >> =============== > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> openzap.conf > >> ========== > >> [span zt] > >> name => OpenZAP FXS > >> number => 1 > >> fxs-channel => 1 > >> > >> [span zt] > >> name => OpenZAP FXO > >> number => 2 > >> fxo-channel => 2 > >> > >> tones.conf (the dialtone and ring tone is set to > Philipping tones) > >> ======== > >> [us] > >> generate-dial => v=-7;%(1000,0,425) > >> detect-dial => 425 > >> > >> generate-ring => v=-7;%(1000,4000,425,480) > >> detect-ring => 425,480 > >> > >> generate-busy => v=-7;%(500,500,480,620) > >> detect-busy => 480,620 > >> > >> generate-attn => v=0;%(200,300,1400,1800) > >> detect-attn => 1400,1800 > >> > >> generate-callwaiting-sas => v=0;%(300,10000,440) > >> detect-callwaiting-sas => 440 > >> > >> generate-callwaiting-cas => v=0;%(80,0,2750,2130) > >> detect-callwaiting-cas => 2750,2130 > >> > >> detect-fail1 => 913.8 > >> detect-fail2 => 1370.6 > >> detect-fail3 => 776.7 > >> > >> LOG OF FIRST CALL (OK) > >> ==================== > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 > >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 > Execute > >> bridge(openzap/2/1/3400534) > >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() > Set codec PCMU > >> 20ms > >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 > channel_outgoing_channel() > >> Connect outbound channel OpenZAP/2:1/3400534 > >> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 > >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 > >> [e5f12114-ea88-11dd-9f5c-290fb4a527a4] > >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 > channel_outgoing_channel() > >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT > >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 > analog_fxo_outgoing_call() > >> Changing state on 2:1 from DOWN to DIALING > >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 > zap_analog_channel_run() > >> ANALOG CHANNEL thread starting. > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running > State Change CS_INIT > >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > >> Executing state handler on 2:1 for DIALING > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT > >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 > channel_on_init() > >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING > >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT > going to sleep > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running > State Change > >> CS_ROUTING > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > ROUTING > >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 > channel_on_routing() > >> OpenZAP/2:1/3400534 CHANNEL ROUTING > >> 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 > >> originate_on_routing() (OpenZAP/2:1/3400534) State Change > CS_ROUTING -> > >> CS_CONSUME_MEDIA > >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > ROUTING going to sleep > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running > State Change > >> CS_CONSUME_MEDIA > >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > CONSUME_MEDIA > >> 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 > zap_analog_channel_run() > >> Detected tone DIAL on 2:1 > >> 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 > on_fxo_signal() got FXO sig > >> 2:1 [TONE_DETECTED] > >> 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 > on_fxo_signal() Unhandled > >> type for channel 2:1 > >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 > zchan_activate_dtmf_buffer() > >> Created DTMF Buffer! > >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 > GENERATE DTMF > >> [3400534] > >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 > zap_analog_channel_run() > >> Changing state on 2:1 from DIALING to UP > >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > >> Executing state handler on 2:1 for UP > >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 > on_fxo_signal() got FXO sig > >> 2:1 [UP] > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 > >> switch_channel_perform_mark_answered() Send signal > OpenZAP/1:1/93400534 > >> [BREAK] > >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 > on_fxo_signal() Channel > >> [OpenZAP/2:1/3400534] has been answered > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive > message [AUDIO_SYNC] > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 > >> switch_channel_perform_answer() OpenZAP/1:1/93400534 > receive message > >> [ANSWER] > >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 > >> channel_receive_message_fxs() Changing state on 1:1 from > IDLE to UP > >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 > >> channel_receive_message_fxs() Channel > [OpenZAP/1:1/93400534] has been > >> answered > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive > message > >> [AUDIO_SYNC] > >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 > >> switch_core_session_perform_receive_message() Send signal > >> OpenZAP/1:1/93400534 [BREAK] > >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 > >> switch_ivr_originate() Originate Resulted in Success: > [OpenZAP/2:1/3400534] > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive > message [AUDIO_SYNC] > >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 > >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive > message > >> [AUDIO_SYNC] > >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 > >> switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 > receive message > >> [BRIDGE] > >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 > >> switch_core_session_perform_receive_message() Send signal > >> OpenZAP/2:1/3400534 [BREAK] > >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 > >> switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 > receive message > >> [BRIDGE] > >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 > >> switch_core_session_perform_receive_message() Send signal > >> OpenZAP/1:1/93400534 [BREAK] > >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 > >> switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) > State Change > >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > >> Executing state handler on 1:1 for UP > >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 > on_fxs_signal() got FXS sig > >> [UP] > >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > CONSUME_MEDIA going to > >> sleep > >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running > State Change > >> CS_EXCHANGE_MEDIA > >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > EXCHANGE_MEDIA > >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 > channel_on_exchange_media() > >> CHANNEL EXCHANGE_MEDIA > >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 > process_event() EVENT > >> [ONHOOK][1:1] STATE [UP] > >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 > process_event() Changing > >> state on 1:1 from UP to DOWN > >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > >> Executing state handler on 1:1 for DOWN > >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 > on_fxs_signal() got FXS sig > >> [STOP] > >> 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 > on_fxs_signal() Hangup > >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] > >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 > >> switch_channel_perform_hangup() Send signal > OpenZAP/1:1/93400534 [KILL] > >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/1:1/93400534 > >> [BREAK] > >> 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 > zap_channel_done() channel done > >> 1:1 > >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 > zap_analog_channel_run() > >> ANALOG CHANNEL 1:1 thread ended. > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 > audio_bridge_thread() > >> OpenZAP/1:1/93400534 ending bridge by request from read > function > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 > audio_bridge_thread() > >> BRIDGE THREAD DONE [OpenZAP/1:1/93400534] > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 > audio_bridge_thread() > >> Send signal OpenZAP/2:1/3400534 [BREAK] > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 > audio_bridge_thread() > >> OpenZAP/1:1/93400534 ending bridge by request from write > function > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 > audio_bridge_thread() > >> OpenZAP/2:1/3400534 receive message [UNBRIDGE] > >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 > >> switch_core_session_perform_receive_message() Send signal > >> OpenZAP/2:1/3400534 [BREAK] > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 > audio_bridge_thread() > >> BRIDGE THREAD DONE [OpenZAP/2:1/3400534] > >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 > audio_bridge_thread() > >> Send signal OpenZAP/1:1/93400534 [BREAK] > >> 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 > >> audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 > >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 > >> switch_channel_perform_hangup() Send signal > OpenZAP/2:1/3400534 [KILL] > >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > EXCHANGE_MEDIA going > >> to sleep > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running > State Change > >> CS_HANGUP > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > HANGUP > >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 > channel_on_hangup() Changing > >> state on 2:1 from UP to HANGUP > >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 > channel_on_hangup() > >> OpenZAP/2:1/3400534 CHANNEL HANGUP > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 > >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 > Standard HANGUP, cause: > >> NORMAL_CLEARING > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > HANGUP going to sleep > >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 > >> switch_core_session_thread() Session 2 > (OpenZAP/2:1/3400534) Locked, Waiting > >> on external entities > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State > EXECUTE going to > >> sleep > >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 > >> switch_core_session_thread() Session 2 > (OpenZAP/2:1/3400534) Ended > >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 > >> switch_core_session_thread() Close Channel > OpenZAP/2:1/3400534 [CS_HANGUP] > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/1:1/93400534) Running > State Change > >> CS_HANGUP > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State > HANGUP > >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 > channel_on_hangup() > >> OpenZAP/1:1/93400534 CHANNEL HANGUP > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 > >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 > Standard HANGUP, > >> cause: NORMAL_CLEARING > >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State > HANGUP going to sleep > >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 > >> switch_core_session_thread() Session 1 > (OpenZAP/1:1/93400534) Locked, > >> Waiting on external entities > >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 > >> switch_core_session_thread() Session 1 > (OpenZAP/1:1/93400534) Ended > >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 > >> switch_core_session_thread() Close Channel > OpenZAP/1:1/93400534 [CS_HANGUP] > >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > >> Executing state handler on 2:1 for HANGUP > >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 > zap_analog_channel_run() > >> Changing state on 2:1 from HANGUP to DOWN > >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > >> Executing state handler on 2:1 for DOWN > >> 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 > on_fxo_signal() got FXO sig > >> 2:1 [STOP] > >> 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 > zap_channel_done() channel done > >> 2:1 > >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 > zap_analog_channel_run() > >> ANALOG CHANNEL 2:1 thread ended. > >> > >> LOG OF FAILED CALLS > >> ================== > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 > >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 > Execute > >> bridge(openzap/2/1/3400534) > >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() > Set codec PCMU > >> 20ms > >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 > channel_outgoing_channel() > >> Connect outbound channel OpenZAP/2:1/3400534 > >> 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 > >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 > >> [079f5420-ea89-11dd-9f5c-290fb4a527a4] > >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 > channel_outgoing_channel() > >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT > >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 > analog_fxo_outgoing_call() > >> Changing state on 2:1 from DOWN to DIALING > >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 > zap_analog_channel_run() > >> ANALOG CHANNEL thread starting. > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running > State Change CS_INIT > >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > >> Executing state handler on 2:1 for DIALING > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT > >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 > channel_on_init() > >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING > >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT > going to sleep > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running > State Change > >> CS_ROUTING > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > ROUTING > >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 > channel_on_routing() > >> OpenZAP/2:1/3400534 CHANNEL ROUTING > >> 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 > >> originate_on_routing() (OpenZAP/2:1/3400534) State Change > CS_ROUTING -> > >> CS_CONSUME_MEDIA > >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > ROUTING going to sleep > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running > State Change > >> CS_CONSUME_MEDIA > >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > CONSUME_MEDIA > >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 > process_event() EVENT > >> [ONHOOK][1:1] STATE [IDLE] > >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 > process_event() Changing > >> state on 1:1 from IDLE to DOWN > >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > >> Executing state handler on 1:1 for DOWN > >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 > on_fxs_signal() got FXS sig > >> [STOP] > >> 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 > on_fxs_signal() Hangup > >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] > >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 > >> switch_channel_perform_hangup() Send signal > OpenZAP/1:1/93400534 [KILL] > >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/1:1/93400534 > >> [BREAK] > >> 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 > zap_channel_done() channel done > >> 1:1 > >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 > zap_analog_channel_run() > >> ANALOG CHANNEL 1:1 thread ended. > >> 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 > >> switch_ivr_originate() Hangup OpenZAP/2:1/3400534 > [CS_CONSUME_MEDIA] > >> [ORIGINATOR_CANCEL] > >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 > >> switch_channel_perform_hangup() Send signal > OpenZAP/2:1/3400534 [KILL] > >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 > >> switch_core_session_signal_state_change() Send signal > OpenZAP/2:1/3400534 > >> [BREAK] > >> 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 > >> switch_ivr_originate() Originate Cancelled by originator > termination Cause: > >> 487 [ORIGINATOR_CANCEL] > >> 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 > audio_bridge_function() > >> Originate Failed. Cause: ORIGINATOR_CANCEL > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State > EXECUTE going to > >> sleep > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/1:1/93400534) Running > State Change > >> CS_HANGUP > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State > HANGUP > >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 > channel_on_hangup() > >> OpenZAP/1:1/93400534 CHANNEL HANGUP > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 > >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 > Standard HANGUP, > >> cause: NORMAL_CLEARING > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/1:1/93400534) State > HANGUP going to sleep > >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 > >> switch_core_session_thread() Session 3 > (OpenZAP/1:1/93400534) Locked, > >> Waiting on external entities > >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 > >> switch_core_session_thread() Session 3 > (OpenZAP/1:1/93400534) Ended > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > CONSUME_MEDIA going to > >> sleep > >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 > >> switch_core_session_thread() Close Channel > OpenZAP/1:1/93400534 [CS_HANGUP] > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 > >> switch_core_session_run() (OpenZAP/2:1/3400534) Running > State Change > >> CS_HANGUP > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > HANGUP > >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 > channel_on_hangup() Changing > >> state on 2:1 from DIALING to HANGUP > >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 > channel_on_hangup() > >> OpenZAP/2:1/3400534 CHANNEL HANGUP > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 > >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 > Standard HANGUP, cause: > >> ORIGINATOR_CANCEL > >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (OpenZAP/2:1/3400534) State > HANGUP going to sleep > >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 > >> switch_core_session_thread() Session 4 > (OpenZAP/2:1/3400534) Locked, Waiting > >> on external entities > >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 > >> switch_core_session_thread() Session 4 > (OpenZAP/2:1/3400534) Ended > >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 > >> switch_core_session_thread() Close Channel > OpenZAP/2:1/3400534 [CS_HANGUP] > >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > >> Executing state handler on 2:1 for HANGUP > >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 > zap_analog_channel_run() > >> Changing state on 2:1 from HANGUP to DOWN > >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > >> Executing state handler on 2:1 for DOWN > >> 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 > on_fxo_signal() got FXO sig > >> 2:1 [STOP] > >> 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 > zap_channel_done() channel done > >> 2:1 > >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 > zap_analog_channel_run() > >> ANALOG CHANNEL 2:1 thread ended. > >> > >> > >> > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ronmccar at gmail.com Mon Jan 26 11:58:59 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Mon, 26 Jan 2009 12:58:59 -0700 Subject: [Freeswitch-users] Inbound calls Message-ID: <3885f4fe0901261158s5768734cjf80d4349735f50be@mail.gmail.com> Hi, If I allow the IPs in a ACL the context always ges changed, is their a way you can have inbound calls be auth'ed via a gateway and then sent to a context. Do I use the directory in that case and not treat it as a external gateway, it would be considered internal then? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/079440ce/attachment.html From telles-listas at devel-it.com.br Mon Jan 26 12:07:17 2009 From: telles-listas at devel-it.com.br (Rodrigo P. Telles) Date: Mon, 26 Jan 2009 18:07:17 -0200 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org> <4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br> <87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> Message-ID: <497E17F5.2020400@devel-it.com.br> Hi Michael, I'm not sure about this, when we need to use G729 on asterisk for example, we pay the digium (U$ 10/channel) licenses. U$ 1.00 dolar = R$ 2.31 (Brazilian Real - local currency). Att., Em 23-01-2009 18:29, Michael Collins escreveu: > On Fri, Jan 23, 2009 at 12:16 PM, Rodrigo P. Telles > wrote: >> Hi Dave, >> >> Down here in Brazil, the bandwidth costs is very high (around U$ 400.00/Mb) so it should be valid only for a "non" third >> world country. >> G729 and G723.1 is almost a law here, if you don't play at least with G729 your ITSP is out of mark share! >> >> My 2 cents from a third world country. > > What is the patent and licensing situation in Brazil? Those are also > factors. $10/port might be cheap in the US but in Brazil it could be > much more? (I'm asking...) > -MC From brian at freeswitch.org Mon Jan 26 12:07:22 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 26 Jan 2009 14:07:22 -0600 Subject: [Freeswitch-users] Inbound calls In-Reply-To: <3885f4fe0901261158s5768734cjf80d4349735f50be@mail.gmail.com> References: <3885f4fe0901261158s5768734cjf80d4349735f50be@mail.gmail.com> Message-ID: Yes you would use the directory... you add a cidr= attr to the user in the directory: Say you add a user into "domain.com": ... ... Then in acl.conf.xml you have something like this: the above entry would create an acl from all users in "domain.com" that then you could use to apply to the sofia profile. You would set the variable user_context on the user and thats what context that user would go into. On another note please don't hijack threads. Click new message and input the address freeswitch-users at lists.freeswitch.org, if you click reply to an existing message and change the subject and body you are hijacking threads which is what you did on this post. Thanks, Brian On Jan 26, 2009, at 1:58 PM, Ron McCarthy wrote: > Hi, > If I allow the IPs in a ACL the context always ges changed, is their > a way you can have inbound calls be auth'ed via a gateway and then > sent to a context. Do I use the directory in that case and not treat > it as a external gateway, it would be considered internal then? > > Thanks! From brian at freeswitch.org Mon Jan 26 12:08:06 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 26 Jan 2009 14:08:06 -0600 Subject: [Freeswitch-users] SIP chat In-Reply-To: <488F0AF1.30900@3c.co.uk> References: <18694165.post@talk.nabble.com> <40E93BCD-2BFF-49E2-A668-DFD15B98A644@freeswitch.org> <18707438.post@talk.nabble.com> <488F0AF1.30900@3c.co.uk> Message-ID: <830B1887-BB31-499D-BE04-7989AD623EB8@freeswitch.org> Oh it was you that Hijacked the thread... sorry Ron... It wasn't you this time! ;) David naughty! :) /b On Jul 29, 2008, at 7:20 AM, David Knell wrote: > Hi - > > chat - sending a messages - only works to a SIP endpoint if the user's > profile is the same > as their host - the relevant code is at lines 75-90 of > sofia_presence.c > > Is this by design - I can't see why it would be, but maybe I'm missing > something. If not, > might I suggest a syntax for the endpoint to send a message to of > [profile/]user at host - > if this is OK, I'll submit a patch. > > Cheers -- > > Dave > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From alhakeem at gmail.com Mon Jan 26 13:30:09 2009 From: alhakeem at gmail.com (Abdul Hakeem) Date: Mon, 26 Jan 2009 21:30:09 -0000 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <497E17F5.2020400@devel-it.com.br> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org><4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br><87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> <497E17F5.2020400@devel-it.com.br> Message-ID: Is Brazil a 3rd world country ? The last I hear Brazil was building aeroplanes, has it's own space and nuclear program and a GNP UK would be envious of. Cheers, AH -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rodrigo P. Telles Sent: 26 January 2009 20:07 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_g729 Hi Michael, I'm not sure about this, when we need to use G729 on asterisk for example, we pay the digium (U$ 10/channel) licenses. U$ 1.00 dolar = R$ 2.31 (Brazilian Real - local currency). Att., Em 23-01-2009 18:29, Michael Collins escreveu: > On Fri, Jan 23, 2009 at 12:16 PM, Rodrigo P. Telles > wrote: >> Hi Dave, >> >> Down here in Brazil, the bandwidth costs is very high (around U$ >> 400.00/Mb) so it should be valid only for a "non" third world country. >> G729 and G723.1 is almost a law here, if you don't play at least with G729 your ITSP is out of mark share! >> >> My 2 cents from a third world country. > > What is the patent and licensing situation in Brazil? Those are also > factors. $10/port might be cheap in the US but in Brazil it could be > much more? (I'm asking...) -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From Laurent.Fabre at kirranet.com Mon Jan 26 14:19:24 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Mon, 26 Jan 2009 23:19:24 +0100 Subject: [Freeswitch-users] Call Groups Message-ID: Hi, I'm having trouble with call groups. They are declared in the directory.xml as mentioned by the documentation : And included in the dialplan : The endpoints are registered, the dialplan match but I get a cause: NO_ROUTE_DESTINATION I obviously screwed up somewhere. Anybody got an hint where? :) Regards, LF -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/8b96833f/attachment.html From msc at freeswitch.org Mon Jan 26 14:56:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Jan 2009 14:56:03 -0800 Subject: [Freeswitch-users] Call Groups In-Reply-To: References: Message-ID: <87f2f3b90901261456o59e039c1gc6795453dcb3ace8@mail.gmail.com> Can you try the groups that come with the default dialplan and see if they work? Just curious. Also, what revision are you running? -MC On Mon, Jan 26, 2009 at 2:19 PM, Laurent Fabre wrote: > Hi, > > > > I'm having trouble with call groups. They are declared in the directory.xml > as mentioned by the documentation : > > > > > > > > > > > > > > > > > > And included in the dialplan : > > > > > > > > > > > > > > > > The endpoints are registered, the dialplan match but I get a cause: > NO_ROUTE_DESTINATION > > > > I obviously screwed up somewhere. Anybody got an hint where? J > > > > Regards, > > > > LF > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jim at archer.net Mon Jan 26 15:16:23 2009 From: jim at archer.net (Jim Archer) Date: Mon, 26 Jan 2009 18:16:23 -0500 Subject: [Freeswitch-users] Trouble with default config - Can't call default extensions Message-ID: <33B63908AE8E654C15C69AB5@cassius> Hi All... I installed FS trunk a few days ago on a Debian Etch AMD64 machine. I configured two Polycom 501 phones to talk to it. From each phone, I can dial extension 9999 to get the MOH and 5000 to get the sample IVR. But, I can not call one phone from the other. One is extension 1000 and the other is 1001. I checked and the sample XML files are there in /usr/local/freeswitch/conf/directory/default It seems that as soon as I dial the second digit, the phone decides it is done waiting for me to dial any more, so it tried to connect me to extension 10. Here is what FS spits on to the console: 2009-01-26 18:10:35 [NOTICE] switch_channel.c:566 switch_channel_set_name() New Channel sofia/internal/1001 at 72.46.5.252 [898cf87c-ebfe-11dd-a8c8-b7a3de0bf1ff] 2009-01-26 18:10:35 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FS->10 in context default 2009-01-26 18:10:35 [NOTICE] switch_ivr.c:1253 switch_ivr_session_transfer() Transfer sofia/internal/1001 at 72.46.5.252 to enum[10 at default] 2009-01-26 18:10:36 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting 2009-01-26 18:10:36 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup sofia/internal/1001 at 72.46.5.252 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-01-26 18:10:36 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 25 (sofia/internal/1001 at 72.46.5.252) Ended 2009-01-26 18:10:36 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/1001 at 72.46.5.252 [CS_HANGUP] Any suggestions would be appreciated, thanks! Jim From chavpaskov at shaw.ca Mon Jan 26 15:20:56 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Mon, 26 Jan 2009 15:20:56 -0800 Subject: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock Message-ID: <497E4558.1030202@shaw.ca> Hi everybody, i got a strange message today: [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock i'm running 1.0.trunk (10803) as a VM on ESXi i have a similar setup of 1.0.1 as VM on ESXi but i've never seen this message. Does anybody have an idea what is wrong here? All tips / hints appreciated. Best Regards Chav From msc at freeswitch.org Mon Jan 26 15:23:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Jan 2009 15:23:19 -0800 Subject: [Freeswitch-users] Trouble with default config - Can't call default extensions In-Reply-To: <33B63908AE8E654C15C69AB5@cassius> References: <33B63908AE8E654C15C69AB5@cassius> Message-ID: <87f2f3b90901261523p44d8bc17v6425c08a66a1cf38@mail.gmail.com> Sounds like the Polycoms are expecting only two digits when you start with a 1. Have these phones been used before? Perhaps you could reset one of them to the factory default and start from scratch and see what happens. -MC On Mon, Jan 26, 2009 at 3:16 PM, Jim Archer wrote: > Hi All... > > I installed FS trunk a few days ago on a Debian Etch AMD64 machine. I > configured two Polycom 501 phones to talk to it. From each phone, I can > dial extension 9999 to get the MOH and 5000 to get the sample IVR. But, I > can not call one phone from the other. One is extension 1000 and the other > is 1001. > > I checked and the sample XML files are there in > /usr/local/freeswitch/conf/directory/default > > It seems that as soon as I dial the second digit, the phone decides it is > done waiting for me to dial any more, so it tried to connect me to > extension 10. Here is what FS spits on to the console: > > 2009-01-26 18:10:35 [NOTICE] switch_channel.c:566 switch_channel_set_name() > New Channel sofia/internal/1001 at 72.46.5.252 > [898cf87c-ebfe-11dd-a8c8-b7a3de0bf1ff] > 2009-01-26 18:10:35 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing FS->10 in context default > 2009-01-26 18:10:35 [NOTICE] switch_ivr.c:1253 > switch_ivr_session_transfer() Transfer sofia/internal/1001 at 72.46.5.252 to > enum[10 at default] > 2009-01-26 18:10:36 [INFO] switch_core_state_machine.c:122 > switch_core_standard_on_routing() No Route, Aborting > 2009-01-26 18:10:36 [NOTICE] switch_core_state_machine.c:123 > switch_core_standard_on_routing() Hangup sofia/internal/1001 at 72.46.5.252 > [CS_ROUTING] [NO_ROUTE_DESTINATION] > 2009-01-26 18:10:36 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 25 (sofia/internal/1001 at 72.46.5.252) > Ended > 2009-01-26 18:10:36 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel sofia/internal/1001 at 72.46.5.252 > [CS_HANGUP] > > Any suggestions would be appreciated, thanks! > > Jim > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Mon Jan 26 15:25:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 26 Jan 2009 17:25:08 -0600 Subject: [Freeswitch-users] Trouble with default config - Can't call default extensions In-Reply-To: <33B63908AE8E654C15C69AB5@cassius> References: <33B63908AE8E654C15C69AB5@cassius> Message-ID: You'll need to fix your dialplan in the polycom itself its sending the invite after you dial 10. /b On Jan 26, 2009, at 5:16 PM, Jim Archer wrote: > enum[10 at default] From ajlong at worldlink.net Mon Jan 26 15:28:33 2009 From: ajlong at worldlink.net (Adam Long) Date: Mon, 26 Jan 2009 18:28:33 -0500 Subject: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock In-Reply-To: <497E4558.1030202@shaw.ca> References: <497E4558.1030202@shaw.ca> Message-ID: <016801c9800d$ce4701e0$6ad505a0$@net> I have seen that randomly during core dumps back when mod_managed was causing core dump on load. I am running VMWare ESX 3.5 not ESXi My curiousity was peaked as well, however I dismissed that as something to do with slow/inacurate timing in vmware. I do not have my ESX interupts tuned at all, just set to default install values. I too would be curious what exactly this means. Regards, -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chav Paskov Sent: Monday, January 26, 2009 6:21 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock Hi everybody, i got a strange message today: [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock i'm running 1.0.trunk (10803) as a VM on ESXi i have a similar setup of 1.0.1 as VM on ESXi but i've never seen this message. Does anybody have an idea what is wrong here? All tips / hints appreciated. Best Regards Chav _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Mon Jan 26 15:32:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 26 Jan 2009 17:32:42 -0600 Subject: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock In-Reply-To: <016801c9800d$ce4701e0$6ad505a0$@net> References: <497E4558.1030202@shaw.ca> <016801c9800d$ce4701e0$6ad505a0$@net> Message-ID: <2A3D5D5A-135C-4A39-8FC7-DDFF9436EA4B@freeswitch.org> It means your clock is slipping .. which is bad... The error is for when you do a migration say between two xen boxes or two openvz boxes so that things will carry on once the migration is done. (or your clock is slipping) /b On Jan 26, 2009, at 5:28 PM, Adam Long wrote: > I have seen that randomly during core dumps back when mod_managed was > causing core dump on load. > I am running VMWare ESX 3.5 not ESXi > > My curiousity was peaked as well, however I dismissed that as > something to > do with slow/inacurate timing in vmware. > I do not have my ESX interupts tuned at all, just set to default > install > values. > > I too would be curious what exactly this means. > > Regards, > -Adam From chavpaskov at shaw.ca Mon Jan 26 15:42:06 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Mon, 26 Jan 2009 15:42:06 -0800 Subject: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock In-Reply-To: <2A3D5D5A-135C-4A39-8FC7-DDFF9436EA4B@freeswitch.org> References: <497E4558.1030202@shaw.ca> <016801c9800d$ce4701e0$6ad505a0$@net> <2A3D5D5A-135C-4A39-8FC7-DDFF9436EA4B@freeswitch.org> Message-ID: <497E4A4E.9060803@shaw.ca> Brian West wrote: > It means your clock is slipping .. which is bad... The error is for > when you do a migration say between two xen boxes or two openvz boxes > so that things will carry on once the migration is done. (or your > clock is slipping) > > > /b > > On Jan 26, 2009, at 5:28 PM, Adam Long wrote: > > >> I have seen that randomly during core dumps back when mod_managed was >> causing core dump on load. >> I am running VMWare ESX 3.5 not ESXi >> >> My curiousity was peaked as well, however I dismissed that as >> something to >> do with slow/inacurate timing in vmware. >> I do not have my ESX interupts tuned at all, just set to default >> install >> values. >> >> I too would be curious what exactly this means. >> >> Regards, >> -Adam >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Thanks for the prompt response. will it help if i set a cron job to keep up the clock correct? i'm just curious because i definitely have no migration process set. Regards Chav From brian at freeswitch.org Mon Jan 26 15:47:01 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 26 Jan 2009 17:47:01 -0600 Subject: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock In-Reply-To: <497E4A4E.9060803@shaw.ca> References: <497E4558.1030202@shaw.ca> <016801c9800d$ce4701e0$6ad505a0$@net> <2A3D5D5A-135C-4A39-8FC7-DDFF9436EA4B@freeswitch.org> <497E4A4E.9060803@shaw.ca> Message-ID: <9C3E0152-4D2D-48FD-B718-AE8FF86A2AA9@freeswitch.org> running ntpd will help but if your clock is slipping that much you might want to re-task the machine as a boat anchor. /b On Jan 26, 2009, at 5:42 PM, Chav Paskov wrote: > Thanks for the prompt response. > will it help if i set a cron job to keep up the clock correct? > i'm just curious because i definitely have no migration process set. > Regards > Chav From jim at archer.net Mon Jan 26 15:55:16 2009 From: jim at archer.net (Jim Archer) Date: Mon, 26 Jan 2009 18:55:16 -0500 Subject: [Freeswitch-users] Trouble with default config - Can't call default extensions In-Reply-To: <87f2f3b90901261523p44d8bc17v6425c08a66a1cf38@mail.gmail.com> References: <33B63908AE8E654C15C69AB5@cassius> <87f2f3b90901261523p44d8bc17v6425c08a66a1cf38@mail.gmail.com> Message-ID: <40AB03C41EBEFDECBCA54053@cassius> Oh, there is a "digit map" in there. All fixed, thanks very much! --On Monday, January 26, 2009 3:23 PM -0800 Michael Collins wrote: > Sounds like the Polycoms are expecting only two digits when you start > with a 1. Have these phones been used before? Perhaps you could reset > one of them to the factory default and start from scratch and see what > happens. > -MC > > On Mon, Jan 26, 2009 at 3:16 PM, Jim Archer wrote: >> Hi All... >> >> I installed FS trunk a few days ago on a Debian Etch AMD64 machine. I >> configured two Polycom 501 phones to talk to it. From each phone, I can >> dial extension 9999 to get the MOH and 5000 to get the sample IVR. But, I >> can not call one phone from the other. One is extension 1000 and the >> other is 1001. >> >> I checked and the sample XML files are there in >> /usr/local/freeswitch/conf/directory/default >> >> It seems that as soon as I dial the second digit, the phone decides it is >> done waiting for me to dial any more, so it tried to connect me to >> extension 10. Here is what FS spits on to the console: >> >> 2009-01-26 18:10:35 [NOTICE] switch_channel.c:566 >> switch_channel_set_name() New Channel sofia/internal/1001 at 72.46.5.252 >> [898cf87c-ebfe-11dd-a8c8-b7a3de0bf1ff] >> 2009-01-26 18:10:35 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing FS->10 in context default >> 2009-01-26 18:10:35 [NOTICE] switch_ivr.c:1253 >> switch_ivr_session_transfer() Transfer sofia/internal/1001 at 72.46.5.252 to >> enum[10 at default] >> 2009-01-26 18:10:36 [INFO] switch_core_state_machine.c:122 >> switch_core_standard_on_routing() No Route, Aborting >> 2009-01-26 18:10:36 [NOTICE] switch_core_state_machine.c:123 >> switch_core_standard_on_routing() Hangup sofia/internal/1001 at 72.46.5.252 >> [CS_ROUTING] [NO_ROUTE_DESTINATION] >> 2009-01-26 18:10:36 [NOTICE] switch_core_session.c:960 >> switch_core_session_thread() Session 25 (sofia/internal/1001 at 72.46.5.252) >> Ended >> 2009-01-26 18:10:36 [NOTICE] switch_core_session.c:962 >> switch_core_session_thread() Close Channel >> sofia/internal/1001 at 72.46.5.252 [CS_HANGUP] >> >> Any suggestions would be appreciated, thanks! >> >> Jim >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ajlong at worldlink.net Mon Jan 26 16:00:48 2009 From: ajlong at worldlink.net (Adam Long) Date: Mon, 26 Jan 2009 19:00:48 -0500 Subject: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock In-Reply-To: <9C3E0152-4D2D-48FD-B718-AE8FF86A2AA9@freeswitch.org> References: <497E4558.1030202@shaw.ca> <016801c9800d$ce4701e0$6ad505a0$@net> <2A3D5D5A-135C-4A39-8FC7-DDFF9436EA4B@freeswitch.org> <497E4A4E.9060803@shaw.ca> <9C3E0152-4D2D-48FD-B718-AE8FF86A2AA9@freeswitch.org> Message-ID: <016b01c98012$4f4d9160$ede8b420$@net> Haha... it is probably either a case of overloading the ESX host server (too many other guests on same machine) Or your ESX isn't pumping out enough interrupts. Have a look here.. this might help. http://kb.vmware.com/selfservice/microsites/search.do?language=en_US&cmd=dis playKC&externalId=2219 Not 100% sure if this is still acurate for ESX 3.5 or ESXi 3.5 -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, January 26, 2009 6:47 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock running ntpd will help but if your clock is slipping that much you might want to re-task the machine as a boat anchor. /b On Jan 26, 2009, at 5:42 PM, Chav Paskov wrote: > Thanks for the prompt response. > will it help if i set a cron job to keep up the clock correct? > i'm just curious because i definitely have no migration process set. > Regards > Chav _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From edpimentl at gmail.com Mon Jan 26 16:08:00 2009 From: edpimentl at gmail.com (EdPimentl) Date: Mon, 26 Jan 2009 19:08:00 -0500 Subject: [Freeswitch-users] Call out to gtalk, asterisk and skype Message-ID: <9dc4a1670901261608g2ded99cchd99fcce55e7bf1ac@mail.gmail.com> Hello Everyone, Has anyone done a scalable deployment of calling to Skype/Gtalk/Asterisk Say many 3-25/50 users with various endpoint (skype/gtalk/asterisk)platforms. http://blog.tmcnet.com/blog/tom-keating/asterisk/skype-for-asterisk-launches.asp http://www.mhspot.com/siptheeskype_skype_trunk_howto.html Thanks in advance, -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/b360aa56/attachment.html From ajlong at worldlink.net Mon Jan 26 16:11:12 2009 From: ajlong at worldlink.net (Adam Long) Date: Mon, 26 Jan 2009 19:11:12 -0500 Subject: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock In-Reply-To: <9C3E0152-4D2D-48FD-B718-AE8FF86A2AA9@freeswitch.org> References: <497E4558.1030202@shaw.ca> <016801c9800d$ce4701e0$6ad505a0$@net> <2A3D5D5A-135C-4A39-8FC7-DDFF9436EA4B@freeswitch.org> <497E4A4E.9060803@shaw.ca> <9C3E0152-4D2D-48FD-B718-AE8FF86A2AA9@freeswitch.org> Message-ID: <016e01c98013$c37fb3f0$4a7f1bd0$@net> Sorry that last link was mangled... If you are running inside ESX might want to have a look here... http://kb.vmware.com/selfservice/microsites/search.do?language=en_US&cmd=dis playKC&externalId=2219 -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, January 26, 2009 6:47 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock running ntpd will help but if your clock is slipping that much you might want to re-task the machine as a boat anchor. /b On Jan 26, 2009, at 5:42 PM, Chav Paskov wrote: > Thanks for the prompt response. > will it help if i set a cron job to keep up the clock correct? > i'm just curious because i definitely have no migration process set. > Regards > Chav _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jim at archer.net Mon Jan 26 16:16:37 2009 From: jim at archer.net (Jim Archer) Date: Mon, 26 Jan 2009 19:16:37 -0500 Subject: [Freeswitch-users] USER_NOT_REGISTERED Message-ID: <588F9CC0AC6193DF57D4B106@cassius> Sorry for all these simple questions, but I'm having a registration issue. I have two Polycom 501 phones and have configured both to FreeSwitch. One is extension 1000 and the other is extension 1001. I can call 1000 from 1001, but not the reverse. If I try to call 1001, I go straight to voice mail and the console tells me that 1001 is not registered: 2009-01-26 19:11:24 [ERR] switch_ivr_originate.c:1391 switch_ivr_originate() Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] Why would it be able to make calls if it is not registered? From brian at freeswitch.org Mon Jan 26 16:16:51 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 26 Jan 2009 18:16:51 -0600 Subject: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock In-Reply-To: <016e01c98013$c37fb3f0$4a7f1bd0$@net> References: <497E4558.1030202@shaw.ca> <016801c9800d$ce4701e0$6ad505a0$@net> <2A3D5D5A-135C-4A39-8FC7-DDFF9436EA4B@freeswitch.org> <497E4A4E.9060803@shaw.ca> <9C3E0152-4D2D-48FD-B718-AE8FF86A2AA9@freeswitch.org> <016e01c98013$c37fb3f0$4a7f1bd0$@net> Message-ID: haha so was that one! :P Anchors Away!!! /b On Jan 26, 2009, at 6:11 PM, Adam Long wrote: > Sorry that last link was mangled... > > If you are running inside ESX might want to have a look here... > http://kb.vmware.com/selfservice/microsites/search.do?language=en_US&cmd=dis > playKC&externalId=2219 > > > -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/cf2b1eff/attachment.html From brian at freeswitch.org Mon Jan 26 16:28:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 26 Jan 2009 18:28:40 -0600 Subject: [Freeswitch-users] USER_NOT_REGISTERED In-Reply-To: <588F9CC0AC6193DF57D4B106@cassius> References: <588F9CC0AC6193DF57D4B106@cassius> Message-ID: <0DDA3C3E-D49C-4E3D-8A59-72EE2C7E3E84@freeswitch.org> Double check your registration because the user isn't registered. Check "sofia status profile internal" /b On Jan 26, 2009, at 6:16 PM, Jim Archer wrote: > Sorry for all these simple questions, but I'm having a registration > issue. > > I have two Polycom 501 phones and have configured both to > FreeSwitch. One > is extension 1000 and the other is extension 1001. I can call 1000 > from > 1001, but not the reverse. If I try to call 1001, I go straight to > voice > mail and the console tells me that 1001 is not registered: > > 2009-01-26 19:11:24 [ERR] switch_ivr_originate.c:1391 > switch_ivr_originate() Cannot create outgoing channel of type [error] > cause: [USER_NOT_REGISTERED] > > Why would it be able to make calls if it is not registered? From gcd at i.ph Mon Jan 26 16:39:21 2009 From: gcd at i.ph (Nandy Dagondon) Date: Tue, 27 Jan 2009 08:39:21 +0800 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <191c3a030901260833l2e07bfffkca9400d33a5483fe@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> <191c3a030901260833l2e07bfffkca9400d33a5483fe@mail.gmail.com> Message-ID: <7d0bfd8c0901261639x7c53f256yabf7347b5d22e7e4@mail.gmail.com> i tested the SVN trunk version. still the same behaviour. -nandy On Tue, Jan 27, 2009 at 12:33 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The unhanded type message just means that mod_openzap does not do anything > with the TONE_DETECTED event that was passed > up from the ozmod_analog. > > On Mon, Jan 26, 2009 at 7:32 AM, Nandy Dagondon wrote: > >> that's great. yes, i'm in the philippines. there's a difference in >> dialtone - it's 425 Hz. >> -nandy >> >> >> >> On Mon, Jan 26, 2009 at 6:01 PM, Michael Collins wrote: >> >>> I have a TDM400 clone and I will see if I can reproduce these >>> symptoms. BTW, are you in the Philippines? Is there any difference in >>> the dial tone there than in the US? >>> -MC >>> >>> On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon wrote: >>> > i monitored the line using another phone. there's indeed dialtone in >>> all >>> > attempts. >>> > i see TONE_DETECTED in the first call but i wonder there's a WARNING >>> message >>> > immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() >>> Unhandled >>> > type for channel 2:1. >>> > the dialtone freq should be okay since it's detected in the first >>> call.could >>> > the WARNING message gives us a hint of a possible problem other than >>> the >>> > dialtone freq? >>> > >>> > okay, i'll try the SVN version next. >>> > >>> > >>> > On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale >>> > wrote: >>> >> >>> >> Its not detecting a dial tone on the failure case. >>> >> Before dialing it waits until it picks up dialtone. >>> >> Try the svn trunk version to see if it works any better or verify >>> there is >>> >> a dialtone on the line. >>> >> >>> >> On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: >>> >> >>> >> hi everybody, >>> >> >>> >> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working >>> using >>> >> IP phones, softphones and digium FXS port. but there's a problem in >>> dialing >>> >> out to PSTN using digium tdm400 fxo - it works fine on the first >>> attempt >>> >> (after starting FS) but it fails on the subsequent attempts. i tested >>> to >>> >> call using the FXS port and IP phone. same problem. >>> >> >>> >> before i place any call, i checked >oz dump 2 1 (show current state = >>> >> DOWN, last state = DOWN) >>> >> >>> >> in the first call, there's this message: >>> >> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for >>> channel >>> >> 2:1 >>> >> but >>> >> >>> >> then i hangup. checked >oz dump 21 (show current state=DOWN, last >>> >> state=HANGUP) >>> >> >>> >> in the 2nd (and subsequent) attempts, the fxo just goes off-hook but >>> >> doesn't send the dtmf tones. >>> >> >oz dump 2 1 (shows current state = DIALING, last state = DOWN) >>> >> >>> >> has anyone encountered this problem before? i appreciate for any help >>> to >>> >> correct this problem. >>> >> >>> >> tks, >>> >> nandy >>> >> >>> >> >>> >> Environment: >>> >> ================== >>> >> kernel 2.6.18-92.1.22.el5 >>> >> FS 1.0.2 >>> >> zaptel 1.4.11 >>> >> oslec >>> >> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) >>> >> >>> >> zaptel.conf >>> >> ======== >>> >> loadzone = us >>> >> defaultzone=us >>> >> channels=1-2 >>> >> alaw=1-4 >>> >> fxsks=2 >>> >> fxoks=1 >>> >> >>> >> >>> >> openzap.conf.xml: >>> >> =============== >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> openzap.conf >>> >> ========== >>> >> [span zt] >>> >> name => OpenZAP FXS >>> >> number => 1 >>> >> fxs-channel => 1 >>> >> >>> >> [span zt] >>> >> name => OpenZAP FXO >>> >> number => 2 >>> >> fxo-channel => 2 >>> >> >>> >> tones.conf (the dialtone and ring tone is set to Philipping tones) >>> >> ======== >>> >> [us] >>> >> generate-dial => v=-7;%(1000,0,425) >>> >> detect-dial => 425 >>> >> >>> >> generate-ring => v=-7;%(1000,4000,425,480) >>> >> detect-ring => 425,480 >>> >> >>> >> generate-busy => v=-7;%(500,500,480,620) >>> >> detect-busy => 480,620 >>> >> >>> >> generate-attn => v=0;%(200,300,1400,1800) >>> >> detect-attn => 1400,1800 >>> >> >>> >> generate-callwaiting-sas => v=0;%(300,10000,440) >>> >> detect-callwaiting-sas => 440 >>> >> >>> >> generate-callwaiting-cas => v=0;%(80,0,2750,2130) >>> >> detect-callwaiting-cas => 2750,2130 >>> >> >>> >> detect-fail1 => 913.8 >>> >> detect-fail2 => 1370.6 >>> >> detect-fail3 => 776.7 >>> >> >>> >> LOG OF FIRST CALL (OK) >>> >> ==================== >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 >>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>> >> bridge(openzap/2/1/3400534) >>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>> PCMU >>> >> 20ms >>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 >>> channel_outgoing_channel() >>> >> Connect outbound channel OpenZAP/2:1/3400534 >>> >> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 >>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>> >> [e5f12114-ea88-11dd-9f5c-290fb4a527a4] >>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 >>> channel_outgoing_channel() >>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>> >> switch_core_session_signal_state_change() Send signal >>> OpenZAP/2:1/3400534 >>> >> [BREAK] >>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 >>> analog_fxo_outgoing_call() >>> >> Changing state on 2:1 from DOWN to DIALING >>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 >>> zap_analog_channel_run() >>> >> ANALOG CHANNEL thread starting. >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>> CS_INIT >>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 >>> zap_analog_channel_run() >>> >> Executing state handler on 2:1 for DIALING >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() >>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>> >> switch_core_session_signal_state_change() Send signal >>> OpenZAP/2:1/3400534 >>> >> [BREAK] >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>> sleep >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>> >> CS_ROUTING >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() >>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>> >> 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 >>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING >>> -> >>> >> CS_CONSUME_MEDIA >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>> >> switch_core_session_signal_state_change() Send signal >>> OpenZAP/2:1/3400534 >>> >> [BREAK] >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to >>> sleep >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>> >> CS_CONSUME_MEDIA >>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>> >> 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 >>> zap_analog_channel_run() >>> >> Detected tone DIAL on 2:1 >>> >> 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO >>> sig >>> >> 2:1 [TONE_DETECTED] >>> >> 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() >>> Unhandled >>> >> type for channel 2:1 >>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 zchan_activate_dtmf_buffer() >>> >> Created DTMF Buffer! >>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 GENERATE >>> DTMF >>> >> [3400534] >>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 >>> zap_analog_channel_run() >>> >> Changing state on 2:1 from DIALING to UP >>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>> zap_analog_channel_run() >>> >> Executing state handler on 2:1 for UP >>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO >>> sig >>> >> 2:1 [UP] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 >>> >> switch_channel_perform_mark_answered() Send signal >>> OpenZAP/1:1/93400534 >>> >> [BREAK] >>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() >>> Channel >>> >> [OpenZAP/2:1/3400534] has been answered >>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>> [AUDIO_SYNC] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 >>> >> switch_channel_perform_answer() OpenZAP/1:1/93400534 receive message >>> >> [ANSWER] >>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 >>> >> channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP >>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 >>> >> channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has been >>> >> answered >>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>> >> [AUDIO_SYNC] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>> >> switch_core_session_perform_receive_message() Send signal >>> >> OpenZAP/1:1/93400534 [BREAK] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 >>> >> switch_ivr_originate() Originate Resulted in Success: >>> [OpenZAP/2:1/3400534] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>> [AUDIO_SYNC] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>> >> [AUDIO_SYNC] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 >>> >> switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive message >>> >> [BRIDGE] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>> >> switch_core_session_perform_receive_message() Send signal >>> >> OpenZAP/2:1/3400534 [BREAK] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 >>> >> switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive >>> message >>> >> [BRIDGE] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>> >> switch_core_session_perform_receive_message() Send signal >>> >> OpenZAP/1:1/93400534 [BREAK] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 >>> >> switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State Change >>> >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 >>> >> switch_core_session_signal_state_change() Send signal >>> OpenZAP/2:1/3400534 >>> >> [BREAK] >>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>> zap_analog_channel_run() >>> >> Executing state handler on 1:1 for UP >>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS >>> sig >>> >> [UP] >>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>> going to >>> >> sleep >>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>> >> CS_EXCHANGE_MEDIA >>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 >>> channel_on_exchange_media() >>> >> CHANNEL EXCHANGE_MEDIA >>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() EVENT >>> >> [ONHOOK][1:1] STATE [UP] >>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() >>> Changing >>> >> state on 1:1 from UP to DOWN >>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>> zap_analog_channel_run() >>> >> Executing state handler on 1:1 for DOWN >>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS >>> sig >>> >> [STOP] >>> >> 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup >>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>> [KILL] >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>> >> switch_core_session_signal_state_change() Send signal >>> OpenZAP/1:1/93400534 >>> >> [BREAK] >>> >> 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>> done >>> >> 1:1 >>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 >>> zap_analog_channel_run() >>> >> ANALOG CHANNEL 1:1 thread ended. >>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 >>> audio_bridge_thread() >>> >> OpenZAP/1:1/93400534 ending bridge by request from read function >>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>> audio_bridge_thread() >>> >> BRIDGE THREAD DONE [OpenZAP/1:1/93400534] >>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>> audio_bridge_thread() >>> >> Send signal OpenZAP/2:1/3400534 [BREAK] >>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 >>> audio_bridge_thread() >>> >> OpenZAP/1:1/93400534 ending bridge by request from write function >>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 >>> audio_bridge_thread() >>> >> OpenZAP/2:1/3400534 receive message [UNBRIDGE] >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 >>> >> switch_core_session_perform_receive_message() Send signal >>> >> OpenZAP/2:1/3400534 [BREAK] >>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>> audio_bridge_thread() >>> >> BRIDGE THREAD DONE [OpenZAP/2:1/3400534] >>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>> audio_bridge_thread() >>> >> Send signal OpenZAP/1:1/93400534 [BREAK] >>> >> 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 >>> >> audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 >>> >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>> >> switch_core_session_signal_state_change() Send signal >>> OpenZAP/2:1/3400534 >>> >> [BREAK] >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >>> going >>> >> to sleep >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>> >> CS_HANGUP >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>> Changing >>> >> state on 2:1 from UP to HANGUP >>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, >>> cause: >>> >> NORMAL_CLEARING >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to >>> sleep >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Locked, >>> Waiting >>> >> on external entities >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 >>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going >>> to >>> >> sleep >>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended >>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>> [CS_HANGUP] >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change >>> >> CS_HANGUP >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, >>> >> cause: NORMAL_CLEARING >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to >>> sleep >>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Locked, >>> >> Waiting on external entities >>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended >>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>> [CS_HANGUP] >>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>> zap_analog_channel_run() >>> >> Executing state handler on 2:1 for HANGUP >>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 >>> zap_analog_channel_run() >>> >> Changing state on 2:1 from HANGUP to DOWN >>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 >>> zap_analog_channel_run() >>> >> Executing state handler on 2:1 for DOWN >>> >> 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO >>> sig >>> >> 2:1 [STOP] >>> >> 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>> done >>> >> 2:1 >>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 >>> zap_analog_channel_run() >>> >> ANALOG CHANNEL 2:1 thread ended. >>> >> >>> >> LOG OF FAILED CALLS >>> >> ================== >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 >>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>> >> bridge(openzap/2/1/3400534) >>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>> PCMU >>> >> 20ms >>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 >>> channel_outgoing_channel() >>> >> Connect outbound channel OpenZAP/2:1/3400534 >>> >> 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 >>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>> >> [079f5420-ea89-11dd-9f5c-290fb4a527a4] >>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 >>> channel_outgoing_channel() >>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>> >> switch_core_session_signal_state_change() Send signal >>> OpenZAP/2:1/3400534 >>> >> [BREAK] >>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 >>> analog_fxo_outgoing_call() >>> >> Changing state on 2:1 from DOWN to DIALING >>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 >>> zap_analog_channel_run() >>> >> ANALOG CHANNEL thread starting. >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>> CS_INIT >>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 >>> zap_analog_channel_run() >>> >> Executing state handler on 2:1 for DIALING >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() >>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>> >> switch_core_session_signal_state_change() Send signal >>> OpenZAP/2:1/3400534 >>> >> [BREAK] >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>> sleep >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>> >> CS_ROUTING >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() >>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>> >> 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 >>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING >>> -> >>> >> CS_CONSUME_MEDIA >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>> >> switch_core_session_signal_state_change() Send signal >>> OpenZAP/2:1/3400534 >>> >> [BREAK] >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going to >>> sleep >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>> >> CS_CONSUME_MEDIA >>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() EVENT >>> >> [ONHOOK][1:1] STATE [IDLE] >>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() >>> Changing >>> >> state on 1:1 from IDLE to DOWN >>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>> zap_analog_channel_run() >>> >> Executing state handler on 1:1 for DOWN >>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got FXS >>> sig >>> >> [STOP] >>> >> 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() Hangup >>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>> [KILL] >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>> >> switch_core_session_signal_state_change() Send signal >>> OpenZAP/1:1/93400534 >>> >> [BREAK] >>> >> 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>> done >>> >> 1:1 >>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 >>> zap_analog_channel_run() >>> >> ANALOG CHANNEL 1:1 thread ended. >>> >> 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 >>> >> switch_ivr_originate() Hangup OpenZAP/2:1/3400534 [CS_CONSUME_MEDIA] >>> >> [ORIGINATOR_CANCEL] >>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 [KILL] >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>> >> switch_core_session_signal_state_change() Send signal >>> OpenZAP/2:1/3400534 >>> >> [BREAK] >>> >> 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 >>> >> switch_ivr_originate() Originate Cancelled by originator termination >>> Cause: >>> >> 487 [ORIGINATOR_CANCEL] >>> >> 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 audio_bridge_function() >>> >> Originate Failed. Cause: ORIGINATOR_CANCEL >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 >>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going >>> to >>> >> sleep >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change >>> >> CS_HANGUP >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard HANGUP, >>> >> cause: NORMAL_CLEARING >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going to >>> sleep >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Locked, >>> >> Waiting on external entities >>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>> going to >>> >> sleep >>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>> [CS_HANGUP] >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>> >> CS_HANGUP >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>> Changing >>> >> state on 2:1 from DIALING to HANGUP >>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, >>> cause: >>> >> ORIGINATOR_CANCEL >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to >>> sleep >>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Locked, >>> Waiting >>> >> on external entities >>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended >>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>> [CS_HANGUP] >>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>> zap_analog_channel_run() >>> >> Executing state handler on 2:1 for HANGUP >>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 >>> zap_analog_channel_run() >>> >> Changing state on 2:1 from HANGUP to DOWN >>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 >>> zap_analog_channel_run() >>> >> Executing state handler on 2:1 for DOWN >>> >> 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got FXO >>> sig >>> >> 2:1 [STOP] >>> >> 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>> done >>> >> 2:1 >>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 >>> zap_analog_channel_run() >>> >> ANALOG CHANNEL 2:1 thread ended. >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at 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 at 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 at 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 at 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 at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090127/f56d4f92/attachment-0001.html From chavpaskov at shaw.ca Mon Jan 26 16:57:25 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Mon, 26 Jan 2009 16:57:25 -0800 Subject: [Freeswitch-users] [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock In-Reply-To: References: <497E4558.1030202@shaw.ca> <016801c9800d$ce4701e0$6ad505a0$@net> <2A3D5D5A-135C-4A39-8FC7-DDFF9436EA4B@freeswitch.org> <497E4A4E.9060803@shaw.ca> <9C3E0152-4D2D-48FD-B718-AE8FF86A2AA9@freeswitch.org> <016e01c98013$c37fb3f0$4a7f1bd0$@net> Message-ID: <497E5BF5.5030503@shaw.ca> Brian West wrote: > haha so was that one! :P Anchors Away!!! > > /b > > On Jan 26, 2009, at 6:11 PM, Adam Long wrote: > >> Sorry that last link was mangled... >> >> If you are running inside ESX might want to have a look here... >> http://kb.vmware.com/selfservice/microsites/search.do?language=en_US&cmd=dis >> >> playKC&externalId=2219 >> >> >> -Adam > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > thanks Adam i think i narrowed it down. this is rather interesting actually. have a look what i found: on the same ESXi i have 3 VMs Centos 5 running 2.6.18-53.el5 - no clock slipping issues Ubuntu 8.04.1 - 2.6.24-19- virtual clock is running faster Ubuntu 8.04.1 - 2.6.24-22-virtual clock running faster So now question is : is this a Distro related or kernel related issue ? /so i guess i'll have to buy a real anchor for my boat :P/ i see that this subject is probably for another forum but probably makes sense to mention that when it comes to freeswitch deployment ,if you do not want to deal with timers and so on you might just pick the right Distro. as for me... i guess i just learned something new ... so today was not lost . Thank you Brian / Adam. Best Regards Chav From ronmccar at gmail.com Mon Jan 26 17:05:20 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Mon, 26 Jan 2009 18:05:20 -0700 Subject: [Freeswitch-users] Another way to set CallerID? Message-ID: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> Hi, I am having a weird issue with setting the callerID number for outbound calls, I have this: Ive set the callerID correcly on a Asterisk box and the carrier sees it and passed it correct, but for some reason any calls from Freeswitch won't work, it either shows up as private or another number, so they are not seeing the CID for whatever reason. This method works for other carrirers just fine, but not them. From what I read this sets the from in the SIP message, maybe for some reason that is goofing it up. Any other way to set the callerID in Freeswitch, that's the only way I have found. FS shows it's connecting to a Sonus, does it need to be set different because of some Sonus issue that im not aware of? Any help would be great. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090126/623c4d3b/attachment.html From steveu at coppice.org Mon Jan 26 18:46:12 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 27 Jan 2009 10:46:12 +0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org><4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br><87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> <497E17F5.2020400@devel-it.com.br> Message-ID: <497E7574.2070907@coppice.org> Hi Abdul, Abdul Hakeem wrote: > Is Brazil a 3rd world country ? The last I hear Brazil was building > aeroplanes, has it's own space and nuclear program and a GNP UK would be > envious of. > Cheers, > AH > What relevance does that have to the current discussion? Brazil is a country with large trade barriers, which skews the cost of hardware from the world market considerably. It is also pretty advanced, technically, and has a local base of electronics manufacturers. That considerably affects the economic tradeoffs in the use computers, telecoms, and other technology in Brazil. If something can be manufactured (or at least pass through final assembly) in Brazil, it will generally be much cheaper than something imported. That means some people find the use of locally made intelligent E1 cards is cheaper than the use of dumb cards from Digium or Sangoma. Regards, Steve From jmesquita at gmail.com Mon Jan 26 19:34:46 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 27 Jan 2009 01:34:46 -0200 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <497E7574.2070907@coppice.org> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org><4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br><87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> <497E17F5.2020400@devel-it.com.br> <497E7574.2070907@coppice.org> Message-ID: Steve, As we speak I am actually negotiating with one of those companies to make a mod for their cards. Khomp has a very nice product and they are exporting to the rest of latin america now. Thanks, Mesquita On Jan 27, 2009, at 12:46 AM, Steve Underwood wrote: > Hi Abdul, > > Abdul Hakeem wrote: >> Is Brazil a 3rd world country ? The last I hear Brazil was building >> aeroplanes, has it's own space and nuclear program and a GNP UK >> would be >> envious of. >> Cheers, >> AH >> > > What relevance does that have to the current discussion? > > Brazil is a country with large trade barriers, which skews the cost of > hardware from the world market considerably. It is also pretty > advanced, > technically, and has a local base of electronics manufacturers. That > considerably affects the economic tradeoffs in the use computers, > telecoms, and other technology in Brazil. If something can be > manufactured (or at least pass through final assembly) in Brazil, it > will generally be much cheaper than something imported. That means > some > people find the use of locally made intelligent E1 cards is cheaper > than > the use of dumb cards from Digium or Sangoma. > > Regards, > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveu at coppice.org Mon Jan 26 20:10:58 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 27 Jan 2009 12:10:58 +0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org><4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br><87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> <497E17F5.2020400@devel-it.com.br> <497E7574.2070907@coppice.org> Message-ID: <497E8952.8030300@coppice.org> Hi Jo?o, Jo?o Mesquita wrote: > Steve, > > As we speak I am actually negotiating with one of those companies to > make a mod for their cards. Khomp has a very nice product and they are > exporting to the rest of latin america now. > It surprises me someone doesn't assemble Tormenta 2 cards in Brazil. Various people have done this in China, and shipped a lot of cards. They have to be a lot cheaper than the Khomp and Digivoice boards. The key issue with the Tormenta 2 is the lack of on board echo cancellation. Although OSLEC performs well, 120 channels of EC is a lot to do on the host CPU. Still, a lot of cards ship without on board EC. Regards, Steve From jgarland at jasongarland.com Mon Jan 26 21:29:57 2009 From: jgarland at jasongarland.com (Jason Garland) Date: Tue, 27 Jan 2009 00:29:57 -0500 Subject: [Freeswitch-users] Another way to set CallerID? In-Reply-To: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> References: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> Message-ID: <9A5CC41C-C0F0-41B5-AFF3-AE62B54C765B@jasongarland.com> You may need to set P-Asserted-Identity or Remote-Party-ID headers Sent from my iPhone On Jan 26, 2009, at 8:05 PM, Ron McCarthy wrote: > Hi, > > I am having a weird issue with setting the callerID number for > outbound calls, I have this: > > data="effective_caller_id_number=17025551234"/> > > > Ive set the callerID correcly on a Asterisk box and the carrier sees > it and passed it correct, but for some reason any calls from > Freeswitch won't work, it either shows up as private or another > number, so they are not seeing the CID for whatever reason. This > method works for other carrirers just fine, but not them. From what > I read this sets the from in the SIP message, maybe for some reason > that is goofing it up. Any other way to set the callerID in > Freeswitch, that's the only way I have found. > > FS shows it's connecting to a Sonus, does it need to be set > different because of some Sonus issue that im not aware of? > > Any help would be great. > > Thanks! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From raul at etellicom.com Mon Jan 26 22:32:44 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 27 Jan 2009 04:32:44 -0200 Subject: [Freeswitch-users] mod_g729 In-Reply-To: References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org><4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br> <87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> <497E17F5.2020400@devel-it.com.br> <497E7574.2070907@coppice.org> Message-ID: <1233037964.8161.30.camel@stargate> Hi Jo?o, Please say hello to Giancarlo at Khomp for me :) Khomp is the best example in Brazil of what good engineering and fair commercial prices can do to a country that suffers from high import taxes. Unfortunately some companies are still inclined to buy imported products/brands instead of favouring local production. Apart from this discriminatory issue, I believe that local research and production can turn the table of trade barriers imposed to development countries. By the way, are you aware of any companies producing SIP phones in Brazil ? Regards, Raul Fragoso On Tue, 2009-01-27 at 01:34 -0200, Jo?o Mesquita wrote: > Steve, > > As we speak I am actually negotiating with one of those companies to > make a mod for their cards. Khomp has a very nice product and they are > exporting to the rest of latin america now. > > Thanks, > > Mesquita > > On Jan 27, 2009, at 12:46 AM, Steve Underwood wrote: > > > Hi Abdul, > > > > Abdul Hakeem wrote: > >> Is Brazil a 3rd world country ? The last I hear Brazil was building > >> aeroplanes, has it's own space and nuclear program and a GNP UK > >> would be > >> envious of. > >> Cheers, > >> AH > >> > > > > What relevance does that have to the current discussion? > > > > Brazil is a country with large trade barriers, which skews the cost of > > hardware from the world market considerably. It is also pretty > > advanced, > > technically, and has a local base of electronics manufacturers. That > > considerably affects the economic tradeoffs in the use computers, > > telecoms, and other technology in Brazil. If something can be > > manufactured (or at least pass through final assembly) in Brazil, it > > will generally be much cheaper than something imported. That means > > some > > people find the use of locally made intelligent E1 cards is cheaper > > than > > the use of dumb cards from Digium or Sangoma. > > > > Regards, > > Steve > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From stevecrozz at gmail.com Mon Jan 26 22:55:55 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 26 Jan 2009 22:55:55 -0800 Subject: [Freeswitch-users] javascript setHangupHook .. looking for a working example Message-ID: <11990ade0901262255x14f010eew122bda9e0f15f6cf@mail.gmail.com> I'm running rev 11131, and I cannot make these hangup events work in javascript... does anyone have a working example? I tried this script from the wiki: http://wiki.freeswitch.org/wiki/Example_Hangup_hook --Stephen From steveu at coppice.org Mon Jan 26 23:23:50 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 27 Jan 2009 15:23:50 +0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <1233037964.8161.30.camel@stargate> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org><4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br> <87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> <497E17F5.2020400@devel-it.com.br> <497E7574.2070907@coppice.org> <1233037964.8161.30.camel@stargate> Message-ID: <497EB686.9080504@coppice.org> Raul Fragoso wrote: > Hi Jo?o, > > Please say hello to Giancarlo at Khomp for me :) > > Khomp is the best example in Brazil of what good engineering and fair > commercial prices can do to a country that suffers from high import > taxes. Unfortunately some companies are still inclined to buy imported > products/brands instead of favouring local production. Apart from this > discriminatory issue, I believe that local research and production can > turn the table of trade barriers imposed to development countries. > Foreigners face huge tariff barriers trying to sell into Brazil. Brazilian makers face little difficulty selling into other markets. If Brazilian makers have any problems in such an environment, I'd be amused to hear them. :-) Steve From pmhshz at gmail.com Mon Jan 26 23:40:53 2009 From: pmhshz at gmail.com (shehzad p) Date: Mon, 26 Jan 2009 23:40:53 -0800 (PST) Subject: [Freeswitch-users] ATA-answering machine question/recommendation In-Reply-To: <87f2f3b90901260203xd447883xe373244207c2fc45@mail.gmail.com> References: <207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45@mail.gmail.com> <1461348542-1232566409-cardhu_decombobulator_blackberry.rim.net-569820458-@bxe161.bisx.prod.on.blackberry> <21640813.post@talk.nabble.com> <87f2f3b90901260203xd447883xe373244207c2fc45@mail.gmail.com> Message-ID: <21680924.post@talk.nabble.com> Hi Michael, My dial plan is: ======================== ==================================== When call come to xxx.xxx.xxx.x system, It answer the call and only wait for 20 seconds (NO playback only wait) and hangup. Please find debug trace for above dialplan on http://pastebin.freeswitch.org/6919 Again I also tested dialplan exact same as shown in the wiki page http://wiki.freeswitch.org/wiki/Mod_vmd#From_Dialplan. But there it is not working as needed. Thanks for your response... msp Michael Collins-11 wrote: > > Did you ever post your dialplan and a debug trace of a call to the > pastebin? If not, please do so and we will check it out. > -MC > > On Sat, Jan 24, 2009 at 5:47 AM, shehzad p wrote: >> >> Hi all, >> >> On my existing Freeswitch 1.0.2, I installed and configured mod_vmd as >> below: >> make mod_vmd-install >> >> Then configured it as on wiki page: >> http://wiki.freeswitch.org/wiki/Mod_vmd >> >> >> After that my dialplan terminates call to another system, where it is >> just >> answered and wait for some time there. >> So that there should be a variable called vmd_detect must be created as >> shown in dialplan http://wiki.freeswitch.org/wiki/Mod_vmd#From_Dialplan. >> >> But eventhough no variable named 'vmd_detect' is created after that.!!! >> Is there i am missing something? Is there another way of using mod_vmd? >> >> Thanks in advance. >> msp >> >> >> >> >> >> >> Lucas Cornelisse wrote: >>> >>> Hi Jonathan, >>> >>> Mod_vmd (voicemail detection) should do the trick. >>> >>> Just search the wiki for mod_vmd, there are a number of ways of using >>> it. >>> >>> >>> Sent from my BlackBerry device on the Rogers Wireless Network >>> >>> -----Original Message----- >>> From: jonathan augenstine >>> >>> Date: Wed, 21 Jan 2009 06:53:41 >>> To: >>> Subject: [Freeswitch-users] ATA-answering machine >>> question/recommendation >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/ATA-answering-machine-question-recommendation-tp21584795p21640813.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/ATA-answering-machine-question-recommendation-tp21584795p21680924.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gmaruzz at celliax.org Tue Jan 27 00:02:39 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 27 Jan 2009 09:02:39 +0100 Subject: [Freeswitch-users] Call out to gtalk, asterisk and skype In-Reply-To: <9dc4a1670901261608g2ded99cchd99fcce55e7bf1ac@mail.gmail.com> References: <9dc4a1670901261608g2ded99cchd99fcce55e7bf1ac@mail.gmail.com> Message-ID: <7b197bef0901270002m66968e6ep7fc6d0191a18748c@mail.gmail.com> Ciao Ed, for Skype, you can have a look at: http://wiki.freeswitch.org/wiki/Skypiax Feel free to ask for more info if the wiki page is not complete/clear. Ciao for now, Giovanni 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 Tue, Jan 27, 2009 at 1:08 AM, EdPimentl wrote: > Hello Everyone, > > Has anyone done a scalable deployment of calling to Skype/Gtalk/Asterisk > Say many 3-25/50 users with various endpoint > (skype/gtalk/asterisk)platforms. > > http://blog.tmcnet.com/blog/tom-keating/asterisk/skype-for-asterisk-launches.asp > http://www.mhspot.com/siptheeskype_skype_trunk_howto.html > > Thanks in advance, > -E > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From telles-listas at devel-it.com.br Tue Jan 27 04:50:08 2009 From: telles-listas at devel-it.com.br (Rodrigo P. Telles) Date: Tue, 27 Jan 2009 10:50:08 -0200 Subject: [Freeswitch-users] mod_g729 In-Reply-To: References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org><4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br><87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> <497E17F5.2020400@devel-it.com.br> Message-ID: <497F0300.8060607@devel-it.com.br> Hi Abdul, Brazil does much more than that but we still on "latim america" and according with 1st world countries, we are considered "non developed country". The most funny thing about that: some people still thinks that we (brazillians) lives in the forest with monkeys :-) Regards, Rodrigo Telles Em 26-01-2009 19:30, Abdul Hakeem escreveu: > Is Brazil a 3rd world country ? The last I hear Brazil was building > aeroplanes, has it's own space and nuclear program and a GNP UK would be > envious of. > Cheers, > AH > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rodrigo > P. Telles > Sent: 26 January 2009 20:07 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_g729 > > Hi Michael, > > I'm not sure about this, when we need to use G729 on asterisk for example, > we pay the digium (U$ 10/channel) licenses. > U$ 1.00 dolar = R$ 2.31 (Brazilian Real - local currency). > > Att., > > Em 23-01-2009 18:29, Michael Collins escreveu: >> On Fri, Jan 23, 2009 at 12:16 PM, Rodrigo P. Telles >> wrote: >>> Hi Dave, >>> >>> Down here in Brazil, the bandwidth costs is very high (around U$ >>> 400.00/Mb) so it should be valid only for a "non" third world country. >>> G729 and G723.1 is almost a law here, if you don't play at least with > G729 your ITSP is out of mark share! >>> My 2 cents from a third world country. >> What is the patent and licensing situation in Brazil? Those are also >> factors. $10/port might be cheap in the US but in Brazil it could be >> much more? (I'm asking...) -MC From regs at kinetix.gr Tue Jan 27 05:05:44 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 27 Jan 2009 15:05:44 +0200 Subject: [Freeswitch-users] mod_radius_cdr questions and thoughts In-Reply-To: <4978C200.7030403@kinetix.gr> References: <4978A2B2.2020905@kinetix.gr> <6569777d0901220909q99e80abvb4c6253afb65dc81@mail.gmail.com> <4978C200.7030403@kinetix.gr> Message-ID: <497F06A8.8070507@kinetix.gr> Hi, After some testing I came to the following conclusions : 1) The problem (timeouts and retries) I describe below only happens when there is no radius server responding on the other side. 2) It only happens when using the latest cvs version of radiusclient. If you use version 1.1.6 it works fine. I also read in the wiki (and found out myself by testing) that : "Currently, the module blocks the thread while it is sending the requests. This may cause threads to hang around longer than expected after a call, if your RADIUS servers are not reachable/responding." which I think is not desirable. Was this kind of behavior, been followed intentionally? I think that the NAS in most (if not all) implementations uses a non-blocking operation in order to proceed with the call. In that way there is not any significant delay (up to 15 seconds if radius is down) in the beginning of the call. Also, I noticed that if the radius acct packet fails, FS does not proceed with the call which is again -in my opinion - wrong. I think that the NAS should be able to continue with the call even if the Acct start or stop failed. For those directly involved in the maintenance of the mod_radius_cdr code : Is it relatively easy to change the blocking behavior of the module? Apostolos Pantsiopoulos wrote: > Chris Parker wrote: >> On Thu, Jan 22, 2009 at 10:45 AM, Apostolos Pantsiopoulos >> > wrote: >> >> I am trying to implement a radius based solution >> using FS. I have seen that the mod_radius_cdr module >> is actively maintained. so I have a few questions/remarks : >> >> 1) When I place a call and my radius server is down, the >> call blocks forever instead of just radius_timeout * radius_retries >> seconds (I have declared only one server). I would expect that >> FS would stop trying to send an Acc Start packet after some >> time and get on with the call. >> >> >> I have not seen this behavior. If you can duplicate this, and >> propose a patch, it would be gladly welcomed. > I rebuilt and retried and the behavior persists. > > The call progress freezes and I get the following in the log : > > 2009-01-22 20:48:32 [DEBUG] switch_core_state_machine.c:435 > switch_core_session_run() (sofia/internal/9333 at xxx.xxx.xxx.xxx) State > ROUTING > 2009-01-22 20:48:32 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/internal/9333 at xx.xxx.xxx.xxx SOFIA ROUTING > 2009-01-22 20:48:32 [DEBUG] mod_radius_cdr.c:152 my_on_routing() > [mod_radius_cdr] Entering my_on_routing > > After I hangup the client and issue a shutdown in FS I get the > following : > > 2009-01-22 20:50:50 [CRIT] sofia.c:794 sofia_profile_thread_run() > Waiting for 1 session(s) > > repeatedly and FS never exits. >> >> >> >> 2) I have also noticed that FS sends only 1 packet (I waited for >> a minute) >> instead of 3 (default in the config) since the first (and second) >> attempt failed. >> If my server was up (the port was responding) but it returned a >> req. failed >> answer would the above time-out be valid? >> >> >> I have not seen this behavior. > The same here after the rebuild. >> >> >> >> 3) When I tried to load the dictionary.freeswitch to my freeradius >> server, it complained : >> >> >> Don't do that. The dictionary is for use with the radiusclient >> library. FreeRADIUS already includes a dictionary for FreeSWITCH >> VSAs ( you may need to uncomment it to have it loaded into FreeRADIUS ). > I cannot find any reference to Freeswitch in the freeradius integrated > dictionaries (in the share folder). Can you pinpoint the > directory that a dictionary.freeswitch (or other FS related > dictionary) resides? >> >> >> 4) The radius attributes included in the current requests are >> a) hard-coded, b) limited in number. I think many of us would like to >> use more attributes. Or even better define what to include (and >> what to >> put in them) using a >> config file (the same maybe?) >> >> >> This has been proposed. There isn't yet a mechanism, though the >> intent is to use a general purpose FS VSA for this. The code needs >> to be added to the mod_radius_cdr module to allow that to be a >> run_time configuration option. > A general purpose VSA that holds only one value or many? Or a mix > (array like)? >> >> >> 5) Does the module send accounting packets only for the a-leg >> of a call or for both legs? (Maybe that could be configurable too). >> >> If anyone is interested in the above questions/remarks please post >> a reply. I would really like to know how many of the mailing list >> users >> are also interested in FS radius support and your opinions on the >> matter. >> >> >> Again, patches are welcome. :) >> >> -Chris >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090127/92facf94/attachment-0001.html From raul at etellicom.com Tue Jan 27 06:04:44 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 27 Jan 2009 12:04:44 -0200 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <497EB686.9080504@coppice.org> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org><4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br> <87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> <497E17F5.2020400@devel-it.com.br> <497E7574.2070907@coppice.org> <1233037964.8161.30.camel@stargate> <497EB686.9080504@coppice.org> Message-ID: <1233065084.8161.48.camel@stargate> Hi Steve, My point is more towards the high import taxes here. A foreign company won't pay as much taxes to export to Brazil as brazilians would pay import taxes for such goods. I will give you a simple example: a Snom 320 phone would cost roughly US $180.00 if bought in the US. Here it would cost at least double the price, but usually more. We pay almost 100% for importing taxes + GST, excluding the greedy profit that dealers and resellers usually apply, but Snom (or their distributors) would pay only a small fraction of that to export phones to Brazil. The irony is that we don't see where all this money is spent, as the country still suffers of a huge social and economical dissimilarity among their people. On the other hand, our current government seems to be making some progress to make exporting from Brazil easier, specially for small companies (international trade is known to be very bureaucratic here). Anyway, this is turning into a politics discussion thread and I know it can annoy some people. Just to stay in the subject thread, I also endorse the support for open codecs rather than paying millions for G.729 licenses. Regards, Raul Fragoso On Tue, 2009-01-27 at 15:23 +0800, Steve Underwood wrote: > Raul Fragoso wrote: > > Hi Jo?o, > > > > Please say hello to Giancarlo at Khomp for me :) > > > > Khomp is the best example in Brazil of what good engineering and fair > > commercial prices can do to a country that suffers from high import > > taxes. Unfortunately some companies are still inclined to buy imported > > products/brands instead of favouring local production. Apart from this > > discriminatory issue, I believe that local research and production can > > turn the table of trade barriers imposed to development countries. > > > Foreigners face huge tariff barriers trying to sell into Brazil. > Brazilian makers face little difficulty selling into other markets. If > Brazilian makers have any problems in such an environment, I'd be amused > to hear them. :-) > > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Tue Jan 27 06:14:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 27 Jan 2009 08:14:09 -0600 Subject: [Freeswitch-users] mod_radius_cdr questions and thoughts In-Reply-To: <497F06A8.8070507@kinetix.gr> References: <4978A2B2.2020905@kinetix.gr> <6569777d0901220909q99e80abvb4c6253afb65dc81@mail.gmail.com> <4978C200.7030403@kinetix.gr> <497F06A8.8070507@kinetix.gr> Message-ID: <191c3a030901270614i23a41bc2o1fbeaf576ab3811c@mail.gmail.com> Sounds like having cake and eating it too. The risk is obvious when using radius or some other additional protocol for AAA that you will have trouble if the server is down. Radius was designed to be fast and redundant, so typically you would have 5 radius servers not one. I did not make mod_radius_cdr but I do pose this question: Where should the burden lie to make sure the calls are not delayed when choosing this option? If you make FS cache all the radius requests it could not complete in a timely manner, the whole point of keeping track of the exact time they occurred is lost and you could have calls that ended before the start packet ever was transmitted because they are cached in some process that will begin to swell with memory remembering all the requests it could not send. Then somehow it needs to gracefully catch up again when the radius server comes back.... This is the same reason I think that direct database CDR is a bad idea. The real answer is that you are not allowed to have your radius server down at all, so you need more than one. I used to be in the dialup business and we had to have backup radius servers for the backup radius servers on a completely different network just in case not only the server was down but the network link to the server and it's backup server. It's like DNS, I can't ever be down or nothing works. On Tue, Jan 27, 2009 at 7:05 AM, Apostolos Pantsiopoulos wrote: > Hi, > > After some testing I came to the following conclusions : > > 1) The problem (timeouts and retries) I describe below only happens when > there is no radius server responding on the other side. > > 2) It only happens when using the latest cvs version of radiusclient. If > you use version 1.1.6 it works fine. > > I also read in the wiki (and found out myself by testing) that : > > "Currently, the module blocks the thread while it is sending the requests. > This may cause threads to hang around longer than expected after a call, if > your RADIUS servers are not reachable/responding." > > which I think is not desirable. Was this kind of behavior, been followed > intentionally? > I think that the NAS in most (if not all) implementations uses a > non-blocking operation > in order to proceed with the call. In that way there is not any significant > delay (up to 15 seconds if radius is down) > in the beginning of the call. > Also, I noticed that if the radius acct packet fails, FS does not proceed > with the call > which is again -in my opinion - wrong. I think that the NAS should be able > to continue with > the call even if the Acct start or stop failed. > > For those directly involved in the maintenance of the mod_radius_cdr code : > > Is it relatively easy to change the blocking behavior of the module? > > > Apostolos Pantsiopoulos wrote: > > Chris Parker wrote: > > On Thu, Jan 22, 2009 at 10:45 AM, Apostolos Pantsiopoulos > wrote: > >> I am trying to implement a radius based solution >> using FS. I have seen that the mod_radius_cdr module >> is actively maintained. so I have a few questions/remarks : >> >> 1) When I place a call and my radius server is down, the >> call blocks forever instead of just radius_timeout * radius_retries >> seconds (I have declared only one server). I would expect that >> FS would stop trying to send an Acc Start packet after some >> time and get on with the call. > > > I have not seen this behavior. If you can duplicate this, and propose a > patch, it would be gladly welcomed. > > I rebuilt and retried and the behavior persists. > > The call progress freezes and I get the following in the log : > > 2009-01-22 20:48:32 [DEBUG] switch_core_state_machine.c:435 > switch_core_session_run() (sofia/internal/9333 at xxx.xxx.xxx.xxx) State > ROUTING > 2009-01-22 20:48:32 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/internal/9333 at xx.xxx.xxx.xxx SOFIA ROUTING > 2009-01-22 20:48:32 [DEBUG] mod_radius_cdr.c:152 my_on_routing() > [mod_radius_cdr] Entering my_on_routing > > After I hangup the client and issue a shutdown in FS I get the following : > > 2009-01-22 20:50:50 [CRIT] sofia.c:794 sofia_profile_thread_run() Waiting > for 1 session(s) > > repeatedly and FS never exits. > > >> >> 2) I have also noticed that FS sends only 1 packet (I waited for a minute) >> instead of 3 (default in the config) since the first (and second) >> attempt failed. >> If my server was up (the port was responding) but it returned a req. >> failed >> answer would the above time-out be valid? > > > I have not seen this behavior. > > The same here after the rebuild. > > >> >> 3) When I tried to load the dictionary.freeswitch to my freeradius >> server, it complained : > > > Don't do that. The dictionary is for use with the radiusclient library. > FreeRADIUS already includes a dictionary for FreeSWITCH VSAs ( you may need > to uncomment it to have it loaded into FreeRADIUS ). > > I cannot find any reference to Freeswitch in the freeradius integrated > dictionaries (in the share folder). Can you pinpoint the > directory that a dictionary.freeswitch (or other FS related dictionary) > resides? > > > >> 4) The radius attributes included in the current requests are >> a) hard-coded, b) limited in number. I think many of us would like to >> use more attributes. Or even better define what to include (and what to >> put in them) using a >> config file (the same maybe?) > > > This has been proposed. There isn't yet a mechanism, though the intent is > to use a general purpose FS VSA for this. The code needs to be added to the > mod_radius_cdr module to allow that to be a run_time configuration option. > > A general purpose VSA that holds only one value or many? Or a mix (array > like)? > > > >> 5) Does the module send accounting packets only for the a-leg >> of a call or for both legs? (Maybe that could be configurable too). >> >> If anyone is interested in the above questions/remarks please post >> a reply. I would really like to know how many of the mailing list users >> are also interested in FS radius support and your opinions on the matter. >> > > Again, patches are welcome. :) > > > -Chris > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090127/794a16cf/attachment.html From helmut.kuper at ewetel.de Tue Jan 27 06:27:47 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 27 Jan 2009 15:27:47 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <497B0563.9090701@ewetel.de> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <497B0563.9090701@ewetel.de> Message-ID: <497F19E3.70900@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, again an update to my little project: I enhanced mod_openzap and ozmod_isdn so that I'm able to start and stop q931ToPcap generation in ozmod_isdn from FS's console. q931ToPcap itself isn't implemented yet. To give you an idea about the enhanced oz command - it's like this: freeswitch at ippbx-prod-node0> oz API CALL [oz()] output: list || dump [] || bounce [] || q931_pcap on|off freeswitch at ippbx-prod-node0> oz q931_pcap API CALL [oz(q931_pcap)] output: - -ERR Usage: oz q931_pcap on|off freeswitch at ippbx-prod-node0> oz q931_pcap 0 on API CALL [oz(q931_pcap 0 on)] output: - -ERR invalid span freeswitch at ippbx-prod-node0> oz q931_pcap 1 on API CALL [oz(q931_pcap 1 on)] output: 2009-01-27 15:21:55 [INFO] mod_openzap.c:2172 oz_function() Starting Q931-to-pcap 2009-01-27 15:21:55 [DEBUG] ozmod_isdn.c:1712 zap_isdn_configure_span() Enabling Q931ToPcap freeswitch at ippbx-prod-node0> oz q931_pcap 1 off API CALL [oz(q931_pcap 1 off)] output: 2009-01-27 15:21:58 [INFO] mod_openzap.c:2178 oz_function() Stopping Q931-to-pcap 2009-01-27 15:21:58 [DEBUG] ozmod_isdn.c:1717 zap_isdn_configure_span() Disabling Q931ToPcap freeswitch at ippbx-prod-node0> As you can see, ozmod_isdn, where I will implement the code receives the oz command option. I appreciate your comments. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl/GeMACgkQ4tZeNddg3dzurACgszPDA573+uZjmY5TWXF9yj5F Vz8An2FvK0FsDLr0JQaHFDETWnLoPSkD =0hkL -----END PGP SIGNATURE----- From regs at kinetix.gr Tue Jan 27 06:32:57 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 27 Jan 2009 16:32:57 +0200 Subject: [Freeswitch-users] mod_radius_cdr questions and thoughts In-Reply-To: <191c3a030901270614i23a41bc2o1fbeaf576ab3811c@mail.gmail.com> References: <4978A2B2.2020905@kinetix.gr> <6569777d0901220909q99e80abvb4c6253afb65dc81@mail.gmail.com> <4978C200.7030403@kinetix.gr> <497F06A8.8070507@kinetix.gr> <191c3a030901270614i23a41bc2o1fbeaf576ab3811c@mail.gmail.com> Message-ID: <497F1B19.6080500@kinetix.gr> Please see in line comments. Anthony Minessale wrote: > Sounds like having cake and eating it too. > > The risk is obvious when using radius or some other additional > protocol for AAA that you will have trouble if the server is down. > Radius was designed to be fast and redundant, so typically you would > have 5 radius servers not one. > > I did not make mod_radius_cdr but I do pose this question: > > Where should the burden lie to make sure the calls are not delayed > when choosing this option? Within the module. A separate thread could deal with the Acct Packet transmission without blocking the flow of the call. After a number of retries the thread should quit trying. > > If you make FS cache all the radius requests it could not complete in > a timely manner, the whole point of keeping track of the exact time > they occurred is lost and you could have calls that ended before the > start packet ever was transmitted because they are cached in some > process that will begin to swell with memory remembering all the > requests it could not send. The acct start and stop packets contain the time info needed for billing purposes Even if the radius packets reaches the server 10 secs after the end of the call there is little harm done. The "real" start and stop times don't have to correspond to the times the packets arrived at the radius server. > > Then somehow it needs to gracefully catch up again when the radius > server comes back.... This is the same reason I think that direct > database CDR is a bad idea. If the retries*timeout time has passed the NAS should give up trying to send the packet. So there is not much catching up to do. The radius packets that failed while the radius server was down don't have to be retransmitted later. All the applications and users that use radius are comfortable with that fact. I for one use a x-checking mechanism (comparing CDRs with radius created CDRs) to verify the integrity of my calls. > > The real answer is that you are not allowed to have your radius server > down at all, so you need more than one. I used to be in the dialup > business and we had to have backup radius servers for the backup > radius servers on a completely different network just in case not only > the server was down but the network link to the server and it's backup > server. It's like DNS, I can't ever be down or nothing works. I always use multiple radius servers (using different routes to my NASes). But sometimes there are other issues that could interfere with the NAS-Radius connectivity. > > > > On Tue, Jan 27, 2009 at 7:05 AM, Apostolos Pantsiopoulos > > wrote: > > Hi, > > After some testing I came to the following conclusions : > > 1) The problem (timeouts and retries) I describe below only > happens when there is no radius server responding on the other side. > > 2) It only happens when using the latest cvs version of > radiusclient. If you use version 1.1.6 it works fine. > > I also read in the wiki (and found out myself by testing) that : > > "Currently, the module blocks the thread while it is sending the > requests. This may cause threads to hang around longer than > expected after a call, if your RADIUS servers are not > reachable/responding." > > which I think is not desirable. Was this kind of behavior, been > followed intentionally? > I think that the NAS in most (if not all) implementations uses a > non-blocking operation > in order to proceed with the call. In that way there is not any > significant delay (up to 15 seconds if radius is down) > in the beginning of the call. > Also, I noticed that if the radius acct packet fails, FS does not > proceed with the call > which is again -in my opinion - wrong. I think that the NAS should > be able to continue with > the call even if the Acct start or stop failed. > > For those directly involved in the maintenance of the > mod_radius_cdr code : > > Is it relatively easy to change the blocking behavior of the module? > > > Apostolos Pantsiopoulos wrote: >> Chris Parker wrote: >>> On Thu, Jan 22, 2009 at 10:45 AM, Apostolos Pantsiopoulos >>> > wrote: >>> >>> I am trying to implement a radius based solution >>> using FS. I have seen that the mod_radius_cdr module >>> is actively maintained. so I have a few questions/remarks : >>> >>> 1) When I place a call and my radius server is down, the >>> call blocks forever instead of just radius_timeout * >>> radius_retries >>> seconds (I have declared only one server). I would expect that >>> FS would stop trying to send an Acc Start packet after some >>> time and get on with the call. >>> >>> >>> I have not seen this behavior. If you can duplicate this, and >>> propose a patch, it would be gladly welcomed. >> I rebuilt and retried and the behavior persists. >> >> The call progress freezes and I get the following in the log : >> >> 2009-01-22 20:48:32 [DEBUG] switch_core_state_machine.c:435 >> switch_core_session_run() (sofia/internal/9333 at xxx.xxx.xxx.xxx >> ) State ROUTING >> 2009-01-22 20:48:32 [DEBUG] mod_sofia.c:130 sofia_on_routing() >> sofia/internal/9333 at xx.xxx.xxx.xxx >> SOFIA ROUTING >> 2009-01-22 20:48:32 [DEBUG] mod_radius_cdr.c:152 my_on_routing() >> [mod_radius_cdr] Entering my_on_routing >> >> After I hangup the client and issue a shutdown in FS I get the >> following : >> >> 2009-01-22 20:50:50 [CRIT] sofia.c:794 sofia_profile_thread_run() >> Waiting for 1 session(s) >> >> repeatedly and FS never exits. >>> >>> >>> >>> 2) I have also noticed that FS sends only 1 packet (I waited >>> for a minute) >>> instead of 3 (default in the config) since the first (and >>> second) >>> attempt failed. >>> If my server was up (the port was responding) but it >>> returned a req. failed >>> answer would the above time-out be valid? >>> >>> >>> I have not seen this behavior. >> The same here after the rebuild. >>> >>> >>> >>> 3) When I tried to load the dictionary.freeswitch to my >>> freeradius >>> server, it complained : >>> >>> >>> Don't do that. The dictionary is for use with the radiusclient >>> library. FreeRADIUS already includes a dictionary for >>> FreeSWITCH VSAs ( you may need to uncomment it to have it loaded >>> into FreeRADIUS ). >> I cannot find any reference to Freeswitch in the freeradius >> integrated dictionaries (in the share folder). Can you pinpoint the >> directory that a dictionary.freeswitch (or other FS related >> dictionary) resides? >>> >>> >>> 4) The radius attributes included in the current requests are >>> a) hard-coded, b) limited in number. I think many of us >>> would like to >>> use more attributes. Or even better define what to include >>> (and what to >>> put in them) using a >>> config file (the same maybe?) >>> >>> >>> This has been proposed. There isn't yet a mechanism, though the >>> intent is to use a general purpose FS VSA for this. The code >>> needs to be added to the mod_radius_cdr module to allow that to >>> be a run_time configuration option. >> A general purpose VSA that holds only one value or many? Or a mix >> (array like)? >>> >>> >>> 5) Does the module send accounting packets only for the a-leg >>> of a call or for both legs? (Maybe that could be >>> configurable too). >>> >>> If anyone is interested in the above questions/remarks >>> please post >>> a reply. I would really like to know how many of the mailing >>> list users >>> are also interested in FS radius support and your opinions >>> on the matter. >>> >>> >>> Again, patches are welcome. :) >>> >>> -Chris >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090127/4cba67c6/attachment.html From brian at freeswitch.org Tue Jan 27 06:45:11 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 27 Jan 2009 08:45:11 -0600 Subject: [Freeswitch-users] javascript setHangupHook .. looking for a working example In-Reply-To: <11990ade0901262255x14f010eew122bda9e0f15f6cf@mail.gmail.com> References: <11990ade0901262255x14f010eew122bda9e0f15f6cf@mail.gmail.com> Message-ID: <81605515-5837-4A87-8F91-D177575EEA97@freeswitch.org> Can you describe what you're trying to do and how you're doing it? Maybe an example? /b On Jan 27, 2009, at 12:55 AM, Stephen Crosby wrote: > I'm running rev 11131, and I cannot make these hangup events work in > javascript... does anyone have a working example? I tried this script > from the wiki: http://wiki.freeswitch.org/wiki/Example_Hangup_hook > > --Stephen > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Tue Jan 27 07:15:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 27 Jan 2009 09:15:41 -0600 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <497F19E3.70900@ewetel.de> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <497B0563.9090701@ewetel.de> <497F19E3.70900@ewetel.de> Message-ID: <191c3a030901270715r538f45e4g615dffa639a63c20@mail.gmail.com> Looks good, you might want to consider making the file name that is generated be an optional argument but it's not necessary just a suggestion. On Tue, Jan 27, 2009 at 8:27 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > > again an update to my little project: > > I enhanced mod_openzap and ozmod_isdn so that I'm able to start and stop > q931ToPcap generation in ozmod_isdn from FS's console. q931ToPcap itself > isn't implemented yet. > > To give you an idea about the enhanced oz command - it's like this: > > freeswitch at ippbx-prod-node0> oz > API CALL [oz()] output: > list || dump [] || bounce [] || > q931_pcap on|off > freeswitch at ippbx-prod-node0> oz q931_pcap > API CALL [oz(q931_pcap)] output: > - -ERR Usage: oz q931_pcap on|off > > freeswitch at ippbx-prod-node0> oz q931_pcap 0 on > API CALL [oz(q931_pcap 0 on)] output: > - -ERR invalid span > > freeswitch at ippbx-prod-node0> oz q931_pcap 1 on > API CALL [oz(q931_pcap 1 on)] output: > > 2009-01-27 15:21:55 [INFO] mod_openzap.c:2172 oz_function() Starting > Q931-to-pcap > 2009-01-27 15:21:55 [DEBUG] ozmod_isdn.c:1712 zap_isdn_configure_span() > Enabling Q931ToPcap > freeswitch at ippbx-prod-node0> oz q931_pcap 1 off > API CALL [oz(q931_pcap 1 off)] output: > > 2009-01-27 15:21:58 [INFO] mod_openzap.c:2178 oz_function() Stopping > Q931-to-pcap > 2009-01-27 15:21:58 [DEBUG] ozmod_isdn.c:1717 zap_isdn_configure_span() > Disabling Q931ToPcap > freeswitch at ippbx-prod-node0> > > > > As you can see, ozmod_isdn, where I will implement the code receives the > oz command option. > > I appreciate your comments. > > > regards > helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkl/GeMACgkQ4tZeNddg3dzurACgszPDA573+uZjmY5TWXF9yj5F > Vz8An2FvK0FsDLr0JQaHFDETWnLoPSkD > =0hkL > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090127/db2fe87d/attachment.html From stevecrozz at gmail.com Tue Jan 27 08:35:54 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Tue, 27 Jan 2009 08:35:54 -0800 Subject: [Freeswitch-users] javascript setHangupHook .. looking for a working example In-Reply-To: <81605515-5837-4A87-8F91-D177575EEA97@freeswitch.org> References: <11990ade0901262255x14f010eew122bda9e0f15f6cf@mail.gmail.com> <81605515-5837-4A87-8F91-D177575EEA97@freeswitch.org> Message-ID: <11990ade0901270835l6f6d032dn257c40cda512f8e0@mail.gmail.com> Of course, I simply want to run this example script from the wiki: http://wiki.freeswitch.org/wiki/Example_Hangup_hook . I'm building a modified voicemail app. I want to fire a custom event when the user confirms the voicemail, but I want to fire the same event in case the user simply hangs up. The problem is, it appears that the method bound to session.setHangupHook is never executed. That's why I just want to see a working example of someone else's script that uses session.setHangupHook. --Stephen On Tue, Jan 27, 2009 at 6:45 AM, Brian West wrote: > Can you describe what you're trying to do and how you're doing it? > Maybe an example? > > /b > > On Jan 27, 2009, at 12:55 AM, Stephen Crosby wrote: > >> I'm running rev 11131, and I cannot make these hangup events work in >> javascript... does anyone have a working example? I tried this script >> from the wiki: http://wiki.freeswitch.org/wiki/Example_Hangup_hook >> >> --Stephen >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From curly2009 at gmx.de Tue Jan 27 08:41:04 2009 From: curly2009 at gmx.de (=?iso-8859-1?Q?=22Franziska_R=F6hler=22?=) Date: Tue, 27 Jan 2009 17:41:04 +0100 Subject: [Freeswitch-users] Can't dial over openzap Message-ID: <20090127164104.207050@gmx.net> Hello, I can't make outbound calls with openzap. I have try to make outbound calls over asterisk and it's work. So I think that the zaptel konfiguration is okay. Or I'm wrong? What's the mistake? I get this error on the console: 2009-01-27 15:32:45 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 111->017XXXX at default 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX [2b330882-1286-4b8b-bfe6-825cee3c8268] 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE [DIALING] freeswitch.log: 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:339 tech_init() Set codec PCMA 20ms 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:931 channel_outgoing_channel() Connect outbound channel OpenZAP/1:1/017XXXX 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX [2b330882-1286-4b8b-bfe6-825cee3c8268] 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:940 channel_outgoing_channel() OpenZAP/1:1/017XXXX State Change CS_NEW -> CS_INIT 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:56 isdn_outgoing_call() Changing state on 1:1 from DOWN to DIALING 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_INIT 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:363 channel_on_init() OpenZAP/1:1/017XXXX State Change CS_INIT -> CS_ROUTING 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT going to sleep 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_ROUTING 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:384 channel_on_routing() OpenZAP/1:1/017XXXX CHANNEL ROUTING 2009-01-27 15:32:45 [DEBUG] switch_ivr_originate.c:57 originate_on_routing() OpenZAP/1:1/017XXXX State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING going to sleep 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_CONSUME_MEDIA 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (OpenZAP/1:1/017XXXX) State CONSUME_MEDIA 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE [DIALING] 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:957 q931_rx_32() WRITE 55 -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a From jgarland at jasongarland.com Tue Jan 27 09:36:54 2009 From: jgarland at jasongarland.com (Jason Garland) Date: Tue, 27 Jan 2009 12:36:54 -0500 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <001301c97cd6$02107b90$063172b0$@net> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> Message-ID: <4ca506420901270936x16eafee5if7370a6a755caadd@mail.gmail.com> If you want Speex support you need to target the chipset manufacturers: Here is the Texas Insturments chipset that Polycom uses in the IP650 CPU is TNETV1050/C55x, rev 2 running at 162MHz with memory at 125MHz. And here are the codecs that Chip supports from TI's datasheet on this chip: http://focus.ti.com/pdfs/bcg/tnetv1050_prod_bulletin.pdf Codec Options* G.711 Codec, G.726, G.729AB, G.723.1A, G.722 wideband codec Speex is not listed, so Polycom can't do Speex. I should note that even the non-HD Polycom phones have this same chip and are capable of doing G.722 some some config tweaking. ;) I'm willing to bet that Cisco doesn't make their own DSP chips either. Find out who makes their chips and put pressure on that chip manufacturer to develop chips with Speex support. On Thu, Jan 22, 2009 at 4:11 PM, Gregory Boehnlein wrote: > > >> You use it on your own risk > > > > Also, G.729 is patent encumbered big-time. Instead of lining the > > pockets of lawyers and mega-corporations by perpetuating the use of a > > crusty old codec we should all twist arms and get our providers, > > device makers, etc. to use Speex. > > Yeah.. let me know when you get Cisco to add Speex support to IOS! ;) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090127/7dca0eeb/attachment.html From msc at freeswitch.org Tue Jan 27 10:57:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 27 Jan 2009 10:57:57 -0800 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <191c3a030901270715r538f45e4g615dffa639a63c20@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <497B0563.9090701@ewetel.de> <497F19E3.70900@ewetel.de> <191c3a030901270715r538f45e4g615dffa639a63c20@mail.gmail.com> Message-ID: <87f2f3b90901271057x323127cas9ace08c2573783c@mail.gmail.com> Helmut, Nice work! Thanks for doing this. -MC On Tue, Jan 27, 2009 at 7:15 AM, Anthony Minessale wrote: > Looks good, you might want to consider making the file name that is > generated be an optional argument but it's not necessary just a suggestion. > > > > On Tue, Jan 27, 2009 at 8:27 AM, Helmut Kuper > wrote: >> >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >> Hello, >> >> >> again an update to my little project: >> >> I enhanced mod_openzap and ozmod_isdn so that I'm able to start and stop >> q931ToPcap generation in ozmod_isdn from FS's console. q931ToPcap itself >> isn't implemented yet. >> >> To give you an idea about the enhanced oz command - it's like this: >> >> freeswitch at ippbx-prod-node0> oz >> API CALL [oz()] output: >> list || dump [] || bounce [] || >> q931_pcap on|off >> freeswitch at ippbx-prod-node0> oz q931_pcap >> API CALL [oz(q931_pcap)] output: >> - -ERR Usage: oz q931_pcap on|off >> >> freeswitch at ippbx-prod-node0> oz q931_pcap 0 on >> API CALL [oz(q931_pcap 0 on)] output: >> - -ERR invalid span >> >> freeswitch at ippbx-prod-node0> oz q931_pcap 1 on >> API CALL [oz(q931_pcap 1 on)] output: >> >> 2009-01-27 15:21:55 [INFO] mod_openzap.c:2172 oz_function() Starting >> Q931-to-pcap >> 2009-01-27 15:21:55 [DEBUG] ozmod_isdn.c:1712 zap_isdn_configure_span() >> Enabling Q931ToPcap >> freeswitch at ippbx-prod-node0> oz q931_pcap 1 off >> API CALL [oz(q931_pcap 1 off)] output: >> >> 2009-01-27 15:21:58 [INFO] mod_openzap.c:2178 oz_function() Stopping >> Q931-to-pcap >> 2009-01-27 15:21:58 [DEBUG] ozmod_isdn.c:1717 zap_isdn_configure_span() >> Disabling Q931ToPcap >> freeswitch at ippbx-prod-node0> >> >> >> >> As you can see, ozmod_isdn, where I will implement the code receives the >> oz command option. >> >> I appreciate your comments. >> >> >> regards >> helmut >> -----BEGIN PGP SIGNATURE----- >> Version: GnuPG v1.4.9 (MingW32) >> >> iEYEARECAAYFAkl/GeMACgkQ4tZeNddg3dzurACgszPDA573+uZjmY5TWXF9yj5F >> Vz8An2FvK0FsDLr0JQaHFDETWnLoPSkD >> =0hkL >> -----END PGP SIGNATURE----- >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Tue Jan 27 11:05:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 27 Jan 2009 11:05:57 -0800 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: <20090127164104.207050@gmx.net> References: <20090127164104.207050@gmx.net> Message-ID: <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> Would you mind giving us some more information? Please use pastebin.freeswitch.org to post your configuration files. Also, if you could review the information here: http://wiki.freeswitch.org/wiki/Reporting_Bugs it will help you gather what you need. See the section on openzap to know what information to supply. That will help us to diagnose what is happening. -MC On Tue, Jan 27, 2009 at 8:41 AM, "Franziska R?hler" wrote: > Hello, > > I can't make outbound calls with openzap. I have try to make outbound calls over asterisk and it's work. So I think that the zaptel konfiguration is okay. Or I'm wrong? What's the mistake? > > I get this error on the console: > > 2009-01-27 15:32:45 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 111->017XXXX at default > 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX [2b330882-1286-4b8b-bfe6-825cee3c8268] > 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE [DIALING] > > > freeswitch.log: > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:339 tech_init() Set codec PCMA 20ms > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:931 channel_outgoing_channel() Connect outbound channel OpenZAP/1:1/017XXXX > 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX [2b330882-1286-4b8b-bfe6-825cee3c8268] > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:940 channel_outgoing_channel() OpenZAP/1:1/017XXXX State Change CS_NEW -> CS_INIT > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] > 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:56 isdn_outgoing_call() Changing state on 1:1 from DOWN to DIALING > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_INIT > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:363 channel_on_init() OpenZAP/1:1/017XXXX State Change CS_INIT -> CS_ROUTING > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT going to sleep > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_ROUTING > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:384 channel_on_routing() OpenZAP/1:1/017XXXX CHANNEL ROUTING > 2009-01-27 15:32:45 [DEBUG] switch_ivr_originate.c:57 originate_on_routing() OpenZAP/1:1/017XXXX State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING going to sleep > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_CONSUME_MEDIA > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (OpenZAP/1:1/017XXXX) State CONSUME_MEDIA > 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE [DIALING] > 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:957 q931_rx_32() WRITE 55 > > > -- > NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL > f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From peder at networkoblivion.com Tue Jan 27 11:17:37 2009 From: peder at networkoblivion.com (peder at networkoblivion.com) Date: Tue, 27 Jan 2009 13:17:37 -0600 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <4ca506420901270936x16eafee5if7370a6a755caadd@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <4ca506420901270936x16eafee5if7370a6a755caadd@mail.gmail.com> Message-ID: <497F5DD1.8080507@networkoblivion.com> > Codec Options* > G.711 Codec, G.726, G.729AB, G.723.1A, G.722 wideband codec > > Speex is not listed, so Polycom can't do Speex. > > I should note that even the non-HD Polycom phones have this same chip > and are capable of doing G.722 some some config tweaking. ;) Are you saying that you have actually gotten the non-HD versions to do g.722? Or are you saying that one might be able to do this? If the former, can you share what you did? From msc at freeswitch.org Tue Jan 27 12:26:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 27 Jan 2009 12:26:07 -0800 Subject: [Freeswitch-users] ATA-answering machine question/recommendation In-Reply-To: <21680924.post@talk.nabble.com> References: <207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45@mail.gmail.com> <1461348542-1232566409-cardhu_decombobulator_blackberry.rim.net-569820458-@bxe161.bisx.prod.on.blackberry> <21640813.post@talk.nabble.com> <87f2f3b90901260203xd447883xe373244207c2fc45@mail.gmail.com> <21680924.post@talk.nabble.com> Message-ID: <87f2f3b90901271226n43d3c1c7lbb2edde19ead9642@mail.gmail.com> I'm not too sure about this one. I see it calling mod_vmd and complaining about needing media. Let's see what Eric Des Courtis has to say on this one. Hold tight please... -MC On Mon, Jan 26, 2009 at 11:40 PM, shehzad p wrote: > > Hi Michael, > > My dial plan is: > ======================== > > > > > data="sofia/external/1111 at xxx.xxx.xxx.x"/> > > > > > ==================================== > When call come to xxx.xxx.xxx.x system, It answer the call and only wait for > 20 seconds (NO playback only wait) and hangup. > Please find debug trace for above dialplan on > http://pastebin.freeswitch.org/6919 > > > Again I also tested dialplan exact same as shown in the wiki page > http://wiki.freeswitch.org/wiki/Mod_vmd#From_Dialplan. > But there it is not working as needed. > > > Thanks for your response... > msp > > > > Michael Collins-11 wrote: >> >> Did you ever post your dialplan and a debug trace of a call to the >> pastebin? If not, please do so and we will check it out. >> -MC >> >> On Sat, Jan 24, 2009 at 5:47 AM, shehzad p wrote: >>> >>> Hi all, >>> >>> On my existing Freeswitch 1.0.2, I installed and configured mod_vmd as >>> below: >>> make mod_vmd-install >>> >>> Then configured it as on wiki page: >>> http://wiki.freeswitch.org/wiki/Mod_vmd >>> >>> >>> After that my dialplan terminates call to another system, where it is >>> just >>> answered and wait for some time there. >>> So that there should be a variable called vmd_detect must be created as >>> shown in dialplan http://wiki.freeswitch.org/wiki/Mod_vmd#From_Dialplan. >>> >>> But eventhough no variable named 'vmd_detect' is created after that.!!! >>> Is there i am missing something? Is there another way of using mod_vmd? >>> >>> Thanks in advance. >>> msp >>> >>> >>> >>> >>> >>> >>> Lucas Cornelisse wrote: >>>> >>>> Hi Jonathan, >>>> >>>> Mod_vmd (voicemail detection) should do the trick. >>>> >>>> Just search the wiki for mod_vmd, there are a number of ways of using >>>> it. >>>> >>>> >>>> Sent from my BlackBerry device on the Rogers Wireless Network >>>> >>>> -----Original Message----- >>>> From: jonathan augenstine >>>> >>>> Date: Wed, 21 Jan 2009 06:53:41 >>>> To: >>>> Subject: [Freeswitch-users] ATA-answering machine >>>> question/recommendation >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/ATA-answering-machine-question-recommendation-tp21584795p21640813.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/ATA-answering-machine-question-recommendation-tp21584795p21680924.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From intralanman at freeswitch.org Tue Jan 27 13:29:15 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 27 Jan 2009 16:29:15 -0500 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <497F0300.8060607@devel-it.com.br> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org><4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br><87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> <497E17F5.2020400@devel-it.com.br> <497F0300.8060607@devel-it.com.br> Message-ID: <497F7CAB.4010406@freeswitch.org> Rodrigo P. Telles wrote: > Hi Abdul, > > Brazil does much more than that but we still on "latim america" and according with 1st world countries, we are > considered "non developed country". > The most funny thing about that: some people still thinks that we (brazillians) lives in the forest with monkeys :-) > so you're saying you don't live in the forest with monkeys? :-P From jgarland at jasongarland.com Tue Jan 27 13:48:52 2009 From: jgarland at jasongarland.com (Jason Garland) Date: Tue, 27 Jan 2009 16:48:52 -0500 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <497F5DD1.8080507@networkoblivion.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <4ca506420901270936x16eafee5if7370a6a755caadd@mail.gmail.com> <497F5DD1.8080507@networkoblivion.com> Message-ID: <4ca506420901271348y1f5e4076ha6f13b7ad07e3b05@mail.gmail.com> Something like this might do it... ;) On Tue, Jan 27, 2009 at 2:17 PM, peder at networkoblivion.com < peder at networkoblivion.com> wrote: > > Codec Options* > > G.711 Codec, G.726, G.729AB, G.723.1A, G.722 wideband codec > > > > Speex is not listed, so Polycom can't do Speex. > > > > I should note that even the non-HD Polycom phones have this same chip > > and are capable of doing G.722 some some config tweaking. ;) > > Are you saying that you have actually gotten the non-HD versions to do > g.722? Or are you saying that one might be able to do this? If the > former, can you share what you did? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090127/55005a4b/attachment.html From msc at freeswitch.org Tue Jan 27 17:12:06 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 27 Jan 2009 17:12:06 -0800 Subject: [Freeswitch-users] Another way to set CallerID? In-Reply-To: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> References: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> Message-ID: Did you get this resolved? Just curious. -MC Sent from my iPhone On Jan 26, 2009, at 5:05 PM, Ron McCarthy wrote: > Hi, > > I am having a weird issue with setting the callerID number for > outbound calls, I have this: > > data="effective_caller_id_number=17025551234"/> > > > Ive set the callerID correcly on a Asterisk box and the carrier sees > it and passed it correct, but for some reason any calls from > Freeswitch won't work, it either shows up as private or another > number, so they are not seeing the CID for whatever reason. This > method works for other carrirers just fine, but not them. From what > I read this sets the from in the SIP message, maybe for some reason > that is goofing it up. Any other way to set the callerID in > Freeswitch, that's the only way I have found. > > FS shows it's connecting to a Sonus, does it need to be set > different because of some Sonus issue that im not aware of? > > Any help would be great. > > Thanks! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ronmccar at gmail.com Tue Jan 27 17:32:48 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Tue, 27 Jan 2009 18:32:48 -0700 Subject: [Freeswitch-users] Another way to set CallerID? In-Reply-To: References: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> Message-ID: <3885f4fe0901271732w15c27cf9v30c50d5acb3b1534@mail.gmail.com> Nope. Tried all three ways, im stumped. I had this issue with another provider and we never figured it out and then I never used them anyways, but with new other provider I have to get it to work. Talked with the carrier, they say we are sending "user" as the CID itself. Super weird, not sure what else to try, I think it's a issue on the providers side, but no clue why, Asterisk can send CID to them no issue. Im going to dig some more and see what happens, hopefully I can figure it out! On Tue, Jan 27, 2009 at 6:12 PM, Michael S Collins wrote: > Did you get this resolved? Just curious. > -MC > > Sent from my iPhone > > On Jan 26, 2009, at 5:05 PM, Ron McCarthy wrote: > > > Hi, > > > > I am having a weird issue with setting the callerID number for > > outbound calls, I have this: > > > > > data="effective_caller_id_number=17025551234"/> > > > > > > Ive set the callerID correcly on a Asterisk box and the carrier sees > > it and passed it correct, but for some reason any calls from > > Freeswitch won't work, it either shows up as private or another > > number, so they are not seeing the CID for whatever reason. This > > method works for other carrirers just fine, but not them. From what > > I read this sets the from in the SIP message, maybe for some reason > > that is goofing it up. Any other way to set the callerID in > > Freeswitch, that's the only way I have found. > > > > FS shows it's connecting to a Sonus, does it need to be set > > different because of some Sonus issue that im not aware of? > > > > Any help would be great. > > > > Thanks! > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090127/5392f944/attachment-0001.html From brian at freeswitch.org Tue Jan 27 17:40:12 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 27 Jan 2009 19:40:12 -0600 Subject: [Freeswitch-users] Another way to set CallerID? In-Reply-To: <3885f4fe0901271732w15c27cf9v30c50d5acb3b1534@mail.gmail.com> References: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> <3885f4fe0901271732w15c27cf9v30c50d5acb3b1534@mail.gmail.com> Message-ID: <4B04766F-E7CA-49D5-91BD-0CD1EB0386FB@freeswitch.org> Collect the outgoing invite and lets look. /b On Jan 27, 2009, at 7:32 PM, Ron McCarthy wrote: > Nope. > > Tried all three ways, im stumped. > > I had this issue with another provider and we never figured it out > and then I never used them anyways, but with new other provider I > have to get it to work. > > Talked with the carrier, they say we are sending "user" as the CID > itself. Super weird, not sure what else to try, I think it's a issue > on the providers side, but no clue why, Asterisk can send CID to > them no issue. > > Im going to dig some more and see what happens, hopefully I can > figure it out! From steveu at coppice.org Tue Jan 27 17:53:00 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 28 Jan 2009 09:53:00 +0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <4ca506420901270936x16eafee5if7370a6a755caadd@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <4ca506420901270936x16eafee5if7370a6a755caadd@mail.gmail.com> Message-ID: <497FBA7C.1060002@coppice.org> Jason Garland wrote: > If you want Speex support you need to target the chipset manufacturers: > > Here is the Texas Insturments chipset that Polycom uses in the IP650 > CPU is TNETV1050/C55x, rev 2 running at 162MHz with memory at 125MHz. > And here are the codecs that Chip supports from TI's datasheet on this chip: > http://focus.ti.com/pdfs/bcg/tnetv1050_prod_bulletin.pdf > > Codec Options* > G.711 Codec, G.726, G.729AB, G.723.1A, G.722 wideband codec > Speex is not listed, so Polycom can't do Speex. > > I should note that even the non-HD Polycom phones have this same chip > and are capable of doing G.722 some some config tweaking. ;) > > I'm willing to bet that Cisco doesn't make their own DSP chips either. > Find out who makes their chips and put pressure on that chip > manufacturer to develop chips with Speex support. You are confusing two things. TI makes the TNETVxxxx range of chips. Their Telogy division makes software for VoIP platforms. The two may be used together or apart, and things may be added to the basic set of Telogy software. For example, in the Linksys SPA series you'll find the TI silicon used with Telogy software, but they also have Asterisk thrown in there to provide voicemail features. I very much doubt that Polycom uses TI's software. They've had their own VoIP software for a very long time, and I expect they still use it. Regards, Steve From msc at freeswitch.org Tue Jan 27 17:56:35 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 27 Jan 2009 17:56:35 -0800 Subject: [Freeswitch-users] is there an event for SIP MESSAGE? In-Reply-To: References: Message-ID: <60F3399A-F2EC-4F1F-90D0-ED45720F2F04@freeswitch.org> Out of curiosity which SIP messages have you been watching for on the event socket? Also, how are you connected to the event socket? Are you subscribing to all events and sifting through them to confirm that no events are being fired when SIP messages are being sent? -MC Sent from my iPhone On Jan 26, 2009, at 9:00 AM, "Cavalera Claudio Luigi" wrote: > Hello freeswitchers, > I'm experimenting with sip clients registered to fs and Instant > Messaging. > I've seen SIP Messages are properly routed by the sofia sip stack in > fs. > However it seems no event is ever generated on the event socket. > Would it be possible to make fs reporting the SIP Messaging? :-) > Best Regards, > Claudio > > > 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. > --- > --- > --- > --- > --- > --- > --- > --- > --- > --- > --- > -------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From intralanman at freeswitch.org Tue Jan 27 18:26:43 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 27 Jan 2009 21:26:43 -0500 Subject: [Freeswitch-users] Another way to set CallerID? In-Reply-To: <3885f4fe0901271732w15c27cf9v30c50d5acb3b1534@mail.gmail.com> References: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> <3885f4fe0901271732w15c27cf9v30c50d5acb3b1534@mail.gmail.com> Message-ID: <497FC263.2090305@freeswitch.org> Ron McCarthy wrote: > Nope. > > Tried all three ways, im stumped. > > I had this issue with another provider and we never figured it out and > then I never used them anyways, but with new other provider I have to > get it to work. > > Talked with the carrier, they say we are sending "user" as the CID > itself. Super weird, not sure what else to try, I think it's a issue > on the providers side, but no clue why, Asterisk can send CID to them > no issue. try setting on your gateway... the fact that it works on asterisk makes me think the carrier is using the From header for callerid. -Ray From ronmccar at gmail.com Tue Jan 27 18:38:17 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Tue, 27 Jan 2009 19:38:17 -0700 Subject: [Freeswitch-users] Another way to set CallerID? In-Reply-To: <497FC263.2090305@freeswitch.org> References: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> <3885f4fe0901271732w15c27cf9v30c50d5acb3b1534@mail.gmail.com> <497FC263.2090305@freeswitch.org> Message-ID: <3885f4fe0901271838m203ff554kaed5d34e64b38a5f@mail.gmail.com> Yeah that fixed it! I have never even seen this option in the docs before, but that sure did the trick. Thanks guys, this was driving me nuts! On Tue, Jan 27, 2009 at 7:26 PM, Raymond Chandler < intralanman at freeswitch.org> wrote: > Ron McCarthy wrote: > > Nope. > > > > Tried all three ways, im stumped. > > > > I had this issue with another provider and we never figured it out and > > then I never used them anyways, but with new other provider I have to > > get it to work. > > > > Talked with the carrier, they say we are sending "user" as the CID > > itself. Super weird, not sure what else to try, I think it's a issue > > on the providers side, but no clue why, Asterisk can send CID to them > > no issue. > try setting on your > gateway... the fact that it works on asterisk makes me think the carrier > is using the From header for callerid. > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090127/3a8f378b/attachment.html From chris.chen2004 at gmail.com Tue Jan 27 19:15:38 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Tue, 27 Jan 2009 22:15:38 -0500 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <4ca506420901271348y1f5e4076ha6f13b7ad07e3b05@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <4ca506420901270936x16eafee5if7370a6a755caadd@mail.gmail.com> <497F5DD1.8080507@networkoblivion.com> <4ca506420901271348y1f5e4076ha6f13b7ad07e3b05@mail.gmail.com> Message-ID: <507898380901271915r32f3259cn13c3576cb097ef72@mail.gmail.com> Great thanks to Jason for sharing Cherebrum's great discovery, this works like a charm on my Ploycom IP 320 with G722 codec. Chris On Tue, Jan 27, 2009 at 4:48 PM, Jason Garland wrote: > > Something like this might do it... ;) > > > > > > > > > > > > > > > > > > On Tue, Jan 27, 2009 at 2:17 PM, peder at networkoblivion.com < > peder at networkoblivion.com> wrote: > >> > Codec Options* >> > G.711 Codec, G.726, G.729AB, G.723.1A, G.722 wideband codec >> > >> > Speex is not listed, so Polycom can't do Speex. >> > >> > I should note that even the non-HD Polycom phones have this same chip >> > and are capable of doing G.722 some some config tweaking. ;) >> >> Are you saying that you have actually gotten the non-HD versions to do >> g.722? Or are you saying that one might be able to do this? If the >> former, can you share what you did? >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090127/ec98dfc6/attachment.html From klaus.teller at gmx.net Tue Jan 27 21:04:02 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Wed, 28 Jan 2009 06:04:02 +0100 Subject: [Freeswitch-users] DTMF with Early Media Disabled Message-ID: <20090128050402.75810@gmx.net> Hi, My settings does not allow me to test the following right now. So I'm wondering if somebody knowledgeable could help me answer the following question. I do know that if i call Freeswitch, i can use Javascript to read DTMF even without answering the call. My question is can i do this even if early media is disabled on the inbound call? Thanks, Klaus. -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a From brian at freeswitch.org Tue Jan 27 21:15:02 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 27 Jan 2009 23:15:02 -0600 Subject: [Freeswitch-users] DTMF with Early Media Disabled In-Reply-To: <20090128050402.75810@gmx.net> References: <20090128050402.75810@gmx.net> Message-ID: If the dtmf is in the media stream ie 2833 and you can't establish media then no you wouldn't. Have you tried to do a pre_answer instead of an answer to establish early media? /b On Jan 27, 2009, at 11:04 PM, Klaus Teller wrote: > Hi, > > My settings does not allow me to test the following right now. So > I'm wondering if somebody knowledgeable could help me answer the > following question. > > I do know that if i call Freeswitch, i can use Javascript to read > DTMF even without answering the call. My question is can i do this > even if early media is disabled on the inbound call? > > Thanks, > > Klaus. > -- From krivushinme at rn-inform.tomsk.ru Tue Jan 27 22:53:58 2009 From: krivushinme at rn-inform.tomsk.ru (=?utf-8?b?0JrRgNC40LLRg9GI0LjQvSDQnNC40YXQsNC40Ls=?=) Date: Wed, 28 Jan 2009 12:53:58 +0600 Subject: [Freeswitch-users] Another way to set CallerID? In-Reply-To: <3885f4fe0901271838m203ff554kaed5d34e64b38a5f@mail.gmail.com> References: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> <497FC263.2090305@freeswitch.org> <3885f4fe0901271838m203ff554kaed5d34e64b38a5f@mail.gmail.com> Message-ID: <200901281253.59018.krivushinme@rn-inform.tomsk.ru> On Wednesday 28 January 2009 08:38:17 Ron McCarthy wrote: > Yeah that fixed it! > > I have never even seen this option in the docs before, but that sure did > the trick. Please, add this to wiki. -- ? ?????????, ???????? ?????? ??????? ?????????? ?????? ????????????????, ??? "??-??????" ?????? ? ?.??????, ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru mail: KrivushinME at rn-inform.tomsk.ru From pmhshz at gmail.com Wed Jan 28 00:54:13 2009 From: pmhshz at gmail.com (shehzad p) Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic Message-ID: <21701744.post@talk.nabble.com> Hi all, Yesterday my Freeswitch server faced a problem when call traffic increased to more than 100. When I start Freeswitch, it works fine and then after some time (approximately 15 to 20 minutes) it stops functioning (means no call is being processed, no CLI command is working and it just freezes) until I restart the freeswitch. I am using Freeswitch 1.0.1. Debug (gdb) trace as on wiki page http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.sh is attached http://www.nabble.com/file/p21701744/fs_debgu.txt fs_debgu.txt -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21701744.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From krice at suspicious.org Wed Jan 28 01:11:16 2009 From: krice at suspicious.org (Ken Rice) Date: Wed, 28 Jan 2009 03:11:16 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21701744.post@talk.nabble.com> Message-ID: Upgrade to trunk... Many many issues have been resolved since 1.0.1 was the current release > From: shehzad p > Reply-To: > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) > To: > Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic > > > Hi all, > > Yesterday my Freeswitch server faced a problem when call traffic increased > to more than 100. > > When I start Freeswitch, it works fine and then after some time > (approximately 15 to 20 minutes) it stops functioning (means no call is > being processed, no CLI command is working and it just freezes) until I > restart the freeswitch. > > I am using Freeswitch 1.0.1. > Debug (gdb) trace as on wiki page > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.sh is attached > http://www.nabble.com/file/p21701744/fs_debgu.txt fs_debgu.txt > -- > View this message in context: > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 > p21701744.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Wed Jan 28 01:55:11 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 28 Jan 2009 10:55:11 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <87f2f3b90901271057x323127cas9ace08c2573783c@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <497B0563.9090701@ewetel.de> <497F19E3.70900@ewetel.de> <191c3a030901270715r538f45e4g615dffa639a63c20@mail.gmail.com> <87f2f3b90901271057x323127cas9ace08c2573783c@mail.gmail.com> Message-ID: <49802B7F.70100@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Guys, today I tried to decode a q931 pcap file directly in a linux cli. I had success and the result looks like this: - -- SENDING ------- Packet number: 00001 --- SpanID: 1 ---------------- Protocol discriminator: Q.931 Call reference value length: 2 Call reference flag: Message sent from originating side Call reference value: 0002 Message type: RELEASE (0x4d) Display 'HK at FreeSWITCH' Information element: Display Length: 13 Display information: HK at FreeSWITCH - -- SENDING ------- Packet number: 00002 --- SpanID: 1 ---------------- Protocol discriminator: Q.931 Call reference value length: 2 Call reference flag: Message sent to originating side Call reference value: 0002 Message type: DISCONNECT (0x45) Cause Information element: Cause Length: 2 .... 0000 = Cause location: User (U) (0) .00. .... = Coding standard: ITU-T standardized coding (0x00) 1... .... = Extension indicator: last octet .001 0000 = Cause value: Normal call clearing (16) 1... .... = Extension indicator: last octet I think this helps server admins to get a much faster access to the decoded packages on a pure server where they have no GUI and hence no wireshark. tshark is needed for this of course. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkmAK38ACgkQ4tZeNddg3dxAEACgsf+GC3jqTvBUYD2pqsgtZgUs s8QAoLIitPAc0I55zKXyw6yTe4MDaDaK =/Ir7 -----END PGP SIGNATURE----- From Claudio.Cavalera at italtel.it Wed Jan 28 02:24:00 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 28 Jan 2009 11:24:00 +0100 Subject: [Freeswitch-users] is there an event for SIP MESSAGE? In-Reply-To: <60F3399A-F2EC-4F1F-90D0-ED45720F2F04@freeswitch.org> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > Out of curiosity which SIP messages have you been watching for on the > event socket? Also, how are you connected to the event socket? Are you > subscribing to all events and sifting through them to confirm that no > events are being fired when SIP messages are being sent? -MC Thanks for your interest Michael, I'll try to explain better since I think this could become a useful feature; fs already provides a lot of features for Unified Communication. Moreover SIP messages are supported almost for free from the sofia sip stack. When experimenting I usually connect to the event socket with netcat: nc localhost 8021 auth ClueCon event plain all If i register two clients to the event socket they can SIP message each other. I'm meaning SIP MESSAGES like this one sent from user 1000 to 1001: http://pastebin.freeswitch.org/6940 I really would like an event to be fired by this SIP MESSAGING in fact I'm almost adding it at mod_sofia.h as #define MY_EVENT_SIP_MESSAGE "sofia::message" :-) Firing an event for every SIP MESSAGE sent through fs would be a great thing, but sadly not sufficient for building a full IM solution. In fact we would also need an API to send SIP MESSAGES from within fs and this is already partially achieved from what I've seen with command like this one sent to the event socket: sendevent SEND_MESSAGE profile: internal content-length: 2 content-type: application/simple-message-summary user: 1001 host: 192.168.1.1 Hi Content-Type: command/reply Reply-Text: +OK This command works and fires an event like this: Command: sendevent%20SEND_MESSAGE profile: internal content-length: 2 content-type: application/simple-message-summary user: 1001 host: 192.168.1.1 Event-Name: SEND_MESSAGE Core-UUID: a7f40fea-8340-4d77-aded-f322f9bee016 FreeSWITCH-Hostname: lallobox FreeSWITCH-IPv4: 192.168.1.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-01-28%2010%3A44%3A07 Event-Date-GMT: Wed,%2028%20Jan%202009%2009%3A44%3A07%20GMT Event-Date-Timestamp: 1233135847314907 Event-Calling-File: mod_event_socket.c Event-Calling-Function: parse_command Event-Calling-Line-Number: 1511 Content-Length: 2 Hi However it would be wonderful to enhance this API with the possibility to select the sending user thus being able to forge sip messages from within fs on behalf of a registered or not user. Best Regards, 2C PS: Michael feel free to contact me off list as you see fit. 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. ----------------------------------------------------------------------------------------------------- From brian at freeswitch.org Wed Jan 28 02:41:54 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 28 Jan 2009 04:41:54 -0600 Subject: [Freeswitch-users] Another way to set CallerID? In-Reply-To: <200901281253.59018.krivushinme@rn-inform.tomsk.ru> References: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> <497FC263.2090305@freeswitch.org> <3885f4fe0901271838m203ff554kaed5d34e64b38a5f@mail.gmail.com> <200901281253.59018.krivushinme@rn-inform.tomsk.ru> Message-ID: The Wiki is a community resource. The best part is anyone can add/ update info on the wiki if they register for an account, its also a good way to give back to the community. http://wiki.freeswitch.org/wiki/Sofia.conf.xml#Gateway This is however on the wiki and maybe it just needs improvement. ;) /b On Jan 28, 2009, at 12:53 AM, ???????? ?????? wrote: > On Wednesday 28 January 2009 08:38:17 Ron McCarthy wrote: >> Yeah that fixed it! >> >> I have never even seen this option in the docs before, but that >> sure did >> the trick. > Please, add this to wiki. From pmhshz at gmail.com Wed Jan 28 03:43:24 2009 From: pmhshz at gmail.com (shehzad p) Date: Wed, 28 Jan 2009 03:43:24 -0800 (PST) Subject: [Freeswitch-users] Method getVariable cause error on FS 1.0.2 in javascript In-Reply-To: <21578116.post@talk.nabble.com> References: <21578116.post@talk.nabble.com> Message-ID: <21704201.post@talk.nabble.com> Hi again, How can i resolve this error, Does it require to modify my js code, Or any freeswitch settings (dialplan or profile related etc.) can solve this? I am waiting for your response. Thanks in advance... -- View this message in context: http://www.nabble.com/Method-getVariable-cause-error-on-FS-1.0.2-in-javascript-tp21578116p21704201.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Wed Jan 28 03:54:34 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 28 Jan 2009 05:54:34 -0600 Subject: [Freeswitch-users] Method getVariable cause error on FS 1.0.2 in javascript In-Reply-To: <21578116.post@talk.nabble.com> References: <21578116.post@talk.nabble.com> Message-ID: You really shouldn't be using the originate method. You're doing more by hand then you should. Care to post a little more detail about how you're using this? /b On Jan 21, 2009, at 1:10 AM, shehzad p wrote: > newsession.originate(session, dialstr); From steveu at coppice.org Wed Jan 28 04:12:13 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 28 Jan 2009 20:12:13 +0800 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <1233065084.8161.48.camel@stargate> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <2d9149cd0901221355o4d0532e4qe5058741eb7c8fd5@mail.gmail.com> <2d9149cd0901221513h1a230e35g2739ccaec120a609@mail.gmail.com> <31F9B257-930F-4113-B738-8A725F9F0279@freeswitch.org> <2d9149cd0901221548g170876cax1e4747a04a94069d@mail.gmail.com> <49790F8E.8000505@coppice.org><4979510C.90303@3c.co.uk> <497A25B8.3010504@devel-it.com.br> <87f2f3b90901231229v7fe6e349k5ee358af506b1d06@mail.gmail.com> <497E17F5.2020400@devel-it.com.br> <497E7574.2070907@coppice.org> <1233037964.8161.30.camel@stargate> <497EB686.9080504@coppice.org> <1233065084.8161.48.camel@stargate> Message-ID: <49804B9D.2020600@coppice.org> Hi Raul, Raul Fragoso wrote: > Hi Steve, > > My point is more towards the high import taxes here. A foreign company > won't pay as much taxes to export to Brazil as brazilians would pay > import taxes for such goods. > I will give you a simple example: a Snom 320 phone would cost roughly US > $180.00 if bought in the US. Here it would cost at least double the > price, but usually more. We pay almost 100% for importing taxes + GST, > excluding the greedy profit that dealers and resellers usually apply, > but Snom (or their distributors) would pay only a small fraction of that > to export phones to Brazil. The irony is that we don't see where all > this money is spent, as the country still suffers of a huge social and > economical dissimilarity among their people. > On the other hand, our current government seems to be making some > progress to make exporting from Brazil easier, specially for small > companies (international trade is known to be very bureaucratic here). > Anyway, this is turning into a politics discussion thread and I know it > can annoy some people. Just to stay in the subject thread, I also > endorse the support for open codecs rather than paying millions for > G.729 licenses. > Well, any discussion of trade immediately has a political element. However, this discussion is quite interesting from an engineering point of view. Engineering is never separate from politics, as so much engineering is working around artificial barriers erected by politics. To many engineers, a lot of decisions seem wacky, because they can't see the political angle. The different pressures guiding decisions like buy local/buy foreign, use ISDN/use MFC/R2, and so on can be quite illuminating. Steve From curly2009 at gmx.de Wed Jan 28 05:24:38 2009 From: curly2009 at gmx.de (curly2009 at gmx.de) Date: Wed, 28 Jan 2009 14:24:38 +0100 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> References: <20090127164104.207050@gmx.net> <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> Message-ID: <20090128132438.309550@gmx.net> I posted the configuration files in pastebin.freeswitch.org > Would you mind giving us some more information? Please use > pastebin.freeswitch.org to post your configuration files. Also, if you > could review the information here: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > it will help you gather what you need. See the section on openzap to > know what information to supply. That will help us to diagnose what is > happening. > -MC > > On Tue, Jan 27, 2009 at 8:41 AM, > wrote: > > Hello, > > > > I can't make outbound calls with openzap. I have try to make outbound > calls over asterisk and it's work. So I think that the zaptel konfiguration > is okay. Or I'm wrong? What's the mistake? > > > > I get this error on the console: > > > > 2009-01-27 15:32:45 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() > Processing 111->017XXXX at default > > 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 > switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX [2b330882-1286-4b8b-bfe6-825cee3c8268] > > 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE > [DIALING] > > > > > > freeswitch.log: > > > > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:339 tech_init() Set codec PCMA > 20ms > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:931 channel_outgoing_channel() > Connect outbound channel OpenZAP/1:1/017XXXX > > 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 > switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX [2b330882-1286-4b8b-bfe6-825cee3c8268] > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:940 channel_outgoing_channel() > OpenZAP/1:1/017XXXX State Change CS_NEW -> CS_INIT > > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 > switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] > > 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:56 isdn_outgoing_call() Changing > state on 1:1 from DOWN to DIALING > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 > switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_INIT > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 > switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:363 channel_on_init() > OpenZAP/1:1/017XXXX State Change CS_INIT -> CS_ROUTING > > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 > switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 > switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT going to sleep > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 > switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_ROUTING > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 > switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:384 channel_on_routing() > OpenZAP/1:1/017XXXX CHANNEL ROUTING > > 2009-01-27 15:32:45 [DEBUG] switch_ivr_originate.c:57 > originate_on_routing() OpenZAP/1:1/017XXXX State Change CS_ROUTING -> CS_CONSUME_MEDIA > > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 > switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 > switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING going to sleep > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 > switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_CONSUME_MEDIA > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:442 > switch_core_session_run() (OpenZAP/1:1/017XXXX) State CONSUME_MEDIA > > 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE > [DIALING] > > 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:957 q931_rx_32() WRITE 55 > > > > > > -- > > NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL > > f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger From anthony.minessale at gmail.com Wed Jan 28 05:54:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 28 Jan 2009 07:54:12 -0600 Subject: [Freeswitch-users] Method getVariable cause error on FS 1.0.2 in javascript In-Reply-To: References: <21578116.post@talk.nabble.com> Message-ID: <191c3a030901280554m6e6b5ce9yb4ac56b5fd2be608@mail.gmail.com> You *never* could get vars from channels that did not successfully originate. Unless originate was a success there is nothing to get variables from. What you probably did was get the variable from the existing session when a failure occurs because pdd time from a failed call is copied across to the A leg. So try session.getVariable not newsession.getVariable On Wed, Jan 28, 2009 at 5:54 AM, Brian West wrote: > You really shouldn't be using the originate method. You're doing more > by hand then you should. Care to post a little more detail about how > you're using this? > > /b > > On Jan 21, 2009, at 1:10 AM, shehzad p wrote: > > > newsession.originate(session, dialstr); > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/3cf8c2b8/attachment.html From klaus.teller at gmx.net Wed Jan 28 05:55:25 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Wed, 28 Jan 2009 14:55:25 +0100 Subject: [Freeswitch-users] DTMF with Early Media Disabled In-Reply-To: References: <20090128050402.75810@gmx.net> Message-ID: <20090128135525.153200@gmx.net> I know it works perfectly when pre_answer is called. That is, when early media is activated. I was just trying to figure out what is the expected behavior when pre_answer is not called. I want to get DTMF from users without having them billed by their carriers. I've heard that some carriers start billing as soon as early media is on. That's why i was wondering if DTMF (inband or out of band) can be received without answering or pre_answering. Thanks, Klaus. -------- Original-Nachricht -------- > Datum: Tue, 27 Jan 2009 23:15:02 -0600 > Von: Brian West > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] DTMF with Early Media Disabled > If the dtmf is in the media stream ie 2833 and you can't establish > media then no you wouldn't. Have you tried to do a pre_answer > instead of an answer to establish early media? > > /b > > On Jan 27, 2009, at 11:04 PM, Klaus Teller wrote: > > > Hi, > > > > My settings does not allow me to test the following right now. So > > I'm wondering if somebody knowledgeable could help me answer the > > following question. > > > > I do know that if i call Freeswitch, i can use Javascript to read > > DTMF even without answering the call. My question is can i do this > > even if early media is disabled on the inbound call? > > > > Thanks, > > > > Klaus. > > -- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a From anthony.minessale at gmail.com Wed Jan 28 06:01:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 28 Jan 2009 08:01:36 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: References: <21701744.post@talk.nabble.com> Message-ID: <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> Also remember, Actually completely uninstall and erase /usr/local/freeswitch and the 1.0.1 source tree and freshly install the new one. If you try to upgrade on top of a release with trunk it will cause more problems for you. On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice wrote: > Upgrade to trunk... Many many issues have been resolved since 1.0.1 was the > current release > > > > From: shehzad p > > Reply-To: > > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) > > To: > > Subject: [Freeswitch-users] Freeswitch freezes on increasing call > traffic > > > > > > Hi all, > > > > Yesterday my Freeswitch server faced a problem when call traffic > increased > > to more than 100. > > > > When I start Freeswitch, it works fine and then after some time > > (approximately 15 to 20 minutes) it stops functioning (means no call is > > being processed, no CLI command is working and it just freezes) until I > > restart the freeswitch. > > > > I am using Freeswitch 1.0.1. > > Debug (gdb) trace as on wiki page > > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.sh is > attached > > http://www.nabble.com/file/p21701744/fs_debgu.txt fs_debgu.txt > > -- > > View this message in context: > > > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 > > p21701744.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/ce10e0d5/attachment.html From anthony.minessale at gmail.com Wed Jan 28 06:03:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 28 Jan 2009 08:03:00 -0600 Subject: [Freeswitch-users] DTMF with Early Media Disabled In-Reply-To: <20090128135525.153200@gmx.net> References: <20090128050402.75810@gmx.net> <20090128135525.153200@gmx.net> Message-ID: <191c3a030901280603p730b70f4yed7cab59f8638ec@mail.gmail.com> sorry, no, you can't do that. On Wed, Jan 28, 2009 at 7:55 AM, Klaus Teller wrote: > I know it works perfectly when pre_answer is called. That is, when early > media is activated. I was just trying to figure out what is the expected > behavior when pre_answer is not called. > > I want to get DTMF from users without having them billed by their carriers. > I've heard that some carriers start billing as soon as early media is on. > That's why i was wondering if DTMF (inband or out of band) can be received > without answering or pre_answering. > > Thanks, > Klaus. > -------- Original-Nachricht -------- > > Datum: Tue, 27 Jan 2009 23:15:02 -0600 > > Von: Brian West > > An: freeswitch-users at lists.freeswitch.org > > Betreff: Re: [Freeswitch-users] DTMF with Early Media Disabled > > > If the dtmf is in the media stream ie 2833 and you can't establish > > media then no you wouldn't. Have you tried to do a pre_answer > > instead of an answer to establish early media? > > > > /b > > > > On Jan 27, 2009, at 11:04 PM, Klaus Teller wrote: > > > > > Hi, > > > > > > My settings does not allow me to test the following right now. So > > > I'm wondering if somebody knowledgeable could help me answer the > > > following question. > > > > > > I do know that if i call Freeswitch, i can use Javascript to read > > > DTMF even without answering the call. My question is can i do this > > > even if early media is disabled on the inbound call? > > > > > > Thanks, > > > > > > Klaus. > > > -- > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL > f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/bba6e13f/attachment-0001.html From anthony.minessale at gmail.com Wed Jan 28 06:07:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 28 Jan 2009 08:07:36 -0600 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: <20090128132438.309550@gmx.net> References: <20090127164104.207050@gmx.net> <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> <20090128132438.309550@gmx.net> Message-ID: <191c3a030901280607r706b4a8cp4cc586fc62cb5086@mail.gmail.com> Is this analog FXO? Analog FXO needs to detect a dial tone to make an outbound call. Do you have the tones configured properly for your location? Are you using the latest revision of the CODE? You should not paste only small snippets of log, it's better to paste the whole log of the call start to finish. On Wed, Jan 28, 2009 at 7:24 AM, wrote: > I posted the configuration files in pastebin.freeswitch.org > > > > Would you mind giving us some more information? Please use > > pastebin.freeswitch.org to post your configuration files. Also, if you > > could review the information here: > > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > > > it will help you gather what you need. See the section on openzap to > > know what information to supply. That will help us to diagnose what is > > happening. > > -MC > > > > On Tue, Jan 27, 2009 at 8:41 AM, > > wrote: > > > Hello, > > > > > > I can't make outbound calls with openzap. I have try to make outbound > > calls over asterisk and it's work. So I think that the zaptel > konfiguration > > is okay. Or I'm wrong? What's the mistake? > > > > > > I get this error on the console: > > > > > > 2009-01-27 15:32:45 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() > > Processing 111->017XXXX at default > > > 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 > > switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX > [2b330882-1286-4b8b-bfe6-825cee3c8268] > > > 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE > > [DIALING] > > > > > > > > > freeswitch.log: > > > > > > > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:339 tech_init() Set codec > PCMA > > 20ms > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:931 > channel_outgoing_channel() > > Connect outbound channel OpenZAP/1:1/017XXXX > > > 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 > > switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX > [2b330882-1286-4b8b-bfe6-825cee3c8268] > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:940 > channel_outgoing_channel() > > OpenZAP/1:1/017XXXX State Change CS_NEW -> CS_INIT > > > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 > > switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX > [BREAK] > > > 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:56 isdn_outgoing_call() Changing > > state on 1:1 from DOWN to DIALING > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 > > switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change > CS_INIT > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 > > switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:363 channel_on_init() > > OpenZAP/1:1/017XXXX State Change CS_INIT -> CS_ROUTING > > > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 > > switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX > [BREAK] > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 > > switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT going to sleep > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 > > switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change > CS_ROUTING > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 > > switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:384 channel_on_routing() > > OpenZAP/1:1/017XXXX CHANNEL ROUTING > > > 2009-01-27 15:32:45 [DEBUG] switch_ivr_originate.c:57 > > originate_on_routing() OpenZAP/1:1/017XXXX State Change CS_ROUTING -> > CS_CONSUME_MEDIA > > > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 > > switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX > [BREAK] > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 > > switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING going to > sleep > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 > > switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change > CS_CONSUME_MEDIA > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:442 > > switch_core_session_run() (OpenZAP/1:1/017XXXX) State CONSUME_MEDIA > > > 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE > > [DIALING] > > > 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:957 q931_rx_32() WRITE 55 > > > > > > > > > -- > > > NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL > > > f?r nur 16,37 EURO/mtl.!* > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: > http://www.gmx.net/de/go/multimessenger > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/c86286c3/attachment.html From sshaw at interwise.com Wed Jan 28 06:52:20 2009 From: sshaw at interwise.com (Simon Shaw) Date: Wed, 28 Jan 2009 16:52:20 +0200 Subject: [Freeswitch-users] Sending SIP calls via outbound proxy In-Reply-To: <87f2f3b90812111259ob0a78bbw747afdc52251c2cb@mail.gmail.com> References: <49417D96.1090805@junctionnetworks.com> <87f2f3b90812111259ob0a78bbw747afdc52251c2cb@mail.gmail.com> Message-ID: <2E3152CB56588D4A91D4F1746A7252FF4D55EE@isr-brass.interwise.com> Is there a mechanism to configure FS to work with an edge proxy? What I am attempting to achieve is a front end proxy that all the clients connect to which simply forwards all messages to FS. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, December 11, 2008 11:00 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending SIP calls via outbound proxy On Thu, Dec 11, 2008 at 12:52 PM, Erick Johnson wrote: > Thanks Dave, > > Actually I realized my problem (stupid mistake of course). For anyone else > trying to use the fs_path variable the value needs to be a fully > qualified SIP > URI, e.g. "bob at bar.com;fs_path=sip:host.domain.net", notice it being > prefaced > with the "sip:", my problem was that I was only entering > the host name. Then somewhere down in mod_sofia it must have decided that > it didn't like that and just closed the channel. Erick, thanks for the clarification! I'll get it put on the wiki right away. -MC > > Hope this helps somebody who gets stuck like I did. > > Cheers, > > Erick > >> Hi Erick, >> >> Not sure if you've tried this (or if it'll help), but you can force >> routing in the dialplan like so: >> >> >> >> Cheers -- >> >> Dave > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From julianokyap at gmail.com Sun Jan 25 22:53:52 2009 From: julianokyap at gmail.com (Julian Yap) Date: Sun, 25 Jan 2009 20:53:52 -1000 Subject: [Freeswitch-users] Pylons example on the wiki In-Reply-To: <20090123233621.283650@gmx.net> References: <20090123233621.283650@gmx.net> Message-ID: This is the page that Phil created: http://wiki.freeswitch.org/wiki/Pylons On Fri, Jan 23, 2009 at 1:36 PM, wrote: > Hello, > > I just want to say that I have uploaded a Pylons wiki page with examples in svn. Pylons is a python web framework, which I think can help people to get started easily with xml_curl. > If there is demand to extend the examples I am happy to do at. Suggestions are always welcome. You can normally find me on irc: phm_it > > Cheers, > Phil > -- > NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL > f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From blansky at interwise.com Wed Jan 28 05:41:21 2009 From: blansky at interwise.com (Boris Lansky) Date: Wed, 28 Jan 2009 15:41:21 +0200 Subject: [Freeswitch-users] Outbound Proxy configuration Message-ID: What way should I configure Freeswitch to get out all the outbound calls through a specific proxy server? Regards, Boris Lansky Unified Communications Telephony Team AT&T Unified Communications Phone: +972.3.976.7604 Fax: +972.3.976.7712 blansky at interwise.com -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 16211 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/67128a62/attachment-0001.png From anthony.minessale at gmail.com Wed Jan 28 07:50:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 28 Jan 2009 09:50:15 -0600 Subject: [Freeswitch-users] mod_radius_cdr questions and thoughts In-Reply-To: <497F1B19.6080500@kinetix.gr> References: <4978A2B2.2020905@kinetix.gr> <6569777d0901220909q99e80abvb4c6253afb65dc81@mail.gmail.com> <4978C200.7030403@kinetix.gr> <497F06A8.8070507@kinetix.gr> <191c3a030901270614i23a41bc2o1fbeaf576ab3811c@mail.gmail.com> <497F1B19.6080500@kinetix.gr> Message-ID: <191c3a030901280750q404bb896t2b91fe4b4f8ff122@mail.gmail.com> We have a saying here in open source..... Patches Welcome! On Tue, Jan 27, 2009 at 8:32 AM, Apostolos Pantsiopoulos wrote: > Please see in line comments. > > Anthony Minessale wrote: > > Sounds like having cake and eating it too. > > The risk is obvious when using radius or some other additional protocol for > AAA that you will have trouble if the server is down. > Radius was designed to be fast and redundant, so typically you would have 5 > radius servers not one. > > I did not make mod_radius_cdr but I do pose this question: > > Where should the burden lie to make sure the calls are not delayed when > choosing this option? > > Within the module. A separate thread could deal with the Acct Packet > transmission without > blocking the flow of the call. After a number of retries the thread should > quit trying. > > > If you make FS cache all the radius requests it could not complete in a > timely manner, the whole point of keeping track of the exact time they > occurred is lost and you could have calls that ended before the start packet > ever was transmitted because they are cached in some process that will begin > to swell with memory remembering all the requests it could not send. > > The acct start and stop packets contain the time info needed for billing > purposes Even if the radius packets reaches the server 10 secs after > the end of the call there is little harm done. The "real" start and stop > times don't have to correspond to the times the packets arrived at the > radius server. > > > Then somehow it needs to gracefully catch up again when the radius server > comes back.... This is the same reason I think that direct database CDR is a > bad idea. > > If the retries*timeout time has passed the NAS should give up trying to > send the packet. So there is not much catching up to do. The radius packets > that failed while the radius server was down don't have to be retransmitted > later. All the applications and users that use radius are comfortable with > that > fact. I for one use a x-checking mechanism (comparing CDRs with radius > created CDRs) to verify the integrity of my calls. > > > The real answer is that you are not allowed to have your radius server down > at all, so you need more than one. I used to be in the dialup business and > we had to have backup radius servers for the backup radius servers on a > completely different network just in case not only the server was down but > the network link to the server and it's backup server. It's like DNS, I > can't ever be down or nothing works. > > I always use multiple radius servers (using different routes to my NASes). > But sometimes there are other issues that could interfere with the > NAS-Radius connectivity. > > > > > On Tue, Jan 27, 2009 at 7:05 AM, Apostolos Pantsiopoulos wrote: > >> Hi, >> >> After some testing I came to the following conclusions : >> >> 1) The problem (timeouts and retries) I describe below only happens when >> there is no radius server responding on the other side. >> >> 2) It only happens when using the latest cvs version of radiusclient. If >> you use version 1.1.6 it works fine. >> >> I also read in the wiki (and found out myself by testing) that : >> >> "Currently, the module blocks the thread while it is sending the requests. >> This may cause threads to hang around longer than expected after a call, if >> your RADIUS servers are not reachable/responding." >> >> which I think is not desirable. Was this kind of behavior, been followed >> intentionally? >> I think that the NAS in most (if not all) implementations uses a >> non-blocking operation >> in order to proceed with the call. In that way there is not any >> significant delay (up to 15 seconds if radius is down) >> in the beginning of the call. >> Also, I noticed that if the radius acct packet fails, FS does not proceed >> with the call >> which is again -in my opinion - wrong. I think that the NAS should be able >> to continue with >> the call even if the Acct start or stop failed. >> >> For those directly involved in the maintenance of the mod_radius_cdr code >> : >> >> Is it relatively easy to change the blocking behavior of the module? >> >> Apostolos Pantsiopoulos wrote: >> >> Chris Parker wrote: >> >> On Thu, Jan 22, 2009 at 10:45 AM, Apostolos Pantsiopoulos < >> regs at kinetix.gr> wrote: >> >>> I am trying to implement a radius based solution >>> using FS. I have seen that the mod_radius_cdr module >>> is actively maintained. so I have a few questions/remarks : >>> >>> 1) When I place a call and my radius server is down, the >>> call blocks forever instead of just radius_timeout * radius_retries >>> seconds (I have declared only one server). I would expect that >>> FS would stop trying to send an Acc Start packet after some >>> time and get on with the call. >> >> >> I have not seen this behavior. If you can duplicate this, and propose a >> patch, it would be gladly welcomed. >> >> I rebuilt and retried and the behavior persists. >> >> The call progress freezes and I get the following in the log : >> >> 2009-01-22 20:48:32 [DEBUG] switch_core_state_machine.c:435 >> switch_core_session_run() (sofia/internal/9333 at xxx.xxx.xxx.xxx) State >> ROUTING >> 2009-01-22 20:48:32 [DEBUG] mod_sofia.c:130 sofia_on_routing() >> sofia/internal/9333 at xx.xxx.xxx.xxx SOFIA ROUTING >> 2009-01-22 20:48:32 [DEBUG] mod_radius_cdr.c:152 my_on_routing() >> [mod_radius_cdr] Entering my_on_routing >> >> After I hangup the client and issue a shutdown in FS I get the following >> : >> >> 2009-01-22 20:50:50 [CRIT] sofia.c:794 sofia_profile_thread_run() Waiting >> for 1 session(s) >> >> repeatedly and FS never exits. >> >> >>> >>> 2) I have also noticed that FS sends only 1 packet (I waited for a >>> minute) >>> instead of 3 (default in the config) since the first (and second) >>> attempt failed. >>> If my server was up (the port was responding) but it returned a req. >>> failed >>> answer would the above time-out be valid? >> >> >> I have not seen this behavior. >> >> The same here after the rebuild. >> >> >>> >>> 3) When I tried to load the dictionary.freeswitch to my freeradius >>> server, it complained : >> >> >> Don't do that. The dictionary is for use with the radiusclient library. >> FreeRADIUS already includes a dictionary for FreeSWITCH VSAs ( you may need >> to uncomment it to have it loaded into FreeRADIUS ). >> >> I cannot find any reference to Freeswitch in the freeradius integrated >> dictionaries (in the share folder). Can you pinpoint the >> directory that a dictionary.freeswitch (or other FS related dictionary) >> resides? >> >> >> >>> 4) The radius attributes included in the current requests are >>> a) hard-coded, b) limited in number. I think many of us would like to >>> use more attributes. Or even better define what to include (and what to >>> put in them) using a >>> config file (the same maybe?) >> >> >> This has been proposed. There isn't yet a mechanism, though the intent is >> to use a general purpose FS VSA for this. The code needs to be added to the >> mod_radius_cdr module to allow that to be a run_time configuration option. >> >> A general purpose VSA that holds only one value or many? Or a mix (array >> like)? >> >> >> >>> 5) Does the module send accounting packets only for the a-leg >>> of a call or for both legs? (Maybe that could be configurable too). >>> >>> If anyone is interested in the above questions/remarks please post >>> a reply. I would really like to know how many of the mailing list users >>> are also interested in FS radius support and your opinions on the matter. >>> >> >> Again, patches are welcome. :) >> >> >> -Chris >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/77e49971/attachment.html From curly2009 at gmx.de Wed Jan 28 08:01:02 2009 From: curly2009 at gmx.de (curly2009 at gmx.de) Date: Wed, 28 Jan 2009 17:01:02 +0100 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: <191c3a030901280607r706b4a8cp4cc586fc62cb5086@mail.gmail.com> References: <20090127164104.207050@gmx.net> <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> <20090128132438.309550@gmx.net> <191c3a030901280607r706b4a8cp4cc586fc62cb5086@mail.gmail.com> Message-ID: <20090128160102.126250@gmx.net> It's ISDN PRI. Digium Wildcard TE110P I don't edit tones.conf. I leave this to the defaults. I using freeswitch-1.0.1 zaptel-1.4.12 I posted the whole log of the call in pastebin.freeswitch.org. > Is this analog FXO? > > Analog FXO needs to detect a dial tone to make an outbound call. > Do you have the tones configured properly for your location? > > Are you using the latest revision of the CODE? > > You should not paste only small snippets of log, it's better to paste the > whole log of the call start to finish. > > > > On Wed, Jan 28, 2009 at 7:24 AM, wrote: > > > I posted the configuration files in pastebin.freeswitch.org > > > > > > > Would you mind giving us some more information? Please use > > > pastebin.freeswitch.org to post your configuration files. Also, if you > > > could review the information here: > > > > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > > > > > it will help you gather what you need. See the section on openzap to > > > know what information to supply. That will help us to diagnose what is > > > happening. > > > -MC > > > > > > On Tue, Jan 27, 2009 at 8:41 AM, > > > wrote: > > > > Hello, > > > > > > > > I can't make outbound calls with openzap. I have try to make > outbound > > > calls over asterisk and it's work. So I think that the zaptel > > konfiguration > > > is okay. Or I'm wrong? What's the mistake? > > > > > > > > I get this error on the console: > > > > > > > > 2009-01-27 15:32:45 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() > > > Processing 111->017XXXX at default > > > > 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 > > > switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX > > [2b330882-1286-4b8b-bfe6-825cee3c8268] > > > > 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE > > > [DIALING] > > > > > > > > > > > > freeswitch.log: > > > > > > > > > > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:339 tech_init() Set codec > > PCMA > > > 20ms > > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:931 > > channel_outgoing_channel() > > > Connect outbound channel OpenZAP/1:1/017XXXX > > > > 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 > > > switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX > > [2b330882-1286-4b8b-bfe6-825cee3c8268] > > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:940 > > channel_outgoing_channel() > > > OpenZAP/1:1/017XXXX State Change CS_NEW -> CS_INIT > > > > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 > > > switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX > > [BREAK] > > > > 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:56 isdn_outgoing_call() > Changing > > > state on 1:1 from DOWN to DIALING > > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 > > > switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change > > CS_INIT > > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 > > > switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT > > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:363 channel_on_init() > > > OpenZAP/1:1/017XXXX State Change CS_INIT -> CS_ROUTING > > > > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 > > > switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX > > [BREAK] > > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 > > > switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT going to > sleep > > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 > > > switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change > > CS_ROUTING > > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 > > > switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING > > > > 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:384 channel_on_routing() > > > OpenZAP/1:1/017XXXX CHANNEL ROUTING > > > > 2009-01-27 15:32:45 [DEBUG] switch_ivr_originate.c:57 > > > originate_on_routing() OpenZAP/1:1/017XXXX State Change CS_ROUTING -> > > CS_CONSUME_MEDIA > > > > 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 > > > switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX > > [BREAK] > > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 > > > switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING going to > > sleep > > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 > > > switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change > > CS_CONSUME_MEDIA > > > > 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:442 > > > switch_core_session_run() (OpenZAP/1:1/017XXXX) State CONSUME_MEDIA > > > > 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE > > > [DIALING] > > > > 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:957 q931_rx_32() WRITE 55 > > > > > > > > > > > > -- > > > > NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL > > > > f?r nur 16,37 EURO/mtl.!* > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at 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 at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit > allen: > > http://www.gmx.net/de/go/multimessenger > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a From brian at freeswitch.org Wed Jan 28 08:11:36 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 28 Jan 2009 10:11:36 -0600 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: <20090128160102.126250@gmx.net> References: <20090127164104.207050@gmx.net> <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> <20090128132438.309550@gmx.net> <191c3a030901280607r706b4a8cp4cc586fc62cb5086@mail.gmail.com> <20090128160102.126250@gmx.net> Message-ID: Have you tried with the latest FreeSWITCH Trunk? /b On Jan 28, 2009, at 10:01 AM, curly2009 at gmx.de wrote: > It's ISDN PRI. Digium Wildcard TE110P > > I don't edit tones.conf. I leave this to the defaults. > > I using > freeswitch-1.0.1 > zaptel-1.4.12 > > I posted the whole log of the call in pastebin.freeswitch.org. From ronmccar at gmail.com Wed Jan 28 08:14:47 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Wed, 28 Jan 2009 09:14:47 -0700 Subject: [Freeswitch-users] Another way to set CallerID? In-Reply-To: References: <3885f4fe0901261705u3c07fad4vbbf1d3ed394e1ac1@mail.gmail.com> <497FC263.2090305@freeswitch.org> <3885f4fe0901271838m203ff554kaed5d34e64b38a5f@mail.gmail.com> <200901281253.59018.krivushinme@rn-inform.tomsk.ru> Message-ID: <3885f4fe0901280814h52cfde50m43b94ae1a7982405@mail.gmail.com> Ahh, I see that now. Yeah I can add a little side not there, I would of never found it, did not think to look for invite :) On Wed, Jan 28, 2009 at 3:41 AM, Brian West wrote: > The Wiki is a community resource. The best part is anyone can add/ > update info on the wiki if they register for an account, its also a > good way to give back to the community. > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#Gateway > > This is however on the wiki and maybe it just needs improvement. ;) > > /b > > > On Jan 28, 2009, at 12:53 AM, ???????? ?????? wrote: > > > On Wednesday 28 January 2009 08:38:17 Ron McCarthy wrote: > >> Yeah that fixed it! > >> > >> I have never even seen this option in the docs before, but that > >> sure did > >> the trick. > > Please, add this to wiki. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/5e3fc2c0/attachment.html From ajlong at worldlink.net Wed Jan 28 08:15:26 2009 From: ajlong at worldlink.net (Adam Long) Date: Wed, 28 Jan 2009 11:15:26 -0500 Subject: [Freeswitch-users] Sending SIP calls via outbound proxy In-Reply-To: <2E3152CB56588D4A91D4F1746A7252FF4D55EE@isr-brass.interwise.com> References: <49417D96.1090805@junctionnetworks.com> <87f2f3b90812111259ob0a78bbw747afdc52251c2cb@mail.gmail.com> <2E3152CB56588D4A91D4F1746A7252FF4D55EE@isr-brass.interwise.com> Message-ID: <012501c98163$a1c6de40$e5549ac0$@net> >From the Wiki.. Specifying SIP Proxy With fs_path You can route a call through a specific SIP proxy by using the "fs_path" directive. Example: sofia/foo/user at that.domain;fs_path=sip:proxy.this.domain Regards, -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Simon Shaw Sent: Wednesday, January 28, 2009 9:52 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending SIP calls via outbound proxy Is there a mechanism to configure FS to work with an edge proxy? What I am attempting to achieve is a front end proxy that all the clients connect to which simply forwards all messages to FS. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, December 11, 2008 11:00 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending SIP calls via outbound proxy On Thu, Dec 11, 2008 at 12:52 PM, Erick Johnson wrote: > Thanks Dave, > > Actually I realized my problem (stupid mistake of course). For anyone else > trying to use the fs_path variable the value needs to be a fully > qualified SIP > URI, e.g. "bob at bar.com;fs_path=sip:host.domain.net", notice it being > prefaced > with the "sip:", my problem was that I was only entering > the host name. Then somewhere down in mod_sofia it must have decided that > it didn't like that and just closed the channel. Erick, thanks for the clarification! I'll get it put on the wiki right away. -MC > > Hope this helps somebody who gets stuck like I did. > > Cheers, > > Erick > >> Hi Erick, >> >> Not sure if you've tried this (or if it'll help), but you can force >> routing in the dialplan like so: >> >> >> >> Cheers -- >> >> Dave > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From curly2009 at gmx.de Wed Jan 28 08:23:14 2009 From: curly2009 at gmx.de (curly2009 at gmx.de) Date: Wed, 28 Jan 2009 17:23:14 +0100 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: References: <20090127164104.207050@gmx.net> <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> <20090128132438.309550@gmx.net> <191c3a030901280607r706b4a8cp4cc586fc62cb5086@mail.gmail.com> <20090128160102.126250@gmx.net> Message-ID: <20090128162314.243640@gmx.net> no I don't have try it. I have to execute "make current" in the Freeswitch source directory, or something else? -------- Original-Nachricht -------- > Datum: Wed, 28 Jan 2009 10:11:36 -0600 > Von: Brian West > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] Can\'t dial over openzap > Have you tried with the latest FreeSWITCH Trunk? > > /b > > On Jan 28, 2009, at 10:01 AM, curly2009 at gmx.de wrote: > > > It's ISDN PRI. Digium Wildcard TE110P > > > > I don't edit tones.conf. I leave this to the defaults. > > > > I using > > freeswitch-1.0.1 > > zaptel-1.4.12 > > > > I posted the whole log of the call in pastebin.freeswitch.org. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a From msc at freeswitch.org Wed Jan 28 09:02:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 28 Jan 2009 09:02:35 -0800 Subject: [Freeswitch-users] is there an event for SIP MESSAGE? In-Reply-To: References: <60F3399A-F2EC-4F1F-90D0-ED45720F2F04@freeswitch.org> Message-ID: <87f2f3b90901280902k658966e2g286e347476f943fd@mail.gmail.com> Cavalera, I spoke with MikeJ and he said that FS simply does not fire an event for SIP IM messages. It would probably take a fair amount of hacking to make this happen. I can see only two choices for you: see if you or someone you know is willing to modify the source code or ask the developers for some paid support at consulting at freeswitch.org. I'm sorry that I didn't have a better answer for you! -MC On Wed, Jan 28, 2009 at 2:24 AM, Cavalera Claudio Luigi wrote: > freeswitch-users-bounces at lists.freeswitch.org wrote: >> Out of curiosity which SIP messages have you been watching for on the >> event socket? Also, how are you connected to the event socket? Are you >> subscribing to all events and sifting through them to confirm that no >> events are being fired when SIP messages are being sent? -MC > > > Thanks for your interest Michael, > I'll try to explain better since I think this could become a useful > feature; fs already provides a lot of features for Unified > Communication. Moreover SIP messages are supported almost for free from > the sofia sip stack. > > When experimenting I usually connect to the event socket with netcat: > nc localhost 8021 > auth ClueCon > event plain all > > If i register two clients to the event socket they can SIP message each > other. > I'm meaning SIP MESSAGES like this one sent from user 1000 to 1001: > http://pastebin.freeswitch.org/6940 > > I really would like an event to be fired by this SIP MESSAGING in fact > I'm almost adding it at mod_sofia.h > as #define MY_EVENT_SIP_MESSAGE "sofia::message" > :-) > > Firing an event for every SIP MESSAGE sent through fs would be a great > thing, but sadly not sufficient for building a full IM solution. > In fact we would also need an API to send SIP MESSAGES from within fs > and this is already partially achieved from what I've seen with command > like this one sent to the event socket: > > sendevent SEND_MESSAGE > profile: internal > content-length: 2 > content-type: application/simple-message-summary > user: 1001 > host: 192.168.1.1 > > Hi > Content-Type: command/reply > Reply-Text: +OK > > This command works and fires an event like this: > > Command: sendevent%20SEND_MESSAGE > profile: internal > content-length: 2 > content-type: application/simple-message-summary > user: 1001 > host: 192.168.1.1 > Event-Name: SEND_MESSAGE > Core-UUID: a7f40fea-8340-4d77-aded-f322f9bee016 > FreeSWITCH-Hostname: lallobox > FreeSWITCH-IPv4: 192.168.1.1 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-01-28%2010%3A44%3A07 > Event-Date-GMT: Wed,%2028%20Jan%202009%2009%3A44%3A07%20GMT > Event-Date-Timestamp: 1233135847314907 > Event-Calling-File: mod_event_socket.c > Event-Calling-Function: parse_command > Event-Calling-Line-Number: 1511 > Content-Length: 2 > > Hi > > > However it would be wonderful to enhance this API with the possibility > to select the sending user thus being able to forge sip messages from > within fs on behalf of a registered or not user. > > > Best Regards, > 2C > > PS: Michael feel free to contact me off list as you see fit. > > > 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. > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Wed Jan 28 09:07:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 28 Jan 2009 09:07:38 -0800 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <49802B7F.70100@ewetel.de> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <497B0563.9090701@ewetel.de> <497F19E3.70900@ewetel.de> <191c3a030901270715r538f45e4g615dffa639a63c20@mail.gmail.com> <87f2f3b90901271057x323127cas9ace08c2573783c@mail.gmail.com> <49802B7F.70100@ewetel.de> Message-ID: <87f2f3b90901280907y3d9e1c57t4a573685318bde49@mail.gmail.com> Nice work! Let us know when you'd like others to try it out. -MC On Wed, Jan 28, 2009 at 1:55 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello Guys, > > today I tried to decode a q931 pcap file directly in a linux cli. I had > success and the result looks like this: > > - -- SENDING ------- Packet number: 00001 --- SpanID: 1 ---------------- > Protocol discriminator: Q.931 > Call reference value length: 2 > Call reference flag: Message sent from originating side > Call reference value: 0002 > Message type: RELEASE (0x4d) > Display 'HK at FreeSWITCH' > Information element: Display > Length: 13 > Display information: HK at FreeSWITCH > > > - -- SENDING ------- Packet number: 00002 --- SpanID: 1 ---------------- > Protocol discriminator: Q.931 > Call reference value length: 2 > Call reference flag: Message sent to originating side > Call reference value: 0002 > Message type: DISCONNECT (0x45) > Cause > Information element: Cause > Length: 2 > .... 0000 = Cause location: User (U) (0) > .00. .... = Coding standard: ITU-T standardized coding (0x00) > 1... .... = Extension indicator: last octet > .001 0000 = Cause value: Normal call clearing (16) > 1... .... = Extension indicator: last octet > > > I think this helps server admins to get a much faster access to the > decoded packages on a pure server where they have no GUI and hence no > wireshark. tshark is needed for this of course. > > > regards > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkmAK38ACgkQ4tZeNddg3dxAEACgsf+GC3jqTvBUYD2pqsgtZgUs > s8QAoLIitPAc0I55zKXyw6yTe4MDaDaK > =/Ir7 > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kokoska.rokoska at post.cz Wed Jan 28 09:27:50 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 28 Jan 2009 18:27:50 +0100 Subject: [Freeswitch-users] DTMF with Early Media Disabled In-Reply-To: <20090128135525.153200@gmx.net> References: <20090128050402.75810@gmx.net> <20090128135525.153200@gmx.net> Message-ID: <49809596.8060704@post.cz> Klaus Teller napsal(a): > I know it works perfectly when pre_answer is called. That is, when early media is activated. I was just trying to figure out what is the expected behavior when pre_answer is not called. > > I want to get DTMF from users without having them billed by their carriers. I've heard that some carriers start billing as soon as early media is on. Really? It violates the law in most Europian countries :-) I think you are from Germany - if such a carrier does it, the best you can do is to contact your Telco Regulator (Die Bundesnetzagentur - www.bundesnetzagentur.de) and ask for official correction. Best regards, kokoska.rokoska From msc at freeswitch.org Wed Jan 28 09:28:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 28 Jan 2009 09:28:35 -0800 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> Message-ID: <87f2f3b90901280928g420fed33pc691f7c61a9a77d0@mail.gmail.com> FYI, I added a note on the reporting bugs page that specifically talks about this scenario: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Upgrading_From_Older_Revisions -MC On Wed, Jan 28, 2009 at 6:01 AM, Anthony Minessale wrote: > Also remember, > Actually completely uninstall and erase /usr/local/freeswitch and the 1.0.1 > source tree and freshly install the new one. > If you try to upgrade on top of a release with trunk it will cause more > problems for you. > > > On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice wrote: >> >> Upgrade to trunk... Many many issues have been resolved since 1.0.1 was >> the >> current release >> >> >> > From: shehzad p >> > Reply-To: >> > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) >> > To: >> > Subject: [Freeswitch-users] Freeswitch freezes on increasing call >> > traffic >> > >> > >> > Hi all, >> > >> > Yesterday my Freeswitch server faced a problem when call traffic >> > increased >> > to more than 100. >> > >> > When I start Freeswitch, it works fine and then after some time >> > (approximately 15 to 20 minutes) it stops functioning (means no call is >> > being processed, no CLI command is working and it just freezes) until I >> > restart the freeswitch. >> > >> > I am using Freeswitch 1.0.1. >> > Debug (gdb) trace as on wiki page >> > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.sh is >> > attached >> > http://www.nabble.com/file/p21701744/fs_debgu.txt fs_debgu.txt >> > -- >> > View this message in context: >> > >> > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 >> > p21701744.html >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Wed Jan 28 09:34:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 28 Jan 2009 09:34:16 -0800 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: References: Message-ID: <87f2f3b90901280934y33d4196u3760da1b1b951797@mail.gmail.com> Could you add some detail to this question? I think we could give a more intelligent answer if we had a better idea of what you are hoping to accomplish. I'm sure the FS community will have some thoughts for you. -MC On Wed, Jan 28, 2009 at 5:41 AM, Boris Lansky wrote: > What way should I configure Freeswitch to get out all the outbound calls > through a specific proxy server? > > > > Regards, > > > > Boris Lansky > Unified Communications Telephony Team > > AT&T Unified Communications > Phone: +972.3.976.7604 > > Fax: +972.3.976.7712 > > blansky at interwise.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Wed Jan 28 09:39:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 28 Jan 2009 09:39:21 -0800 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: <20090128162314.243640@gmx.net> References: <20090127164104.207050@gmx.net> <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> <20090128132438.309550@gmx.net> <191c3a030901280607r706b4a8cp4cc586fc62cb5086@mail.gmail.com> <20090128160102.126250@gmx.net> <20090128162314.243640@gmx.net> Message-ID: <87f2f3b90901280939x1153a33fvafb40383809f4eba@mail.gmail.com> I just added a small section to the reporting bugs page on the wiki. It specifically talks about this subject: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Upgrading_From_Older_Revisions In simple terms, you are best to copy your configs to a safe location, then delete your /usr/src/freeswitch directory, then delete your fs source directory, and then do a fresh checkout & install. Personally, on Linux I like doing the "quick and dirty install": http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install After your fresh install is complete, go back and make your customizations, like editing the modules.conf and modules.conf.xml, installing openzap, etc. Then test and report back. -MC On Wed, Jan 28, 2009 at 8:23 AM, wrote: > no I don't have try it. > > I have to execute "make current" in the Freeswitch source directory, or something else? > > -------- Original-Nachricht -------- >> Datum: Wed, 28 Jan 2009 10:11:36 -0600 >> Von: Brian West >> An: freeswitch-users at lists.freeswitch.org >> Betreff: Re: [Freeswitch-users] Can\'t dial over openzap > >> Have you tried with the latest FreeSWITCH Trunk? >> >> /b >> >> On Jan 28, 2009, at 10:01 AM, curly2009 at gmx.de wrote: >> >> > It's ISDN PRI. Digium Wildcard TE110P >> > >> > I don't edit tones.conf. I leave this to the defaults. >> > >> > I using >> > freeswitch-1.0.1 >> > zaptel-1.4.12 >> > >> > I posted the whole log of the call in pastebin.freeswitch.org. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL > f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From e.schmidbauer at gmail.com Wed Jan 28 09:40:11 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Wed, 28 Jan 2009 12:40:11 -0500 Subject: [Freeswitch-users] portaudio error Message-ID: <2cef777b0901280940n6ab98cdeu4a9561cd3226ed98@mail.gmail.com> hey guys im getting an error when trying to use a USB mic with portaudio. wondering if anyone could help me solve this problem. im using ubuntu intrepid studio. portaudio is loaded correctly and sees the all the devices. when i do a "pa call" using the built in sound card, i am able to connect to the call. but when i set the "pa indev #7" (#7 is the USB mic) and then do a "pa call" i get the following error >pa call 4800 2009-01-28 12:33:09 [NOTICE] switch_channel.c:566 switch_channel_set_name() New Channel portaudio/4800 [bae25c96-ed61-11dd-a1c3-d19a8f52ba15] 2009-01-28 12:33:09 [ERR] mod_portaudio.c:1350 engage_device() Error opening audio device retrying 2009-01-28 12:33:11 [ERR] mod_portaudio.c:1359 engage_device() Can't open audio device 2009-01-28 12:33:11 [NOTICE] mod_portaudio.c:1739 place_call() Close Channel portaudio/4800 [CS_NEW] API CALL [pa(call 4800)] output: FAIL:Device Error! here is the output from "pa devlist" > pa devlist API CALL [pa(devlist)] output: 0;/dev/dsp;16;16;r,o 1;/dev/dsp1;16;0; 2;Intel ICH6: Intel ICH6 (hw:0,0);2;2; 3;Intel ICH6: Intel ICH6 - MIC ADC (hw:0,1);2;0; 4;Intel ICH6: Intel ICH6 - MIC2 ADC (hw:0,2);2;0; 5;Intel ICH6: Intel ICH6 - ADC2 (hw:0,3);2;0; 6;Intel ICH6: Intel ICH6 - IEC958 (hw:0,4);0;2; 7;Samson C01U : USB Audio (hw:1,0);2;0;i 8;front;0;2; 9;default;128;128; 10;dmix;0;2; any input would be much appreciated. thanks again for developing such a fine piece of software! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/035a191c/attachment.html From brian at freeswitch.org Wed Jan 28 09:43:51 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 28 Jan 2009 11:43:51 -0600 Subject: [Freeswitch-users] portaudio error In-Reply-To: <2cef777b0901280940n6ab98cdeu4a9561cd3226ed98@mail.gmail.com> References: <2cef777b0901280940n6ab98cdeu4a9561cd3226ed98@mail.gmail.com> Message-ID: <649220E0-42AF-4F79-9328-3E6111436D99@freeswitch.org> Maybe the Mic can't be opened at 48000? Can you verify what rate you're trying to use? /b On Jan 28, 2009, at 11:40 AM, e schmidbauer wrote: > hey guys im getting an error when trying to use a USB mic with > portaudio. wondering if anyone could help me solve this problem. > im using ubuntu intrepid studio. portaudio is loaded correctly and > sees the all the devices. when i do a "pa call" using the built in > sound card, i am able to connect to the call. but when i set the "pa > indev #7" (#7 is the USB mic) and then do a "pa call" i get the > following error > > >pa call 4800 > 2009-01-28 12:33:09 [NOTICE] switch_channel.c:566 > switch_channel_set_name() New Channel portaudio/4800 [bae25c96- > ed61-11dd-a1c3-d19a8f52ba15] > 2009-01-28 12:33:09 [ERR] mod_portaudio.c:1350 engage_device() Error > opening audio device retrying > 2009-01-28 12:33:11 [ERR] mod_portaudio.c:1359 engage_device() Can't > open audio device > 2009-01-28 12:33:11 [NOTICE] mod_portaudio.c:1739 place_call() Close > Channel portaudio/4800 [CS_NEW] > API CALL [pa(call 4800)] output: > FAIL:Device Error! > > here is the output from "pa devlist" > > > pa devlist > API CALL [pa(devlist)] output: > 0;/dev/dsp;16;16;r,o > 1;/dev/dsp1;16;0; > 2;Intel ICH6: Intel ICH6 (hw:0,0);2;2; > 3;Intel ICH6: Intel ICH6 - MIC ADC (hw:0,1);2;0; > 4;Intel ICH6: Intel ICH6 - MIC2 ADC (hw:0,2);2;0; > 5;Intel ICH6: Intel ICH6 - ADC2 (hw:0,3);2;0; > 6;Intel ICH6: Intel ICH6 - IEC958 (hw:0,4);0;2; > 7;Samson C01U : USB Audio (hw:1,0);2;0;i > 8;front;0;2; > 9;default;128;128; > 10;dmix;0;2; > > any input would be much appreciated. thanks again for developing > such a fine piece of software! From e.schmidbauer at gmail.com Wed Jan 28 10:07:27 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Wed, 28 Jan 2009 13:07:27 -0500 Subject: [Freeswitch-users] portaudio error In-Reply-To: <649220E0-42AF-4F79-9328-3E6111436D99@freeswitch.org> References: <2cef777b0901280940n6ab98cdeu4a9561cd3226ed98@mail.gmail.com> <649220E0-42AF-4F79-9328-3E6111436D99@freeswitch.org> Message-ID: <2cef777b0901281007w7cb465cla763f62bc204a534@mail.gmail.com> portaudio.conf.xml is set to 48000, and 10i On Wed, Jan 28, 2009 at 12:43 PM, Brian West wrote: > Maybe the Mic can't be opened at 48000? Can you verify what rate > you're trying to use? > > /b > > On Jan 28, 2009, at 11:40 AM, e schmidbauer wrote: > > > hey guys im getting an error when trying to use a USB mic with > > portaudio. wondering if anyone could help me solve this problem. > > im using ubuntu intrepid studio. portaudio is loaded correctly and > > sees the all the devices. when i do a "pa call" using the built in > > sound card, i am able to connect to the call. but when i set the "pa > > indev #7" (#7 is the USB mic) and then do a "pa call" i get the > > following error > > > > >pa call 4800 > > 2009-01-28 12:33:09 [NOTICE] switch_channel.c:566 > > switch_channel_set_name() New Channel portaudio/4800 [bae25c96- > > ed61-11dd-a1c3-d19a8f52ba15] > > 2009-01-28 12:33:09 [ERR] mod_portaudio.c:1350 engage_device() Error > > opening audio device retrying > > 2009-01-28 12:33:11 [ERR] mod_portaudio.c:1359 engage_device() Can't > > open audio device > > 2009-01-28 12:33:11 [NOTICE] mod_portaudio.c:1739 place_call() Close > > Channel portaudio/4800 [CS_NEW] > > API CALL [pa(call 4800)] output: > > FAIL:Device Error! > > > > here is the output from "pa devlist" > > > > > pa devlist > > API CALL [pa(devlist)] output: > > 0;/dev/dsp;16;16;r,o > > 1;/dev/dsp1;16;0; > > 2;Intel ICH6: Intel ICH6 (hw:0,0);2;2; > > 3;Intel ICH6: Intel ICH6 - MIC ADC (hw:0,1);2;0; > > 4;Intel ICH6: Intel ICH6 - MIC2 ADC (hw:0,2);2;0; > > 5;Intel ICH6: Intel ICH6 - ADC2 (hw:0,3);2;0; > > 6;Intel ICH6: Intel ICH6 - IEC958 (hw:0,4);0;2; > > 7;Samson C01U : USB Audio (hw:1,0);2;0;i > > 8;front;0;2; > > 9;default;128;128; > > 10;dmix;0;2; > > > > any input would be much appreciated. thanks again for developing > > such a fine piece of software! > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/ac486fbd/attachment.html From brian at freeswitch.org Wed Jan 28 10:27:28 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 28 Jan 2009 12:27:28 -0600 Subject: [Freeswitch-users] portaudio error In-Reply-To: <2cef777b0901281007w7cb465cla763f62bc204a534@mail.gmail.com> References: <2cef777b0901280940n6ab98cdeu4a9561cd3226ed98@mail.gmail.com> <649220E0-42AF-4F79-9328-3E6111436D99@freeswitch.org> <2cef777b0901281007w7cb465cla763f62bc204a534@mail.gmail.com> Message-ID: Output of "pa dump" ? /b On Jan 28, 2009, at 12:07 PM, e schmidbauer wrote: > portaudio.conf.xml is set to 48000, and 10i From palletboy at gmail.com Wed Jan 28 08:41:45 2009 From: palletboy at gmail.com (J. G.) Date: Wed, 28 Jan 2009 11:41:45 -0500 Subject: [Freeswitch-users] Hello! Message-ID: <3093591d0901280841i3c5355ado6511acfe49d99030@mail.gmail.com> FS users, Just joined the growing list of FS users and found this list. I spend a lot of time in Asterisk (own a 48 seat call center that uses an Asterisk distro as our main switch, etc) and am playing with my first integration between FS and Asterisk, so far so good - looking forward to Anders' release of Phonebooth for the masses. Anyway, glad to be on the list! J. -- ----- Jason Gehman General Manager North Voice Communications www.NorthVC.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/28bfefc4/attachment.html From mike at jerris.com Wed Jan 28 11:10:14 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 28 Jan 2009 14:10:14 -0500 Subject: [Freeswitch-users] is there an event for SIP MESSAGE? In-Reply-To: References: Message-ID: <7E8E9F0C-0356-4289-BBF9-EE654C39526D@jerris.com> On Jan 28, 2009, at 5:24 AM, Cavalera Claudio Luigi wrote: > freeswitch-users-bounces at lists.freeswitch.org wrote: >> Out of curiosity which SIP messages have you been watching for on the >> event socket? Also, how are you connected to the event socket? Are >> you >> subscribing to all events and sifting through them to confirm that no >> events are being fired when SIP messages are being sent? -MC > > > Thanks for your interest Michael, > I'll try to explain better since I think this could become a useful > feature; fs already provides a lot of features for Unified > Communication. Moreover SIP messages are supported almost for free > from > the sofia sip stack. > > When experimenting I usually connect to the event socket with netcat: > nc localhost 8021 > auth ClueCon > event plain all > > If i register two clients to the event socket they can SIP message > each > other. > I'm meaning SIP MESSAGES like this one sent from user 1000 to 1001: > http://pastebin.freeswitch.org/6940 > > I really would like an event to be fired by this SIP MESSAGING in fact > I'm almost adding it at mod_sofia.h > as #define MY_EVENT_SIP_MESSAGE "sofia::message" > :-) > > Firing an event for every SIP MESSAGE sent through fs would be a great > thing, but sadly not sufficient for building a full IM solution. > In fact we would also need an API to send SIP MESSAGES from within fs > and this is already partially achieved from what I've seen with > command > like this one sent to the event socket: > > sendevent SEND_MESSAGE > profile: internal > content-length: 2 > content-type: application/simple-message-summary > user: 1001 > host: 192.168.1.1 > > Hi > Content-Type: command/reply > Reply-Text: +OK > > This command works and fires an event like this: > > Command: sendevent%20SEND_MESSAGE > profile: internal > content-length: 2 > content-type: application/simple-message-summary > user: 1001 > host: 192.168.1.1 > Event-Name: SEND_MESSAGE > Core-UUID: a7f40fea-8340-4d77-aded-f322f9bee016 > FreeSWITCH-Hostname: lallobox > FreeSWITCH-IPv4: 192.168.1.1 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-01-28%2010%3A44%3A07 > Event-Date-GMT: Wed,%2028%20Jan%202009%2009%3A44%3A07%20GMT > Event-Date-Timestamp: 1233135847314907 > Event-Calling-File: mod_event_socket.c > Event-Calling-Function: parse_command > Event-Calling-Line-Number: 1511 > Content-Length: 2 > > Hi > > > However it would be wonderful to enhance this API with the possibility > to select the sending user thus being able to forge sip messages from > within fs on behalf of a registered or not user. > > > Best Regards, > 2C > > PS: Michael feel free to contact me off list as you see fit. > > > 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. > ----------------------------------------------------------------------------------------------------- I am sorry to see that your e-mail is confidential. If it were not, I would have responded to tell you that there is already an api command to send chat messages in a protocol agnostic way (see http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_chat) . In the future if you can send non-confidential emails to the list it will greatly assist us in being able to respond to your questions. Mike From mike at jerris.com Wed Jan 28 11:16:06 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 28 Jan 2009 14:16:06 -0500 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: References: Message-ID: <7160CD9F-F06E-4964-943A-F4B0C12150CC@jerris.com> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010725.html On Jan 28, 2009, at 8:41 AM, Boris Lansky wrote: > What way should I configure Freeswitch to get out all the outbound > calls through a specific proxy server? > > Regards, > > Boris Lansky > Unified Communications Telephony Team > AT&T Unified Communications > Phone: +972.3.976.7604 > Fax: +972.3.976.7712 > blansky at interwise.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/234aa01d/attachment.html From anthony.minessale at gmail.com Wed Jan 28 11:49:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 28 Jan 2009 13:49:03 -0600 Subject: [Freeswitch-users] Hello! In-Reply-To: <3093591d0901280841i3c5355ado6511acfe49d99030@mail.gmail.com> References: <3093591d0901280841i3c5355ado6511acfe49d99030@mail.gmail.com> Message-ID: <191c3a030901281149k708e48ex1898d04f5b27fed5@mail.gmail.com> Have fun. We also have an irc channel #freeswitch on irc.freenode.net On Wed, Jan 28, 2009 at 10:41 AM, J. G. wrote: > FS users, > Just joined the growing list of FS users and found this list. I spend a > lot of time in Asterisk (own a 48 seat call center that uses an Asterisk > distro as our main switch, etc) and am playing with my first integration > between FS and Asterisk, so far so good - looking forward to Anders' release > of Phonebooth for the masses. > > Anyway, glad to be on the list! > J. > > -- > ----- > Jason Gehman > General Manager > North Voice Communications > www.NorthVC.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/3d0648bb/attachment.html From msc at freeswitch.org Wed Jan 28 13:14:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 28 Jan 2009 13:14:35 -0800 Subject: [Freeswitch-users] Hello! In-Reply-To: <3093591d0901280841i3c5355ado6511acfe49d99030@mail.gmail.com> References: <3093591d0901280841i3c5355ado6511acfe49d99030@mail.gmail.com> Message-ID: <87f2f3b90901281314u53bd009avd91bde2d9861ef8f@mail.gmail.com> And don't forget our "comprehensive" documentation site: wiki.freeswitch.org :) -MC On Wed, Jan 28, 2009 at 8:41 AM, J. G. wrote: > FS users, > Just joined the growing list of FS users and found this list. I spend a lot > of time in Asterisk (own a 48 seat call center that uses an Asterisk distro > as our main switch, etc) and am playing with my first integration between FS > and Asterisk, so far so good - looking forward to Anders' release of > Phonebooth for the masses. > > Anyway, glad to be on the list! > J. > > -- > ----- > Jason Gehman > General Manager > North Voice Communications > www.NorthVC.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From bdeacon at highergear.com Wed Jan 28 14:31:29 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Wed, 28 Jan 2009 14:31:29 -0800 Subject: [Freeswitch-users] Possible to mock out _freeswitch.py? Message-ID: <1233181889.4757.35.camel@dev03.cal.highergear.com> Greetings, Actually any advice on how to test or diagnose my mod_python code would be appreciated. For the most part I've been trying to keep as much of my code as possible separated from the actual interaction with freeswitch so that it's separately testable. But I will occasionally run into: [ERR] mod_python.c:121 eval_some_python() Error importing module This latest is, I think some missing modules in site_packages, but it'd be nice to get some output hinting at that. So nice would be able to have a mocked out _freeswitch so the rest of my unit test stuff could run. Or failing that, is there a switch I can flip to get some of the python errors output to the log? TIA, Brian From brian at freeswitch.org Wed Jan 28 14:39:16 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 28 Jan 2009 16:39:16 -0600 Subject: [Freeswitch-users] Possible to mock out _freeswitch.py? In-Reply-To: <1233181889.4757.35.camel@dev03.cal.highergear.com> References: <1233181889.4757.35.camel@dev03.cal.highergear.com> Message-ID: Have you see this http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/languages/mod_python/python_example.py?r=9052 /b On Jan 28, 2009, at 4:31 PM, Brian Deacon wrote: > Greetings, > > Actually any advice on how to test or diagnose my mod_python code > would > be appreciated. For the most part I've been trying to keep as much of > my code as possible separated from the actual interaction with > freeswitch so that it's separately testable. But I will occasionally > run into: > [ERR] mod_python.c:121 eval_some_python() Error importing module > > This latest is, I think some missing modules in site_packages, but > it'd > be nice to get some output hinting at that. So nice would be able to > have a mocked out _freeswitch so the rest of my unit test stuff could > run. Or failing that, is there a switch I can flip to get some of the > python errors output to the log? > > TIA, > Brian > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdeacon at highergear.com Wed Jan 28 16:07:41 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Wed, 28 Jan 2009 16:07:41 -0800 Subject: [Freeswitch-users] Possible to mock out _freeswitch.py? In-Reply-To: References: <1233181889.4757.35.camel@dev03.cal.highergear.com> Message-ID: <1233187661.4757.46.camel@dev03.cal.highergear.com> I have. But 'says here: http://wiki.freeswitch.org/wiki/Mod_python#Can_I_test_scripts_using_python_shell.3F That the "import _freeswitch" in freeswitch.py breaks except when the script is run by mod_python. (At which point there is some python voodoo going on here that is beyond me. Is _freeswitch a runtime-generated SWIGism?) So I was thinking I could dump a _freeswitch.py onto the PYTHONPATH when trying to debug code outside of mod_python. But probably easier for trivial things would be to just get the python error output to spit to the log, but it seems to be getting swallowed. Or was there something else in python_example.py that I've been too dense to notice? Brian On Wed, 2009-01-28 at 16:39 -0600, Brian West wrote: > Have you see this > > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/languages/mod_python/python_example.py?r=9052 > > /b > > On Jan 28, 2009, at 4:31 PM, Brian Deacon wrote: > > > Greetings, > > > > Actually any advice on how to test or diagnose my mod_python code > > would > > be appreciated. For the most part I've been trying to keep as much of > > my code as possible separated from the actual interaction with > > freeswitch so that it's separately testable. But I will occasionally > > run into: > > [ERR] mod_python.c:121 eval_some_python() Error importing module > > > > This latest is, I think some missing modules in site_packages, but > > it'd > > be nice to get some output hinting at that. So nice would be able to > > have a mocked out _freeswitch so the rest of my unit test stuff could > > run. Or failing that, is there a switch I can flip to get some of the > > python errors output to the log? > > > > TIA, > > Brian > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Jan 28 16:20:22 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 28 Jan 2009 18:20:22 -0600 Subject: [Freeswitch-users] Possible to mock out _freeswitch.py? In-Reply-To: <1233187661.4757.46.camel@dev03.cal.highergear.com> References: <1233181889.4757.35.camel@dev03.cal.highergear.com> <1233187661.4757.46.camel@dev03.cal.highergear.com> Message-ID: <084BC3EC-C8AB-4E9A-9139-411A0A92D790@freeswitch.org> On Jan 28, 2009, at 6:07 PM, Brian Deacon wrote: > I have. But 'says here: > http://wiki.freeswitch.org/wiki/Mod_python#Can_I_test_scripts_using_python_shell.3F > > That the "import _freeswitch" in freeswitch.py breaks except when the > script is run by mod_python. (At which point there is some python > voodoo going on here that is beyond me. Is _freeswitch a > runtime-generated SWIGism?) Correct you can't do that unless its running in mod_python inside FreeSWITCH. > > > So I was thinking I could dump a _freeswitch.py onto the PYTHONPATH > when > trying to debug code outside of mod_python. But probably easier for > trivial things would be to just get the python error output to spit to > the log, but it seems to be getting swallowed. > > Or was there something else in python_example.py that I've been too > dense to notice? I think the key is to know you can only run it inside FreeSWITCH. > > > Brian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/bb4329af/attachment-0001.html From anthony.minessale at gmail.com Wed Jan 28 16:23:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 28 Jan 2009 18:23:16 -0600 Subject: [Freeswitch-users] Possible to mock out _freeswitch.py? In-Reply-To: <1233187661.4757.46.camel@dev03.cal.highergear.com> References: <1233181889.4757.35.camel@dev03.cal.highergear.com> <1233187661.4757.46.camel@dev03.cal.highergear.com> Message-ID: <191c3a030901281623y1b81e8cat16c31eac6cf3b74a@mail.gmail.com> very much voodoo going on. The freeswitch.py is generated by swig to link the FS api into FS itself so when the python script runs it uses the glue to get to the .so who in turn references symbols back into FS itself. So it can't run outside of FS because it depends on symbols from the core. On Wed, Jan 28, 2009 at 6:07 PM, Brian Deacon wrote: > I have. But 'says here: > > http://wiki.freeswitch.org/wiki/Mod_python#Can_I_test_scripts_using_python_shell.3F > > That the "import _freeswitch" in freeswitch.py breaks except when the > script is run by mod_python. (At which point there is some python > voodoo going on here that is beyond me. Is _freeswitch a > runtime-generated SWIGism?) > > So I was thinking I could dump a _freeswitch.py onto the PYTHONPATH when > trying to debug code outside of mod_python. But probably easier for > trivial things would be to just get the python error output to spit to > the log, but it seems to be getting swallowed. > > Or was there something else in python_example.py that I've been too > dense to notice? > > Brian > > > On Wed, 2009-01-28 at 16:39 -0600, Brian West wrote: > > Have you see this > > > > > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/languages/mod_python/python_example.py?r=9052 > > > > /b > > > > On Jan 28, 2009, at 4:31 PM, Brian Deacon wrote: > > > > > Greetings, > > > > > > Actually any advice on how to test or diagnose my mod_python code > > > would > > > be appreciated. For the most part I've been trying to keep as much of > > > my code as possible separated from the actual interaction with > > > freeswitch so that it's separately testable. But I will occasionally > > > run into: > > > [ERR] mod_python.c:121 eval_some_python() Error importing module > > > > > > This latest is, I think some missing modules in site_packages, but > > > it'd > > > be nice to get some output hinting at that. So nice would be able to > > > have a mocked out _freeswitch so the rest of my unit test stuff could > > > run. Or failing that, is there a switch I can flip to get some of the > > > python errors output to the log? > > > > > > TIA, > > > Brian > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090128/8a3f6b7d/attachment.html From bdeacon at highergear.com Wed Jan 28 18:41:19 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Wed, 28 Jan 2009 18:41:19 -0800 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python Message-ID: <1233196879.4757.54.camel@dev03.cal.highergear.com> Greetings, So I see from nabble and on the wiki that others have had problems with sqlalchemy and the hangup hook, but I appear to be having a more basic problem. I'm running FS 1.0.2, python 2.4.3, and SqlAlchemy 0.4.7 Here's an extension in my public.xml that works: With this silly DummyRouteHandler.py: import os from freeswitch import * #import sqlalchemy def fsapi(session, stream, env, args): stream.write('sofia/internal/1000%10.48.5.207') (Note the commented out sqlalchemy import. Uncommenting the import gets me an "error importing module" or "error reloading module" in the fs_cli output) Oooh, and just now trying to reproduce I got my first crash. Thoughts? Brian From msc at freeswitch.org Wed Jan 28 19:16:57 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 28 Jan 2009 19:16:57 -0800 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <1233196879.4757.54.camel@dev03.cal.highergear.com> References: <1233196879.4757.54.camel@dev03.cal.highergear.com> Message-ID: <45C07A45-AA15-4AC2-8ED0-73F4485CF190@freeswitch.org> Sent from my iPhone On Jan 28, 2009, at 6:41 PM, Brian Deacon wrote: > Greetings, > > So I see from nabble and on the wiki that others have had problems > with > sqlalchemy and the hangup hook, but I appear to be having a more basic > problem. > > I'm running FS 1.0.2, python 2.4.3, and SqlAlchemy 0.4.7 > > Here's an extension in my public.xml that works: > > expression="^6468715186$"> > data="group_confirm_file=voicemail/8000/vm-hello.wav" /> > > > > data="${destination_number_python}" /> > > > > With this silly DummyRouteHandler.py: > import os > from freeswitch import * > #import sqlalchemy > > def fsapi(session, stream, env, args): > stream.write('sofia/internal/1000%10.48.5.207') > > (Note the commented out sqlalchemy import. Uncommenting the import > gets > me an "error importing module" or "error reloading module" in the > fs_cli > output) > > Oooh, and just now trying to reproduce I got my first crash. > > Thoughts? > Use Lua? ;) Seriously, Lua is a breeze to learn and is more stable than Python. Python is *not* designed to be embedded however Lua is designed precisely for that reason. Think WoW. Other than that I believe that your experiences with Python might help the devs uncover more issues. Please visit the "reporting bugs" page on the wiki and start collecting the information that will help them solve this one. -MC > Brian > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From blansky at interwise.com Wed Jan 28 23:22:24 2009 From: blansky at interwise.com (Boris Lansky) Date: Thu, 29 Jan 2009 09:22:24 +0200 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: <87f2f3b90901280934y33d4196u3760da1b1b951797@mail.gmail.com> References: <87f2f3b90901280934y33d4196u3760da1b1b951797@mail.gmail.com> Message-ID: Hi Michael, What I would like to achieve is that Freeswitch will send all its SIP messages to call terminator through a specific Proxy server. From what I know I can cause it to do it by including "Path" header in a User Agent REGISTER message but I would like the FS to behave such way "automatically" due to its configuration. So how looks such a configuration I would like to know. Regards, Boris Lansky Unified Communications Telephony Team AT&T Unified Communications Phone: +972.3.976.7604 Fax: +972.3.976.7712 blansky at interwise.com -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 28, 2009 7:34 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outbound Proxy configuration Could you add some detail to this question? I think we could give a more intelligent answer if we had a better idea of what you are hoping to accomplish. I'm sure the FS community will have some thoughts for you. -MC On Wed, Jan 28, 2009 at 5:41 AM, Boris Lansky wrote: > What way should I configure Freeswitch to get out all the outbound calls > through a specific proxy server? > > > > Regards, > > > > Boris Lansky > Unified Communications Telephony Team > > AT&T Unified Communications > Phone: +972.3.976.7604 > > Fax: +972.3.976.7712 > > blansky at interwise.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From blansky at interwise.com Wed Jan 28 23:27:34 2009 From: blansky at interwise.com (Boris Lansky) Date: Thu, 29 Jan 2009 09:27:34 +0200 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: <7160CD9F-F06E-4964-943A-F4B0C12150CC@jerris.com> References: <7160CD9F-F06E-4964-943A-F4B0C12150CC@jerris.com> Message-ID: Sorry for the stupid question but in what configuration file should I add such line "sofia/foo/user at that.domain ;fs_path=sip:proxy.this.domain" ? Regards, Boris Lansky Unified Communications Telephony Team AT&T Unified Communications Phone: +972.3.976.7604 Fax: +972.3.976.7712 blansky at interwise.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Wednesday, January 28, 2009 9:16 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outbound Proxy configuration http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/0107 25.html On Jan 28, 2009, at 8:41 AM, Boris Lansky wrote: What way should I configure Freeswitch to get out all the outbound calls through a specific proxy server? Regards, Boris Lansky Unified Communications Telephony Team AT&T Unified Communications Phone: +972.3.976.7604 Fax: +972.3.976.7712 blansky at interwise.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/c74d7a01/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 16211 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/c74d7a01/attachment-0001.png From Claudio.Cavalera at italtel.it Thu Jan 29 02:47:51 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 29 Jan 2009 11:47:51 +0100 Subject: [Freeswitch-users] is there an event for SIP MESSAGE? In-Reply-To: <87f2f3b90901280902k658966e2g286e347476f943fd@mail.gmail.com> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > Cavalera, You can call me Claudio, I'm embarassed by not having full control over my email client. :-\ > I spoke with MikeJ and he said that FS simply does not fire an event > for SIP IM messages. It would probably take a fair amount of hacking > to make this happen. I can see only two choices for you: see if you or > someone you know is willing to modify the source code or ask the > developers for some paid support at consulting at freeswitch.org. > > I'm sorry that I didn't have a better answer for you! -MC No problem, I will try to add it by myself for now, maybe looking at another event as example. For now I've found the relevant code in sofia.c void sofia_handle_sip_r_message(...) and the case nua_i_message Ciao, Claudio Internet E. Mail 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. From helmut.kuper at ewetel.de Thu Jan 29 02:54:33 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 29 Jan 2009 11:54:33 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <49802B7F.70100@ewetel.de> References: <4972046E.8020102@ewetel.de> <109394AB-8EEA-4652-B453-2ACFE8CE50C3@freeswitch.org> <49720E24.3050806@ewetel.de> <2752B7AB-C1FC-4B51-A38C-176C9572E633@freeswitch.org> <497B0563.9090701@ewetel.de> <497F19E3.70900@ewetel.de> <191c3a030901270715r538f45e4g615dffa639a63c20@mail.gmail.com> <87f2f3b90901271057x323127cas9ace08c2573783c@mail.gmail.com> <49802B7F.70100@ewetel.de> Message-ID: <49818AE9.7030605@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, a further update: The code is implemented and seems to work as it should. I still can do outgoing calls when q931ToPcap is active :) So it shouldn't be too wrong what I did in the openzap code. The pcap File is generated in FS's log dir either with default filename or with optional given filename. Existing files of same name will be overwritten without a warning. I attached a screenshot of such a pcap file viewed in wireshark. Currently I work on a little linux cli perl script, which displays the decoded Q931 packets on a CLI as showed in my yesterday mail. One thing I couldn't solve is, how to get the span_id when FS receives or send a Q931 packet? In ozmod_isdn functions "static int zap_isdn_921_23" (Receive) and "static int q931_rx_32" (Send) I'm converting *pvt to zap_span_t like this: zap_span_t *span = (zap_span_t *) pvt; Then I try to access span_id by span->span_id, but it seems it's allways zero. Any ideas? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkmBiugACgkQ4tZeNddg3dwQ+QCfTzb3Notfkx0NlLqXd6xHeE1J UYYAn1uzMBoFvUTB1uMK0wdD1IMnS3tX =p/F4 -----END PGP SIGNATURE----- -------------- next part -------------- A non-text attachment was scrubbed... Name: q9312PcapWS.jpg Type: image/jpeg Size: 153860 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/73be2660/attachment-0001.jpg From stkn at freeswitch.org Thu Jan 29 03:02:40 2009 From: stkn at freeswitch.org (Stefan Knoblich) Date: Thu, 29 Jan 2009 12:02:40 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <49818AE9.7030605@ewetel.de> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> Message-ID: <200901291202.41144.stkn@freeswitch.org> Am Thursday 29 January 2009 schrieb Helmut Kuper: > Hello, > > > a further update: > > The code is implemented and seems to work as it should. I still can do > outgoing calls when q931ToPcap is active :) So it shouldn't be too wrong > what I did in the openzap code. > The pcap File is generated in FS's log dir either with default filename > or with optional given filename. Existing files of same name will be > overwritten without a warning. > > I attached a screenshot of such a pcap file viewed in wireshark. > Currently I work on a little linux cli perl script, which displays the > decoded Q931 packets on a CLI as showed in my yesterday mail. > > One thing I couldn't solve is, how to get the span_id when FS receives > or send a Q931 packet? > > In ozmod_isdn functions "static int zap_isdn_921_23" (Receive) and > "static int q931_rx_32" (Send) I'm converting *pvt to zap_span_t like this: > > zap_span_t *span = (zap_span_t *) pvt; > > Then I try to access span_id by span->span_id, but it seems it's allways > zero. > > > Any ideas? > > regards > helmut > > Hi, --- a/src/ozmod/ozmod_isdn/ozmod_isdn.c +++ b/src/ozmod/ozmod_isdn/ozmod_isdn.c @@ -1970,7 +1993,7 @@ static ZIO_SIG_CONFIGURE_FUNCTION(isdn_configure_span) span, &isdn_data->q931); - Q921SetLogCB(&isdn_data->q921, &zap_isdn_q921_log, isdn_data); + Q921SetLogCB(&isdn_data->q921, &zap_isdn_q921_log, span); Q921SetLogLevel(&isdn_data->q921, (Q921LogLevel_t)q921loglevel); Q931InitTrunk(&isdn_data->q931, @@ -1983,7 +2006,7 @@ static ZIO_SIG_CONFIGURE_FUNCTION(isdn_configure_span) &isdn_data->q921, span); - Q931SetLogCB(&isdn_data->q931, &zap_isdn_q931_log, isdn_data); + Q931SetLogCB(&isdn_data->q931, &zap_isdn_q931_log, span); Q931SetLogLevel(&isdn_data->q931, (Q931LogLevel_t)q931loglevel); /* Register new event hander CB */ -- Stefan Knoblich Web: http://stkn.techmage.de/ http://oss.axsentis.de/people/stkn/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net From Claudio.Cavalera at italtel.it Thu Jan 29 03:40:52 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 29 Jan 2009 12:40:52 +0100 Subject: [Freeswitch-users] is there an event for SIP MESSAGE? In-Reply-To: <7E8E9F0C-0356-4289-BBF9-EE654C39526D@jerris.com> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > I am sorry to see that your e-mail is confidential. If it > were not, I would have responded to tell you that there is already an > api command > to send chat messages in a protocol agnostic way (see > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_chat) > . In the future if you can send non-confidential emails to the list > it will greatly assist us in being able to respond to your questions. > > Mike Mike thanks for the hint, I've tested this api and works well for sip messages! It's an effective alternative to the sendevent SEND_MESSAGE api. As I sad before I can't do much to control my mail client because confidentially footer are put in my exchange server. If this is a problem I can subscribe to the list with an unofficial mail account. Ciao, Claudio 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. ----------------------------------------------------------------------------------------------------- From cstomi.levlist at gmail.com Thu Jan 29 05:20:31 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Thu, 29 Jan 2009 14:20:31 +0100 Subject: [Freeswitch-users] record session in fifo In-Reply-To: <191c3a030901260847j660bb791w27657a08bdc232d0@mail.gmail.com> References: <4979F41F.70605@gmail.com> <191c3a030901231045m784f0cd7s6994b763d02b3e26@mail.gmail.com> <497D8782.2070603@gmail.com> <497DE0C9.5090005@gmail.com> <191c3a030901260847j660bb791w27657a08bdc232d0@mail.gmail.com> Message-ID: <4981AD1F.2040002@gmail.com> Hello, I'm sorry for still bothering you with this issue. but thanks to somewhat missunderstanding/misleading by us, it shows up that our customers need this feature for the complicated predictive calls too. There is an idea, could you let me know if it is achiveable. and if it's how? So we record sofia channel, but the media pause/resume is happening on loopback channel. the idea is to have a switch_set_flag_recursive function, which set the flag not only to the current channel, but to all connected channels. we need to push CS_PAUSE_MEDIA flag to the sofia channel. Thanks in advance, Tamas Here is the callflow again: 1, originate a loopback channel via event socket 2, loopback-b channel is hunting the dialplan, wich decide routing, caller_id, the need for recordings and so forth, and bridge a sofia call 3. the record_session is running on the sofia channel with bridge_pre_execute magic vars 4 loopback-a channel is pushed into the fifo 5 a script get the fifo::info via event socket 6 originate a call to the consumer with the proper strategy with &fifo out application 7 sofia channel is bridged to the consumer 8 loopback channels die Anthony Minessale ?rta: > yes some code was missing for some reason, try again > > > On Mon, Jan 26, 2009 at 10:11 AM, Tamas Cseke wrote: > > >> Hello, >> >> I tested with the attached patch. >> It is working fine in a normal case. >> >> I have only problems with the automatic calls, because in this case the >> loopback channel is in the fifo, but the record_session is running on the >> sofia channel. >> Maybe it could be sort out with putting the bug pause/resume functions into >> api function, what I should turn on and off on demand? >> Anyway, I quess this is a bit extreme circumstance, and it isn't so >> important to us now. >> >> Thanks, >> Tamas >> >> Tamas Cseke ?rta: >> >> Hello, >> >>> Thank you your help. >>> >>> I tested with r11489, but moh is still recorded in fifo. >>> >>> I quess you I should test the CF_PAUSE_BUGS in r11466. But I didn't find >>> where you check this flag. >>> Is it maybe possible you forget to commit something? >>> >>> Thanks, >>> Tamas >>> >>> >>> I didn't find where you >>> Anthony Minessale ?rta: >>> >>> >>> >>>> please test latest trunk. >>>> Patch added to pause media bugs while not in a bridge which should pause >>>> recordings and cut out the moh. >>>> >>>> >>>> On Fri, Jan 23, 2009 at 10:45 AM, Tamas Cseke >>> >>>>> wrote: >>>>> >>>> >>>> >>>>> Hello, >>>>> >>>>> we would like to distribute calls with fifo and record these sessions >>>>> but we'd like to skip the recording while the caller is waiting. >>>>> (we don't need to record the hold music, just the speech with the fifo >>>>> consumer.) >>>>> >>>>> I tried >>>>> >>>>> >>>>> >>>>> >>>>> but it doesn't work because the channel is answered immediately when the >>>>> caller is pushed into the fifo. >>>>> (I don't know if there exists any other channel flag that could be use >>>>> here) >>>>> >>>>> I also tried fifo_record_template. >>>>> but it records the session from the point of view of the consumer's >>>>> session, and after the bridge the recording is stopped. >>>>> we would like to record the whole session into a single file even after >>>>> calltransfers >>>>> >>>>> moreover we'd like to use some kind of predcitive dialing >>>>> which >>>>> 1, originate a loopback channel via event socket >>>>> 2, loopback-b channel is hunting the dialplan, wich decide routing, >>>>> caller_id, the need for recordings and so forth, and bridge a sofia call >>>>> 3. the record_session is running on the sofia channel with >>>>> bridge_pre_execute magic vars >>>>> 4 loopback-a channel is pushed into the fifo >>>>> 5 a script get the fifo::info via event socket >>>>> 6 originate a call to the consumer with the proper strategy with &fifo >>>>> out application >>>>> 7 sofia channel is bridged to the consumer >>>>> 8 loopback channels die >>>>> >>>>> after transfers everything is recorded into one file. >>>>> but the problem here is again the unwanted recording in the fifo while >>>>> the caller is waiting >>>>> >>>>> Could you please advise me any solution, if there is? >>>>> >>>>> >>>>> Thank you, >>>>> Tamas >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Thu Jan 29 05:49:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Jan 2009 07:49:06 -0600 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <200901291202.41144.stkn@freeswitch.org> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> Message-ID: <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> Would you like to check in the patch into tree now? We can give you commit access and we call all work on it together. On Thu, Jan 29, 2009 at 5:02 AM, Stefan Knoblich wrote: > Am Thursday 29 January 2009 schrieb Helmut Kuper: > > Hello, > > > > > > a further update: > > > > The code is implemented and seems to work as it should. I still can do > > outgoing calls when q931ToPcap is active :) So it shouldn't be too wrong > > what I did in the openzap code. > > The pcap File is generated in FS's log dir either with default filename > > or with optional given filename. Existing files of same name will be > > overwritten without a warning. > > > > I attached a screenshot of such a pcap file viewed in wireshark. > > Currently I work on a little linux cli perl script, which displays the > > decoded Q931 packets on a CLI as showed in my yesterday mail. > > > > One thing I couldn't solve is, how to get the span_id when FS receives > > or send a Q931 packet? > > > > In ozmod_isdn functions "static int zap_isdn_921_23" (Receive) and > > "static int q931_rx_32" (Send) I'm converting *pvt to zap_span_t like > this: > > > > zap_span_t *span = (zap_span_t *) pvt; > > > > Then I try to access span_id by span->span_id, but it seems it's allways > > zero. > > > > > > Any ideas? > > > > regards > > helmut > > > > > > Hi, > > --- a/src/ozmod/ozmod_isdn/ozmod_isdn.c > +++ b/src/ozmod/ozmod_isdn/ozmod_isdn.c > @@ -1970,7 +1993,7 @@ static > ZIO_SIG_CONFIGURE_FUNCTION(isdn_configure_span) > span, > &isdn_data->q931); > > - Q921SetLogCB(&isdn_data->q921, &zap_isdn_q921_log, isdn_data); > + Q921SetLogCB(&isdn_data->q921, &zap_isdn_q921_log, span); > Q921SetLogLevel(&isdn_data->q921, (Q921LogLevel_t)q921loglevel); > > Q931InitTrunk(&isdn_data->q931, > > @@ -1983,7 +2006,7 @@ static > ZIO_SIG_CONFIGURE_FUNCTION(isdn_configure_span) > &isdn_data->q921, > span); > > - Q931SetLogCB(&isdn_data->q931, &zap_isdn_q931_log, isdn_data); > + Q931SetLogCB(&isdn_data->q931, &zap_isdn_q931_log, span); > Q931SetLogLevel(&isdn_data->q931, (Q931LogLevel_t)q931loglevel); > > /* Register new event hander CB */ > > > -- > Stefan Knoblich > > Web: http://stkn.techmage.de/ http://oss.axsentis.de/people/stkn/ > Email: stkn at freeswitch.org > IRC: #freeswitch-de @ irc.freenode.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/9a17185f/attachment.html From anthony.minessale at gmail.com Thu Jan 29 05:59:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Jan 2009 07:59:55 -0600 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <45C07A45-AA15-4AC2-8ED0-73F4485CF190@freeswitch.org> References: <1233196879.4757.54.camel@dev03.cal.highergear.com> <45C07A45-AA15-4AC2-8ED0-73F4485CF190@freeswitch.org> Message-ID: <191c3a030901290559l69c5a2b3t2905fbbf3467b764@mail.gmail.com> There is a big chance some of your problems can come from your system copy of python. This is one of the reasons we were against using system libs because you need python built a certain way to work right with embedded code. If you really want to get to the bottom of it you may want to explore installing a non standard copy of python the way we used to do and compare results. On Wed, Jan 28, 2009 at 9:16 PM, Michael S Collins wrote: > > > Sent from my iPhone > > On Jan 28, 2009, at 6:41 PM, Brian Deacon > wrote: > > > Greetings, > > > > So I see from nabble and on the wiki that others have had problems > > with > > sqlalchemy and the hangup hook, but I appear to be having a more basic > > problem. > > > > I'm running FS 1.0.2, python 2.4.3, and SqlAlchemy 0.4.7 > > > > Here's an extension in my public.xml that works: > > > > > expression="^6468715186$"> > > > data="group_confirm_file=voicemail/8000/vm-hello.wav" /> > > > > > > > > > data="${destination_number_python}" /> > > > > > > > > With this silly DummyRouteHandler.py: > > import os > > from freeswitch import * > > #import sqlalchemy > > > > def fsapi(session, stream, env, args): > > stream.write('sofia/internal/1000%10.48.5.207') > > > > (Note the commented out sqlalchemy import. Uncommenting the import > > gets > > me an "error importing module" or "error reloading module" in the > > fs_cli > > output) > > > > Oooh, and just now trying to reproduce I got my first crash. > > > > Thoughts? > > > > Use Lua? ;) > Seriously, Lua is a breeze to learn and is more stable than Python. > Python is *not* designed to be embedded however Lua is designed > precisely for that reason. Think WoW. > > Other than that I believe that your experiences with Python might help > the devs uncover more issues. Please visit the "reporting bugs" page > on the wiki and start collecting the information that will help them > solve this one. > > -MC > > > Brian > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/fbbff9a5/attachment-0001.html From helmut.kuper at ewetel.de Thu Jan 29 06:10:31 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 29 Jan 2009 15:10:31 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <200901291202.41144.stkn@freeswitch.org> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> Message-ID: <4981B8D7.20809@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Stefan, I'm not sure what you want to tell me with the diff. Maybe I should hook into the CallBack log function to get the span_id? After your hint I looked into FS's concole log and found that even the Q931 callback log function displays span_id 0 I think this line: zap_log("Span", "Q.931", span->span_id, (int)level, "%s", msg); produces this line: 2009-01-29 11:16:20 [DEBUG] Span:0 Q.931() Receiving message from Layer4 (size: 212, type: 69) in FS console. There span_id is zero despite the fact that span_id in openzap.conf.xml is "1" openzap seems to set the span_id during zap_span_create(). The first span gets the span_id 1. Unfortuantely neither FS console nor my code displays the span id as expected. I would like to have it, because I want to put it into pcap file to give a hint in wireshark from which span the traffic is. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkmBuNcACgkQ4tZeNddg3dw5ugCgl2E1E/wiaGrebuh8G0WM9+gr iMoAoI0DmmkIC2B/CwxNiZQKYB7EmcDP =jDTx -----END PGP SIGNATURE----- From helmut.kuper at ewetel.de Thu Jan 29 06:10:46 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 29 Jan 2009 15:10:46 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> Message-ID: <4981B8E6.9070708@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Anthony, yes, I think I can upload it. Can you send me the datails please? regards Helmut Am 29.01.2009 14:49, schrieb Anthony Minessale: > Would you like to check in the patch into tree now? > We can give you commit access and we call all work on it together. > > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkmBuOUACgkQ4tZeNddg3dwkHQCgpxFkyHGJJO6Pii0zI1C3jx26 RdUAnRi5g0cjCzeFQ+0pwC5Zz2kKE93E =YpxO -----END PGP SIGNATURE----- From curly2009 at gmx.de Thu Jan 29 06:42:52 2009 From: curly2009 at gmx.de (curly2009 at gmx.de) Date: Thu, 29 Jan 2009 15:42:52 +0100 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: <87f2f3b90901280939x1153a33fvafb40383809f4eba@mail.gmail.com> References: <20090127164104.207050@gmx.net> <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> <20090128132438.309550@gmx.net> <191c3a030901280607r706b4a8cp4cc586fc62cb5086@mail.gmail.com> <20090128160102.126250@gmx.net> <20090128162314.243640@gmx.net> <87f2f3b90901280939x1153a33fvafb40383809f4eba@mail.gmail.com> Message-ID: <20090129144252.138670@gmx.net> I tried with the latest FreeSwitch Trunk and it's work now. But now two snom phones can do only inbound calls. when I do a outbound call I get this error on the console: 2009-01-29 15:26:43 [DEBUG] sofia.c:3785 sofia_handle_sip_i_invite() IP 10.xx.xx.xx Rejected by acl "domains". Falling back to Digest auth. 2009-01-29 15:26:43 [DEBUG] sofia.c:3785 sofia_handle_sip_i_invite() IP 10.xx.xx.xx Rejected by acl "domains". Falling back to Digest auth. 2009-01-29 15:26:43 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/1001 at 10.xx.xx.xx [ac2988cf-008c-4bd2-9aea-7de3d328d7ff] 2009-01-29 15:26:43 [DEBUG] sofia.c:4311 sofia_handle_sip_i_invite() Setting NAT mode based on rfc1918 2009-01-29 15:26:43 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/internal/1001 at 10.xx.xx.xx entering state [received] 2009-01-29 15:26:43 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 866376124 866376124 IN IP4 10.xx.xx.xx s=call c=IN IP4 10.xx.xx.xx t=0 0 m=audio 54874 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:mbfUCh51dy9DGVqPTcOaY4IYILRQfwzVT7zyIo7j a=ptime:20 2009-01-29 15:26:43 [ERR] sofia_glue.c:2367 sofia_glue_negotiate_sdp() a=crypto in RTP/AVP, refer to rfc3711 2009-01-29 15:26:43 [NOTICE] sofia.c:2741 sofia_handle_sip_i_state() Hangup sofia/internal/1001 at 10.xx.xx.xx [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-01-29 15:26:43 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/internal/1001 at 10.xx.xx.xx [KILL] 2009-01-29 15:26:43 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 10.xx.xx.xx [BREAK] 2009-01-29 15:26:43 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/1001 at 10.xx.xx.xx) Running State Change CS_HANGUP 2009-01-29 15:26:43 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/1001 at 10.xx.xx.xx) State HANGUP 2009-01-29 15:26:43 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/1001 at 10.xx.xx.xx hanging up, cause: INCOMPATIBLE_DESTINATION 2009-01-29 15:26:43 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 488 2009-01-29 15:26:43 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1001 at 10.xx.xx.xx Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2009-01-29 15:26:43 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/1001 at 10.xx.xx.xx) State HANGUP going to sleep 2009-01-29 15:26:43 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 2 (sofia/internal/1001 at 10.xx.xx.xx) Locked, Waiting on external entities 2009-01-29 15:26:43 [NOTICE] switch_core_session.c:957 xswitch_core_session_thread() Session 2 (sofia/internal/1001 at 10.xx.xx.xx) Ended 2009-01-29 15:26:43 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/internal/1001 at 10.xx.xx.xx [CS_HANGUP] > I just added a small section to the reporting bugs page on the wiki. > It specifically talks about this subject: > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Upgrading_From_Older_Revisions > > In simple terms, you are best to copy your configs to a safe location, > then delete your /usr/src/freeswitch directory, then delete your fs > source directory, and then do a fresh checkout & install. Personally, > on Linux I like doing the "quick and dirty install": > http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install > > After your fresh install is complete, go back and make your > customizations, like editing the modules.conf and modules.conf.xml, > installing openzap, etc. Then test and report back. > -MC > > On Wed, Jan 28, 2009 at 8:23 AM, wrote: > > no I don't have try it. > > > > I have to execute "make current" in the Freeswitch source directory, or > something else? > > > > -------- Original-Nachricht -------- > >> Datum: Wed, 28 Jan 2009 10:11:36 -0600 > >> Von: Brian West > >> An: freeswitch-users at lists.freeswitch.org > >> Betreff: Re: [Freeswitch-users] Can\'t dial over openzap > > > >> Have you tried with the latest FreeSWITCH Trunk? > >> > >> /b > >> > >> On Jan 28, 2009, at 10:01 AM, curly2009 at gmx.de wrote: > >> > >> > It's ISDN PRI. Digium Wildcard TE110P > >> > > >> > I don't edit tones.conf. I leave this to the defaults. > >> > > >> > I using > >> > freeswitch-1.0.1 > >> > zaptel-1.4.12 > >> > > >> > I posted the whole log of the call in pastebin.freeswitch.org. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > -- > > NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL > > f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger From mike at jerris.com Thu Jan 29 07:19:00 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Jan 2009 10:19:00 -0500 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: <20090129144252.138670@gmx.net> References: <20090127164104.207050@gmx.net> <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> <20090128132438.309550@gmx.net> <191c3a030901280607r706b4a8cp4cc586fc62cb5086@mail.gmail.com> <20090128160102.126250@gmx.net> <20090128162314.243640@gmx.net> <87f2f3b90901280939x1153a33fvafb40383809f4eba@mail.gmail.com> <20090129144252.138670@gmx.net> Message-ID: Snom filed a bug with us that we should not accept a=crypto in the RTP/ AVP as it is an rfc violation but left their defaults to still send that. Please file a complaint with snom that their default are not right. There is a setting to change it but I can not recall what it is. Mike On Jan 29, 2009, at 9:42 AM, curly2009 at gmx.de wrote: > 2009-01-29 15:26:43 [ERR] sofia_glue.c:2367 > sofia_glue_negotiate_sdp() a=crypto in RTP/AVP, refer to rfc3711 From pmhshz at gmail.com Thu Jan 29 07:42:03 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 29 Jan 2009 07:42:03 -0800 (PST) Subject: [Freeswitch-users] ATA-answering machine question/recommendation In-Reply-To: <87f2f3b90901271226n43d3c1c7lbb2edde19ead9642@mail.gmail.com> References: <207e7a5e0901210653t2ffa1dcdka0b4e7015119ce45@mail.gmail.com> <1461348542-1232566409-cardhu_decombobulator_blackberry.rim.net-569820458-@bxe161.bisx.prod.on.blackberry> <21640813.post@talk.nabble.com> <87f2f3b90901260203xd447883xe373244207c2fc45@mail.gmail.com> <21680924.post@talk.nabble.com> <87f2f3b90901271226n43d3c1c7lbb2edde19ead9642@mail.gmail.com> Message-ID: <21729361.post@talk.nabble.com> is there any configuration required in profile or somewhere else... or is it bug... thanks msp Michael Collins-11 wrote: > > I'm not too sure about this one. I see it calling mod_vmd and > complaining about needing media. Let's see what Eric Des Courtis has > to say on this one. Hold tight please... > > -MC > > On Mon, Jan 26, 2009 at 11:40 PM, shehzad p wrote: >> >> Hi Michael, >> >> My dial plan is: >> ======================== >> >> >> >> >> > data="sofia/external/1111 at xxx.xxx.xxx.x"/> >> >> >> >> >> ==================================== >> When call come to xxx.xxx.xxx.x system, It answer the call and only wait >> for >> 20 seconds (NO playback only wait) and hangup. >> Please find debug trace for above dialplan on >> http://pastebin.freeswitch.org/6919 >> >> >> Again I also tested dialplan exact same as shown in the wiki page >> http://wiki.freeswitch.org/wiki/Mod_vmd#From_Dialplan. >> But there it is not working as needed. >> >> >> Thanks for your response... >> msp >> >> >> >> Michael Collins-11 wrote: >>> >>> Did you ever post your dialplan and a debug trace of a call to the >>> pastebin? If not, please do so and we will check it out. >>> -MC >>> >>> On Sat, Jan 24, 2009 at 5:47 AM, shehzad p wrote: >>>> >>>> Hi all, >>>> >>>> On my existing Freeswitch 1.0.2, I installed and configured mod_vmd as >>>> below: >>>> make mod_vmd-install >>>> >>>> Then configured it as on wiki page: >>>> http://wiki.freeswitch.org/wiki/Mod_vmd >>>> >>>> >>>> After that my dialplan terminates call to another system, where it is >>>> just >>>> answered and wait for some time there. >>>> So that there should be a variable called vmd_detect must be created as >>>> shown in dialplan >>>> http://wiki.freeswitch.org/wiki/Mod_vmd#From_Dialplan. >>>> >>>> But eventhough no variable named 'vmd_detect' is created after that.!!! >>>> Is there i am missing something? Is there another way of using mod_vmd? >>>> >>>> Thanks in advance. >>>> msp >>>> >>>> >>>> >>>> >>>> >>>> >>>> Lucas Cornelisse wrote: >>>>> >>>>> Hi Jonathan, >>>>> >>>>> Mod_vmd (voicemail detection) should do the trick. >>>>> >>>>> Just search the wiki for mod_vmd, there are a number of ways of using >>>>> it. >>>>> >>>>> >>>>> Sent from my BlackBerry device on the Rogers Wireless Network >>>>> >>>>> -----Original Message----- >>>>> From: jonathan augenstine >>>>> >>>>> Date: Wed, 21 Jan 2009 06:53:41 >>>>> To: >>>>> Subject: [Freeswitch-users] ATA-answering machine >>>>> question/recommendation >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/ATA-answering-machine-question-recommendation-tp21584795p21640813.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/ATA-answering-machine-question-recommendation-tp21584795p21680924.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/ATA-answering-machine-question-recommendation-tp21584795p21729361.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Jan 29 08:01:32 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 29 Jan 2009 10:01:32 -0600 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <4981B8E6.9070708@ewetel.de> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> Message-ID: <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> How do you want to do this? Do you currently have an SVN account? Or do you wanna do this via Jira? /b On Jan 29, 2009, at 8:10 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi Anthony, > > yes, I think I can upload it. Can you send me the datails please? > > regards > Helmut > > Am 29.01.2009 14:49, schrieb Anthony Minessale: >> Would you like to check in the patch into tree now? >> We can give you commit access and we call all work on it together. >> >> > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkmBuOUACgkQ4tZeNddg3dwkHQCgpxFkyHGJJO6Pii0zI1C3jx26 > RdUAnRi5g0cjCzeFQ+0pwC5Zz2kKE93E > =YpxO > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Jan 29 08:03:41 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 29 Jan 2009 10:03:41 -0600 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: References: <20090127164104.207050@gmx.net> <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> <20090128132438.309550@gmx.net> <191c3a030901280607r706b4a8cp4cc586fc62cb5086@mail.gmail.com> <20090128160102.126250@gmx.net> <20090128162314.243640@gmx.net> <87f2f3b90901280939x1153a33fvafb40383809f4eba@mail.gmail.com> <20090129144252.138670@gmx.net> Message-ID: <88DBFACC-6A6A-4D2B-AE78-7FA0A8B4E1C7@freeswitch.org> You go to the identity under "RTP/SAVP:" and change it to optional /b On Jan 29, 2009, at 9:19 AM, Michael Jerris wrote: > Snom filed a bug with us that we should not accept a=crypto in the > RTP/ > AVP as it is an rfc violation but left their defaults to still send > that. Please file a complaint with snom that their default are not > right. There is a setting to change it but I can not recall what it > is. > > Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/a8a9f756/attachment-0001.html From intralanman at freeswitch.org Thu Jan 29 08:06:16 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 29 Jan 2009 11:06:16 -0500 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: References: <7160CD9F-F06E-4964-943A-F4B0C12150CC@jerris.com> Message-ID: <4981D3F8.2030207@freeswitch.org> Boris Lansky wrote: > Sorry for the stupid question but in what configuration file should I add such line "sofia/foo/user at that.domain ;fs_path=sip:proxy.this.domain" ? > > > appology accepted... look in the dialplan, there should be plenty of documentations on using the dialplan on our wiki... wiki.freeswitch.org... look for sofia syntax too on the mod_sofia page -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/b04d6b2f/attachment.html From pmhshz at gmail.com Thu Jan 29 08:06:58 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 29 Jan 2009 08:06:58 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> Message-ID: <21729863.post@talk.nabble.com> Hi Anthony, I found interesting result while testing Freeswitch, and it might be cause of freezing out of freeswitch, I updated my system (as you told) with latest stable version Freeswitch 1.0.2 First of all I set sps to 100, Then I sends call approximately 100 per seconds, Freeswitch works fine and handles all the calls very well. After that I send 130 calls per seconds, and magic happen now, Freeswitch handles first 100 calls only. all the preceding calls were failed (even not appeared in freeswitch console why?) When I put ngrep trace, System responds with 503 Maximum Calls In Progress. (as below) ########################################################### # U FSFSFSFSFS -> GWGWGWGWGW SIP/2.0 503 Maximum Calls In Progress. Via: SIP/2.0/UDP GWGWGWGWGW;branch=z9hG4bK53eafabb;rport=5060. From: "99999" ;tag=as2e10c170. To: ;tag=K3jSUFrDHpmmB. Call-ID: 0feb229e58afe7be17110deb361dc234 at GWGWGWGWGW. CSeq: 102 INVITE. Retry-After: 300. User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Length: 0. . ##################################################################### Now another issue to note down is that, After all above happened and active calls comes to zero, I just make a single call which also fails with response 503 - Maximum Calls In Progress. Is this intended behaviour, should I increase SPS to overcome this. or something like bug. Please let me know what should be the resolution for this. Thanks, msp Anthony Minessale-2 wrote: > > Also remember, > Actually completely uninstall and erase /usr/local/freeswitch and the > 1.0.1 > source tree and freshly install the new one. > If you try to upgrade on top of a release with trunk it will cause more > problems for you. > > > On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice wrote: > >> Upgrade to trunk... Many many issues have been resolved since 1.0.1 was >> the >> current release >> >> >> > From: shehzad p >> > Reply-To: >> > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) >> > To: >> > Subject: [Freeswitch-users] Freeswitch freezes on increasing call >> traffic >> > >> > >> > Hi all, >> > >> > Yesterday my Freeswitch server faced a problem when call traffic >> increased >> > to more than 100. >> > >> > When I start Freeswitch, it works fine and then after some time >> > (approximately 15 to 20 minutes) it stops functioning (means no call >> is >> > being processed, no CLI command is working and it just freezes) until I >> > restart the freeswitch. >> > >> > I am using Freeswitch 1.0.1. >> > Debug (gdb) trace as on wiki page >> > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.sh is >> attached >> > http://www.nabble.com/file/p21701744/fs_debgu.txt fs_debgu.txt >> > -- >> > View this message in context: >> > >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 >> > p21701744.html >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21729863.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Jan 29 08:15:51 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 29 Jan 2009 10:15:51 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21729863.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> Message-ID: <8A884633-44E7-468F-B507-7205FFF2C33C@freeswitch.org> You might also want to set max-proceeding on the sofia profile. /b On Jan 29, 2009, at 10:06 AM, shehzad p wrote: > > When I put ngrep trace, System responds with 503 Maximum Calls In > Progress. > (as below) From anthony.minessale at gmail.com Thu Jan 29 08:16:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Jan 2009 10:16:30 -0600 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <4981B8D7.20809@ewetel.de> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <4981B8D7.20809@ewetel.de> Message-ID: <191c3a030901290816q7ae965f1qa1f8758724e53c19@mail.gmail.com> I think his patch was setting the span itself as the private data to the callback rather than the private_data which was currently being set. On Thu, Jan 29, 2009 at 8:10 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi Stefan, > > I'm not sure what you want to tell me with the diff. Maybe I should hook > into the CallBack log function to get the span_id? > > After your hint I looked into FS's concole log and found that even the > Q931 callback log function displays span_id 0 > > I think this line: > > zap_log("Span", "Q.931", span->span_id, (int)level, "%s", msg); > > > produces this line: > > 2009-01-29 11:16:20 [DEBUG] Span:0 Q.931() Receiving message from Layer4 > (size: 212, type: 69) > > in FS console. There span_id is zero despite the fact that span_id in > openzap.conf.xml is "1" > > openzap seems to set the span_id during zap_span_create(). The first > span gets the span_id 1. > > Unfortuantely neither FS console nor my code displays the span id as > expected. > > I would like to have it, because I want to put it into pcap file to give > a hint in wireshark from which span the traffic is. > > > regards > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkmBuNcACgkQ4tZeNddg3dw5ugCgl2E1E/wiaGrebuh8G0WM9+gr > iMoAoI0DmmkIC2B/CwxNiZQKYB7EmcDP > =jDTx > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/1bb0e388/attachment.html From helmut.kuper at ewetel.de Thu Jan 29 08:19:04 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 29 Jan 2009 17:19:04 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> Message-ID: <4981D6F8.4060409@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Brian, no, I have no FS SVN account and I have no experience with jira and only a few with svn. So I'm looking for the easiest way ... I'm not sure if it is possible to do a svn commit in my openzap directory and svn then merges the code correctly with current openzap in trunk. regards Helmut Am 29.01.2009 17:01, schrieb Brian West: > How do you want to do this? Do you currently have an SVN account? Or > do you wanna do this via Jira? > > /b -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkmB1vgACgkQ4tZeNddg3dzWGQCeOB+JJLY/Ns6HaYRarMhF6EVj UCgAn1Cu1BDfuSuBquZMWL6cLLioNeMb =rDEe -----END PGP SIGNATURE----- From anthony.minessale at gmail.com Thu Jan 29 08:31:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Jan 2009 10:31:19 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21729863.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> Message-ID: <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> When you get it in that state what do you see when you execute fsctl sps is the sps a very low number? Did the sps drop by itself from the value you originally set it to? Are you using 32 bit? if so try all of these commands in your shell before starting FS ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 244 ulimit -l unlimited DO NOT put them in a script unless you source the script with . . myscript or they will be undone instantly when the script exits BTW, I said to try latest trunk not 1.0.2 We can only debug the development code at this point. On Thu, Jan 29, 2009 at 10:06 AM, shehzad p wrote: > > Hi Anthony, > > I found interesting result while testing Freeswitch, and it might be cause > of freezing out of freeswitch, > > I updated my system (as you told) with latest stable version Freeswitch > 1.0.2 > First of all I set sps to 100, > Then I sends call approximately 100 per seconds, Freeswitch works fine and > handles all the calls very well. > > After that I send 130 calls per seconds, and magic happen now, Freeswitch > handles first 100 calls only. > all the preceding calls were failed (even not appeared in freeswitch > console > why?) > > When I put ngrep trace, System responds with 503 Maximum Calls In Progress. > (as below) > ########################################################### > # > U FSFSFSFSFS -> GWGWGWGWGW > SIP/2.0 503 Maximum Calls In Progress. > Via: SIP/2.0/UDP GWGWGWGWGW;branch=z9hG4bK53eafabb;rport=5060. > From: "99999" ;tag=as2e10c170. > To: ;tag=K3jSUFrDHpmmB. > Call-ID: 0feb229e58afe7be17110deb361dc234 at GWGWGWGWGW. > CSeq: 102 INVITE. > Retry-After: 300. > User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, > REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Length: 0. > . > ##################################################################### > > > Now another issue to note down is that, > After all above happened and active calls comes to zero, > I just make a single call which also fails with response 503 - Maximum > Calls > In Progress. > > > Is this intended behaviour, should I increase SPS to overcome this. or > something like bug. > > Please let me know what should be the resolution for this. > > Thanks, > msp > > > > Anthony Minessale-2 wrote: > > > > Also remember, > > Actually completely uninstall and erase /usr/local/freeswitch and the > > 1.0.1 > > source tree and freshly install the new one. > > If you try to upgrade on top of a release with trunk it will cause more > > problems for you. > > > > > > On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice wrote: > > > >> Upgrade to trunk... Many many issues have been resolved since 1.0.1 was > >> the > >> current release > >> > >> > >> > From: shehzad p > >> > Reply-To: > >> > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) > >> > To: > >> > Subject: [Freeswitch-users] Freeswitch freezes on increasing call > >> traffic > >> > > >> > > >> > Hi all, > >> > > >> > Yesterday my Freeswitch server faced a problem when call traffic > >> increased > >> > to more than 100. > >> > > >> > When I start Freeswitch, it works fine and then after some time > >> > (approximately 15 to 20 minutes) it stops functioning (means no call > >> is > >> > being processed, no CLI command is working and it just freezes) until > I > >> > restart the freeswitch. > >> > > >> > I am using Freeswitch 1.0.1. > >> > Debug (gdb) trace as on wiki page > >> > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.sh is > >> attached > >> > http://www.nabble.com/file/p21701744/fs_debgu.txt fs_debgu.txt > >> > -- > >> > View this message in context: > >> > > >> > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 > >> > p21701744.html > >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at 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 at 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 at hotmail.com < > MSN%3Aanthony_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 < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21729863.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/973ff32d/attachment-0001.html From anthony.minessale at gmail.com Thu Jan 29 08:32:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Jan 2009 10:32:30 -0600 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <4981D6F8.4060409@ewetel.de> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> Message-ID: <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> Can you join irc and we can make you an account and give you the instructions? irc.freenode.net #freeswitch or just the java applet on our homepage. On Thu, Jan 29, 2009 at 10:19 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello Brian, > > no, I have no FS SVN account and I have no experience with jira and only > a few with svn. So I'm looking for the easiest way ... > I'm not sure if it is possible to do a svn commit in my openzap > directory and svn then merges the code correctly with current openzap in > trunk. > > > regards > Helmut > > Am 29.01.2009 17:01, schrieb Brian West: > > How do you want to do this? Do you currently have an SVN account? Or > > do you wanna do this via Jira? > > > > /b > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAkmB1vgACgkQ4tZeNddg3dzWGQCeOB+JJLY/Ns6HaYRarMhF6EVj > UCgAn1Cu1BDfuSuBquZMWL6cLLioNeMb > =rDEe > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/fbd463bd/attachment.html From anthony.minessale at gmail.com Thu Jan 29 08:44:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Jan 2009 10:44:11 -0600 Subject: [Freeswitch-users] record session in fifo In-Reply-To: <4981AD1F.2040002@gmail.com> References: <4979F41F.70605@gmail.com> <191c3a030901231045m784f0cd7s6994b763d02b3e26@mail.gmail.com> <497D8782.2070603@gmail.com> <497DE0C9.5090005@gmail.com> <191c3a030901260847j660bb791w27657a08bdc232d0@mail.gmail.com> <4981AD1F.2040002@gmail.com> Message-ID: <191c3a030901290844j51180ddej5e59372e6b971b9a@mail.gmail.com> I don't think there is a very elegant way to do this and even if I find one i can live with it would be at least 2k worth of work to add it. Why do you need the loopback channel at all? Can't you just originate the sofia channel directly instead of bridging it to loopback and adding 2 more channels per call? On Thu, Jan 29, 2009 at 7:20 AM, Tamas Cseke wrote: > Hello, > > I'm sorry for still bothering you with this issue. > but thanks to somewhat missunderstanding/misleading by us, it shows up that > our customers need this feature for the complicated > predictive calls too. > > There is an idea, could you let me know if it is achiveable. and if it's > how? > So we record sofia channel, but the media pause/resume is happening on > loopback channel. > the idea is to have a switch_set_flag_recursive function, which set the > flag not only to the current channel, but to all connected channels. > we need to push CS_PAUSE_MEDIA flag to the sofia channel. > > Thanks in advance, > Tamas > > Here is the callflow again: > 1, originate a loopback channel via event socket > 2, loopback-b channel is hunting the dialplan, wich decide routing, > caller_id, the need for recordings and so forth, and bridge a sofia call > 3. the record_session is running on the sofia channel with > bridge_pre_execute magic vars > 4 loopback-a channel is pushed into the fifo > 5 a script get the fifo::info via event socket > 6 originate a call to the consumer with the proper strategy with &fifo > out application > 7 sofia channel is bridged to the consumer > 8 loopback channels die > > > > Anthony Minessale ?rta: > > yes some code was missing for some reason, try again > > > > > > On Mon, Jan 26, 2009 at 10:11 AM, Tamas Cseke >wrote: > > > > > >> Hello, > >> > >> I tested with the attached patch. > >> It is working fine in a normal case. > >> > >> I have only problems with the automatic calls, because in this case the > >> loopback channel is in the fifo, but the record_session is running on > the > >> sofia channel. > >> Maybe it could be sort out with putting the bug pause/resume functions > into > >> api function, what I should turn on and off on demand? > >> Anyway, I quess this is a bit extreme circumstance, and it isn't so > >> important to us now. > >> > >> Thanks, > >> Tamas > >> > >> Tamas Cseke ?rta: > >> > >> Hello, > >> > >>> Thank you your help. > >>> > >>> I tested with r11489, but moh is still recorded in fifo. > >>> > >>> I quess you I should test the CF_PAUSE_BUGS in r11466. But I didn't > find > >>> where you check this flag. > >>> Is it maybe possible you forget to commit something? > >>> > >>> Thanks, > >>> Tamas > >>> > >>> > >>> I didn't find where you > >>> Anthony Minessale ?rta: > >>> > >>> > >>> > >>>> please test latest trunk. > >>>> Patch added to pause media bugs while not in a bridge which should > pause > >>>> recordings and cut out the moh. > >>>> > >>>> > >>>> On Fri, Jan 23, 2009 at 10:45 AM, Tamas Cseke < > cstomi.levlist at gmail.com > >>>> > >>>>> wrote: > >>>>> > >>>> > >>>> > >>>>> Hello, > >>>>> > >>>>> we would like to distribute calls with fifo and record these sessions > >>>>> but we'd like to skip the recording while the caller is waiting. > >>>>> (we don't need to record the hold music, just the speech with the > fifo > >>>>> consumer.) > >>>>> > >>>>> I tried > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> but it doesn't work because the channel is answered immediately when > the > >>>>> caller is pushed into the fifo. > >>>>> (I don't know if there exists any other channel flag that could be > use > >>>>> here) > >>>>> > >>>>> I also tried fifo_record_template. > >>>>> but it records the session from the point of view of the consumer's > >>>>> session, and after the bridge the recording is stopped. > >>>>> we would like to record the whole session into a single file even > after > >>>>> calltransfers > >>>>> > >>>>> moreover we'd like to use some kind of predcitive dialing > >>>>> which > >>>>> 1, originate a loopback channel via event socket > >>>>> 2, loopback-b channel is hunting the dialplan, wich decide routing, > >>>>> caller_id, the need for recordings and so forth, and bridge a sofia > call > >>>>> 3. the record_session is running on the sofia channel with > >>>>> bridge_pre_execute magic vars > >>>>> 4 loopback-a channel is pushed into the fifo > >>>>> 5 a script get the fifo::info via event socket > >>>>> 6 originate a call to the consumer with the proper strategy with > &fifo > >>>>> out application > >>>>> 7 sofia channel is bridged to the consumer > >>>>> 8 loopback channels die > >>>>> > >>>>> after transfers everything is recorded into one file. > >>>>> but the problem here is again the unwanted recording in the fifo > while > >>>>> the caller is waiting > >>>>> > >>>>> Could you please advise me any solution, if there is? > >>>>> > >>>>> > >>>>> Thank you, > >>>>> Tamas > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/53e19f87/attachment.html From curly2009 at gmx.de Thu Jan 29 08:50:37 2009 From: curly2009 at gmx.de (curly2009 at gmx.de) Date: Thu, 29 Jan 2009 17:50:37 +0100 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: <88DBFACC-6A6A-4D2B-AE78-7FA0A8B4E1C7@freeswitch.org> References: <20090127164104.207050@gmx.net> <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> <20090128132438.309550@gmx.net> <191c3a030901280607r706b4a8cp4cc586fc62cb5086@mail.gmail.com> <20090128160102.126250@gmx.net> <20090128162314.243640@gmx.net> <87f2f3b90901280939x1153a33fvafb40383809f4eba@mail.gmail.com> <20090129144252.138670@gmx.net> <88DBFACC-6A6A-4D2B-AE78-7FA0A8B4E1C7@freeswitch.org> Message-ID: <20090129165037.278750@gmx.net> Thanks... it works! > You go to the identity under "RTP/SAVP:" and change it to optional > > /b > > On Jan 29, 2009, at 9:19 AM, Michael Jerris wrote: > > > Snom filed a bug with us that we should not accept a=crypto in the > > RTP/ > > AVP as it is an rfc violation but left their defaults to still send > > that. Please file a complaint with snom that their default are not > > right. There is a setting to change it but I can not recall what it > > is. > > > > Mike > -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL f?r nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a From brian at freeswitch.org Thu Jan 29 08:55:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 29 Jan 2009 10:55:02 -0600 Subject: [Freeswitch-users] Can't dial over openzap In-Reply-To: <20090129165037.278750@gmx.net> References: <20090127164104.207050@gmx.net> <87f2f3b90901271105n4533d36bja9ff5f4eb0580f11@mail.gmail.com> <20090128132438.309550@gmx.net> <191c3a030901280607r706b4a8cp4cc586fc62cb5086@mail.gmail.com> <20090128160102.126250@gmx.net> <20090128162314.243640@gmx.net> <87f2f3b90901280939x1153a33fvafb40383809f4eba@mail.gmail.com> <20090129144252.138670@gmx.net> <88DBFACC-6A6A-4D2B-AE78-7FA0A8B4E1C7@freeswitch.org> <20090129165037.278750@gmx.net> Message-ID: <4F94EE9F-8498-4D84-B692-38F4BC56E217@freeswitch.org> Also email snom and ask them to make that the default setting. I have added this to the FAQ also. /b On Jan 29, 2009, at 10:50 AM, curly2009 at gmx.de wrote: > > Thanks... it works! > >> You go to the identity under "RTP/SAVP:" and change it to optional >> >> /b > From msc at freeswitch.org Thu Jan 29 09:48:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 29 Jan 2009 09:48:36 -0800 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: <4981D3F8.2030207@freeswitch.org> References: <7160CD9F-F06E-4964-943A-F4B0C12150CC@jerris.com> <4981D3F8.2030207@freeswitch.org> Message-ID: <87f2f3b90901290948w898fb68u89dd2ddd436b8838@mail.gmail.com> On Thu, Jan 29, 2009 at 8:06 AM, Raymond Chandler wrote: > Boris Lansky wrote: > > Sorry for the stupid question but in what configuration file should I add > such line "sofia/foo/user at that.domain;fs_path=sip:proxy.this.domain" ? http://wiki.freeswitch.org/wiki/Sofia#Specifying_SIP_Proxy_With_fs_path I believe that may indeed do what you want. Give it a try and report back. Also, please consider joining us on the IRC channel. More information here: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_IRC_Channel -MC > > > > appology accepted... look in the dialplan, there should be plenty of > documentations on using the dialplan on our wiki... wiki.freeswitch.org... > look for sofia syntax too on the mod_sofia page > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From bdeacon at highergear.com Thu Jan 29 10:01:56 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Thu, 29 Jan 2009 10:01:56 -0800 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <45C07A45-AA15-4AC2-8ED0-73F4485CF190@freeswitch.org> References: <1233196879.4757.54.camel@dev03.cal.highergear.com> <45C07A45-AA15-4AC2-8ED0-73F4485CF190@freeswitch.org> Message-ID: <1233252116.4757.60.camel@dev03.cal.highergear.com> On Wed, 2009-01-28 at 19:16 -0800, Michael S Collins wrote: > > > (Note the commented out sqlalchemy import. Uncommenting the import > > gets > > me an "error importing module" or "error reloading module" in the > > fs_cli > > output) > > > > Oooh, and just now trying to reproduce I got my first crash. > > > > Thoughts? > > > > Use Lua? ;) > Seriously, Lua is a breeze to learn and is more stable than Python. > Python is *not* designed to be embedded however Lua is designed > precisely for that reason. Think WoW. > > Other than that I believe that your experiences with Python might help > the devs uncover more issues. Please visit the "reporting bugs" page > on the wiki and start collecting the information that will help them > solve this one. [Insert Smedley-cursing here.] Hmm, I think an investment in Lua here is going to be a hard sell. Would you expect the JavaScript implementation to be as robust as the Lua implementation? (And then I'll need to eliminate my ignorance on what javascript can do with databases when it's not sandboxed by a browser.) I'll see if I can dig up a usable stack trace for the python problem. My gdb-fu is weak. Brian From asannucci at gmail.com Thu Jan 29 12:12:56 2009 From: asannucci at gmail.com (Andrea) Date: Thu, 29 Jan 2009 15:12:56 -0500 Subject: [Freeswitch-users] Users LANl - Users WAN Message-ID: <157D36A8124D4198B246B8B8027B68F1@quos> Hi, i came from asterisk and don't understand some things. I have to configure the internal LAN users in directory/default I have to configure the external WAN users in sip_profiles/external (all softphones are behind a router - NAT) I have to configure the gateway (Voip Providers in sip_profiles/internal) it's right? Thank you in advance - Andrea - From intralanman at freeswitch.org Thu Jan 29 12:26:48 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 29 Jan 2009 15:26:48 -0500 Subject: [Freeswitch-users] Users LANl - Users WAN In-Reply-To: <157D36A8124D4198B246B8B8027B68F1@quos> References: <157D36A8124D4198B246B8B8027B68F1@quos> Message-ID: <49821108.5040908@freeswitch.org> Andrea wrote: > Hi, > > i came from asterisk and don't understand some things. > > I have to configure the internal LAN users in directory/default > I have to configure the external WAN users in sip_profiles/external (all > softphones are behind a router - NAT) > no, all users are configured in the user directory > I have to configure the gateway (Voip Providers in sip_profiles/internal) > it's right? yes, you configure your gateways in the sip profile -Ray From bdeacon at highergear.com Thu Jan 29 12:31:06 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Thu, 29 Jan 2009 12:31:06 -0800 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <191c3a030901290559l69c5a2b3t2905fbbf3467b764@mail.gmail.com> References: <1233196879.4757.54.camel@dev03.cal.highergear.com> <45C07A45-AA15-4AC2-8ED0-73F4485CF190@freeswitch.org> <191c3a030901290559l69c5a2b3t2905fbbf3467b764@mail.gmail.com> Message-ID: <1233261066.4757.64.camel@dev03.cal.highergear.com> On Thu, 2009-01-29 at 07:59 -0600, Anthony Minessale wrote: > There is a big chance some of your problems can come from your system > copy of python. > This is one of the reasons we were against using system libs because > you need python built a certain way to > work right with embedded code. > > If you really want to get to the bottom of it you may want to explore > installing a non standard copy of python the way > we used to do and compare results. I'm in the middle of trying an upgrade to python 2.5 to see if that makes a difference. But trying a "non-standard" build would be worth exploring. Can you point me in the right direction? Brian From asannucci at gmail.com Thu Jan 29 12:48:50 2009 From: asannucci at gmail.com (Andrea) Date: Thu, 29 Jan 2009 15:48:50 -0500 Subject: [Freeswitch-users] Users LANl - Users WAN References: <157D36A8124D4198B246B8B8027B68F1@quos> <49821108.5040908@freeswitch.org> Message-ID: <09C39C8569514C94962E7E53A26590E7@quos> Thank you Ray Second question :) how can i resolve the NAT problem with WAN users (softphone or IPphone) behind a router - NAT? The freeswitch server work with all needed ports open I have this problem: LAN softphone call WAN softphone all ok WAN softhone call LAN softphone the line drop after 30-32 seconds WAN softphone call other WAN Softphone the line drop after 30-32 seconds LAN softphone call other LAN softphone all ok Any idea? - Andrea - From anthony.minessale at gmail.com Thu Jan 29 12:49:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Jan 2009 14:49:53 -0600 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <1233261066.4757.64.camel@dev03.cal.highergear.com> References: <1233196879.4757.54.camel@dev03.cal.highergear.com> <45C07A45-AA15-4AC2-8ED0-73F4485CF190@freeswitch.org> <191c3a030901290559l69c5a2b3t2905fbbf3467b764@mail.gmail.com> <1233261066.4757.64.camel@dev03.cal.highergear.com> Message-ID: <191c3a030901291249t3159f33bg875b90da74564289@mail.gmail.com> We used to build python 2.5.1 and install it into the freeswitch prefix we did ./configure --prefix=/usr/local/freeswitch --enable-threads CFLAGSFORSHARED="-fPIC" But the makefile no longer will find that one so....... you could move your python out of the way on your box and try installing this as your system python instead. ./configure --prefix=/your/real/python/prefix --enable-threads CFLAGSFORSHARED="-fPIC" if you are really brave you could delete the Makefile in mod_python and put back the old one that builds and installs custom python for you into the FS prefix http://fisheye.freeswitch.org/browse/~raw,r=6393/FreeSWITCH/src/mod/languages/mod_python/Makefile the important thing is the --enable-threads and CFLAGSFORSHARED="-fPIC" on whatever one you use. On Thu, Jan 29, 2009 at 2:31 PM, Brian Deacon wrote: > On Thu, 2009-01-29 at 07:59 -0600, Anthony Minessale wrote: > > There is a big chance some of your problems can come from your system > > copy of python. > > This is one of the reasons we were against using system libs because > > you need python built a certain way to > > work right with embedded code. > > > > If you really want to get to the bottom of it you may want to explore > > installing a non standard copy of python the way > > we used to do and compare results. > > I'm in the middle of trying an upgrade to python 2.5 to see if that > makes a difference. But trying a "non-standard" build would be worth > exploring. Can you point me in the right direction? > > Brian > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/665a6169/attachment.html From Daniell at airg.com Thu Jan 29 11:05:35 2009 From: Daniell at airg.com (Daniel Liang) Date: Thu, 29 Jan 2009 11:05:35 -0800 Subject: [Freeswitch-users] How to break a playback with loops Message-ID: <0B02E756F603CC409EB553879B090CC80A1B6DC2@HPEXCHVS01.exchange.airg> Hi, I am trying to playing back a wav file a few times, and on receiving a dtmf event, it stops playing the file. The command I sent is: sendmsg call-command: execute execute-app-name: playback execute-app-arg: loops: 4 On receiving a dtmf event, I send a break command: sendmsg call-command: execute execute-app-name: break However, that only breaks one loop of the playback. And I have to press a dtmf for 4 times to stop the file. I tried sending 4 break commands all at once. But it didn't work. And even worse, my program stops receiving dtmfs anymore. My guess is that the first break stops the first loop, but before the second loop starts, the following 3 breaks stop something else that I don't know what they are. Any help is appreciated. Thanks. Daniel Liang -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/d1d1180f/attachment-0001.html From msc at freeswitch.org Thu Jan 29 13:34:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 29 Jan 2009 13:34:30 -0800 Subject: [Freeswitch-users] How to break a playback with loops In-Reply-To: <0B02E756F603CC409EB553879B090CC80A1B6DC2@HPEXCHVS01.exchange.airg> References: <0B02E756F603CC409EB553879B090CC80A1B6DC2@HPEXCHVS01.exchange.airg> Message-ID: <87f2f3b90901291334i27052568yc90a83d250cc1b76@mail.gmail.com> Could you please file a bug report on this? jira.freeswitch.org. Also, follow the bug reporting guidelines here: http://wiki.freeswitch.org/wiki/Reporting_Bugs Thanks, MC On Thu, Jan 29, 2009 at 11:05 AM, Daniel Liang wrote: > Hi, > > I am trying to playing back a wav file a few times, and on receiving a dtmf > event, it stops playing the file. The command I sent is: > > sendmsg > call-command: execute > execute-app-name: playback > execute-app-arg: > loops: 4 > > On receiving a dtmf event, I send a break command: > > sendmsg > call-command: execute > execute-app-name: break > > However, that only breaks one loop of the playback. And I have to press a > dtmf for 4 times to stop the file. > > I tried sending 4 break commands all at once. But it didn't work. And even > worse, my program stops receiving dtmfs anymore. My guess is that the first > break stops the first loop, but before the second loop starts, the following > 3 breaks stop something else that I don't know what they are. > > Any help is appreciated. > > Thanks. > Daniel Liang > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Thu Jan 29 13:37:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 29 Jan 2009 13:37:31 -0800 Subject: [Freeswitch-users] Users LANl - Users WAN In-Reply-To: <09C39C8569514C94962E7E53A26590E7@quos> References: <157D36A8124D4198B246B8B8027B68F1@quos> <49821108.5040908@freeswitch.org> <09C39C8569514C94962E7E53A26590E7@quos> Message-ID: <87f2f3b90901291337y47014debmf45e972232d2efd0@mail.gmail.com> On Thu, Jan 29, 2009 at 12:48 PM, Andrea wrote: > Thank you Ray > > Second question :) > > how can i resolve the NAT problem with WAN users (softphone or IPphone) > behind a router - NAT? > Have you had a chance to review the information here? http://wiki.freeswitch.org/wiki/NAT It might help you get over the hump. -MC > The freeswitch server work with all needed ports open > > I have this problem: > > LAN softphone call WAN softphone all ok > WAN softhone call LAN softphone the line drop after 30-32 seconds > WAN softphone call other WAN Softphone the line drop after 30-32 seconds > LAN softphone call other LAN softphone all ok > > Any idea? > > - Andrea - > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Thu Jan 29 13:54:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Jan 2009 15:54:25 -0600 Subject: [Freeswitch-users] How to break a playback with loops In-Reply-To: <87f2f3b90901291334i27052568yc90a83d250cc1b76@mail.gmail.com> References: <0B02E756F603CC409EB553879B090CC80A1B6DC2@HPEXCHVS01.exchange.airg> <87f2f3b90901291334i27052568yc90a83d250cc1b76@mail.gmail.com> Message-ID: <191c3a030901291354t5cc16a22h32b5defb755c1d3c@mail.gmail.com> The break function with the all app is better for this api break all This lets you still break even if the playback is in event lock I did add to trunk a patch so you can do it with the app you can now add "all" as the app args to break completely. On Thu, Jan 29, 2009 at 3:34 PM, Michael Collins wrote: > Could you please file a bug report on this? jira.freeswitch.org. Also, > follow the bug reporting guidelines here: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > Thanks, > MC > > On Thu, Jan 29, 2009 at 11:05 AM, Daniel Liang wrote: > > Hi, > > > > I am trying to playing back a wav file a few times, and on receiving a > dtmf > > event, it stops playing the file. The command I sent is: > > > > sendmsg > > call-command: execute > > execute-app-name: playback > > execute-app-arg: > > loops: 4 > > > > On receiving a dtmf event, I send a break command: > > > > sendmsg > > call-command: execute > > execute-app-name: break > > > > However, that only breaks one loop of the playback. And I have to press a > > dtmf for 4 times to stop the file. > > > > I tried sending 4 break commands all at once. But it didn't work. And > even > > worse, my program stops receiving dtmfs anymore. My guess is that the > first > > break stops the first loop, but before the second loop starts, the > following > > 3 breaks stop something else that I don't know what they are. > > > > Any help is appreciated. > > > > Thanks. > > Daniel Liang > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/ef1635aa/attachment.html From cstomi.levlist at gmail.com Thu Jan 29 14:15:22 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Thu, 29 Jan 2009 23:15:22 +0100 Subject: [Freeswitch-users] record session in fifo In-Reply-To: <191c3a030901290844j51180ddej5e59372e6b971b9a@mail.gmail.com> References: <4979F41F.70605@gmail.com> <191c3a030901231045m784f0cd7s6994b763d02b3e26@mail.gmail.com> <497D8782.2070603@gmail.com> <497DE0C9.5090005@gmail.com> <191c3a030901260847j660bb791w27657a08bdc232d0@mail.gmail.com> <4981AD1F.2040002@gmail.com> <191c3a030901290844j51180ddej5e59372e6b971b9a@mail.gmail.com> Message-ID: Hello, Thank you for the response. yes originating sofia channel directly could be an option. the only reason we use loopback is that our routing is in the dialplan (different proxies according to time and prefix matching). and the entity originating the call doesn't have to deal with them. so these are handled the same way as a "normal" manual call. otherwise I guess we have to duplicate this logic. so we didn't find elegant solution for this neither thus I asked maybe there is. Regards, Tamas On Thu, Jan 29, 2009 at 5:44 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > > I don't think there is a very elegant way to do this and even if I find one > i can live with it would be at least 2k worth of work to add it. > > Why do you need the loopback channel at all? Can't you just originate the > sofia channel directly instead of bridging it to loopback and adding 2 more > channels per call? > > > > > On Thu, Jan 29, 2009 at 7:20 AM, Tamas Cseke wrote: > >> Hello, >> >> I'm sorry for still bothering you with this issue. >> but thanks to somewhat missunderstanding/misleading by us, it shows up >> that our customers need this feature for the complicated >> predictive calls too. >> >> There is an idea, could you let me know if it is achiveable. and if it's >> how? >> So we record sofia channel, but the media pause/resume is happening on >> loopback channel. >> the idea is to have a switch_set_flag_recursive function, which set the >> flag not only to the current channel, but to all connected channels. >> we need to push CS_PAUSE_MEDIA flag to the sofia channel. >> >> Thanks in advance, >> Tamas >> >> Here is the callflow again: >> 1, originate a loopback channel via event socket >> 2, loopback-b channel is hunting the dialplan, wich decide routing, >> caller_id, the need for recordings and so forth, and bridge a sofia call >> 3. the record_session is running on the sofia channel with >> bridge_pre_execute magic vars >> 4 loopback-a channel is pushed into the fifo >> 5 a script get the fifo::info via event socket >> 6 originate a call to the consumer with the proper strategy with &fifo >> out application >> 7 sofia channel is bridged to the consumer >> 8 loopback channels die >> >> >> >> Anthony Minessale ?rta: >> > yes some code was missing for some reason, try again >> > >> > >> > On Mon, Jan 26, 2009 at 10:11 AM, Tamas Cseke > >wrote: >> > >> > >> >> Hello, >> >> >> >> I tested with the attached patch. >> >> It is working fine in a normal case. >> >> >> >> I have only problems with the automatic calls, because in this case the >> >> loopback channel is in the fifo, but the record_session is running on >> the >> >> sofia channel. >> >> Maybe it could be sort out with putting the bug pause/resume functions >> into >> >> api function, what I should turn on and off on demand? >> >> Anyway, I quess this is a bit extreme circumstance, and it isn't so >> >> important to us now. >> >> >> >> Thanks, >> >> Tamas >> >> >> >> Tamas Cseke ?rta: >> >> >> >> Hello, >> >> >> >>> Thank you your help. >> >>> >> >>> I tested with r11489, but moh is still recorded in fifo. >> >>> >> >>> I quess you I should test the CF_PAUSE_BUGS in r11466. But I didn't >> find >> >>> where you check this flag. >> >>> Is it maybe possible you forget to commit something? >> >>> >> >>> Thanks, >> >>> Tamas >> >>> >> >>> >> >>> I didn't find where you >> >>> Anthony Minessale ?rta: >> >>> >> >>> >> >>> >> >>>> please test latest trunk. >> >>>> Patch added to pause media bugs while not in a bridge which should >> pause >> >>>> recordings and cut out the moh. >> >>>> >> >>>> >> >>>> On Fri, Jan 23, 2009 at 10:45 AM, Tamas Cseke < >> cstomi.levlist at gmail.com >> >>>> >> >>>>> wrote: >> >>>>> >> >>>> >> >>>> >> >>>>> Hello, >> >>>>> >> >>>>> we would like to distribute calls with fifo and record these >> sessions >> >>>>> but we'd like to skip the recording while the caller is waiting. >> >>>>> (we don't need to record the hold music, just the speech with the >> fifo >> >>>>> consumer.) >> >>>>> >> >>>>> I tried >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> but it doesn't work because the channel is answered immediately when >> the >> >>>>> caller is pushed into the fifo. >> >>>>> (I don't know if there exists any other channel flag that could be >> use >> >>>>> here) >> >>>>> >> >>>>> I also tried fifo_record_template. >> >>>>> but it records the session from the point of view of the consumer's >> >>>>> session, and after the bridge the recording is stopped. >> >>>>> we would like to record the whole session into a single file even >> after >> >>>>> calltransfers >> >>>>> >> >>>>> moreover we'd like to use some kind of predcitive dialing >> >>>>> which >> >>>>> 1, originate a loopback channel via event socket >> >>>>> 2, loopback-b channel is hunting the dialplan, wich decide routing, >> >>>>> caller_id, the need for recordings and so forth, and bridge a sofia >> call >> >>>>> 3. the record_session is running on the sofia channel with >> >>>>> bridge_pre_execute magic vars >> >>>>> 4 loopback-a channel is pushed into the fifo >> >>>>> 5 a script get the fifo::info via event socket >> >>>>> 6 originate a call to the consumer with the proper strategy with >> &fifo >> >>>>> out application >> >>>>> 7 sofia channel is bridged to the consumer >> >>>>> 8 loopback channels die >> >>>>> >> >>>>> after transfers everything is recorded into one file. >> >>>>> but the problem here is again the unwanted recording in the fifo >> while >> >>>>> the caller is waiting >> >>>>> >> >>>>> Could you please advise me any solution, if there is? >> >>>>> >> >>>>> >> >>>>> Thank you, >> >>>>> Tamas >> >>>>> >> >>>>> >> >>>>> _______________________________________________ >> >>>>> Freeswitch-users mailing list >> >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at 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 at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/509766e5/attachment-0001.html From bdeacon at highergear.com Thu Jan 29 15:07:51 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Thu, 29 Jan 2009 15:07:51 -0800 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <191c3a030901291249t3159f33bg875b90da74564289@mail.gmail.com> References: <1233196879.4757.54.camel@dev03.cal.highergear.com> <45C07A45-AA15-4AC2-8ED0-73F4485CF190@freeswitch.org> <191c3a030901290559l69c5a2b3t2905fbbf3467b764@mail.gmail.com> <1233261066.4757.64.camel@dev03.cal.highergear.com> <191c3a030901291249t3159f33bg875b90da74564289@mail.gmail.com> Message-ID: <1233270471.4757.74.camel@dev03.cal.highergear.com> On Thu, 2009-01-29 at 14:49 -0600, Anthony Minessale wrote: > We used to build python 2.5.1 and install it into the freeswitch > prefix > > we did ./configure --prefix=/usr/local/freeswitch > --enable-threads CFLAGSFORSHARED="-fPIC" > > But the makefile no longer will find that one so....... > > you could move your python out of the way on your box and try > installing this as your system python instead. > > ./configure --prefix=/your/real/python/prefix > --enable-threads CFLAGSFORSHARED="-fPIC" > > if you are really brave you could delete the Makefile in mod_python > and put back the old one > that builds and installs custom python for you into the FS prefix > > http://fisheye.freeswitch.org/browse/~raw,r=6393/FreeSWITCH/src/mod/languages/mod_python/Makefile > > > the important thing is the --enable-threads and > CFLAGSFORSHARED="-fPIC" on whatever one you use. > So this is the error that gdb shows me, do you think this is the kind of thing that the "special" compile would clear up? 2009-01-29 15:57:12 [NOTICE] mod_python.c:107 eval_some_python() Invoking py module: callcontrol.routing.DummyRouteHandler 2009-01-29 15:57:12 [ERR] mod_python.c:121 eval_some_python() Error importing module Traceback (most recent call last): File "/usr/local/freeswitch/python/callcontrol/routing/DummyRouteHandler.py", line 19, in ? import sqlalchemy File "/usr/lib/python2.4/site-packages/sqlalchemy/__init__.py", line 8, in ? from sqlalchemy.types import \ File "/usr/lib/python2.4/site-packages/sqlalchemy/types.py", line 25, in ? import datetime as dt ImportError: /usr/lib/python2.4/lib-dynload/datetime.so: undefined symbol: _Py_ZeroStruct And I believe I understood the answer I got on IRC, but just want to confirm... If I remove my standard build of python 2.4 and do the special build of 2.5.4, I don't need to do anything more than bounce freeswitch? Or do I need to rebuild freeswitch so it builds mod_python against my new version of python? Thanks again, you guys have been quite helpful and responsive. I'm doubling what I'm paying you right now. :) Brian From anthony.minessale at gmail.com Thu Jan 29 15:35:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Jan 2009 17:35:52 -0600 Subject: [Freeswitch-users] record session in fifo In-Reply-To: References: <4979F41F.70605@gmail.com> <191c3a030901231045m784f0cd7s6994b763d02b3e26@mail.gmail.com> <497D8782.2070603@gmail.com> <497DE0C9.5090005@gmail.com> <191c3a030901260847j660bb791w27657a08bdc232d0@mail.gmail.com> <4981AD1F.2040002@gmail.com> <191c3a030901290844j51180ddej5e59372e6b971b9a@mail.gmail.com> Message-ID: <191c3a030901291535w241e96anb8ab51a328710f98@mail.gmail.com> you could make the loopback channel execute the eval app and do the originate to the sofia channel from the dialplan. or make the loopback chan exec a lua or js and fire an originate command and exit This way you don't have the loopback a and b leg as well as the sofia chan. On Thu, Jan 29, 2009 at 4:15 PM, Tamas Cseke wrote: > Hello, > > Thank you for the response. > > yes originating sofia channel directly could be an option. the only reason > we use loopback is that > our routing is in the dialplan (different proxies according to time and > prefix matching). and the entity originating the call doesn't have to deal > with them. so these are handled the same way as a "normal" manual call. > otherwise I guess we have to duplicate this logic. so we didn't find elegant > solution for this neither thus I asked maybe there is. > > Regards, > Tamas > > > On Thu, Jan 29, 2009 at 5:44 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> >> I don't think there is a very elegant way to do this and even if I find >> one i can live with it would be at least 2k worth of work to add it. >> >> Why do you need the loopback channel at all? Can't you just originate the >> sofia channel directly instead of bridging it to loopback and adding 2 more >> channels per call? >> >> >> >> >> On Thu, Jan 29, 2009 at 7:20 AM, Tamas Cseke wrote: >> >>> Hello, >>> >>> I'm sorry for still bothering you with this issue. >>> but thanks to somewhat missunderstanding/misleading by us, it shows up >>> that our customers need this feature for the complicated >>> predictive calls too. >>> >>> There is an idea, could you let me know if it is achiveable. and if it's >>> how? >>> So we record sofia channel, but the media pause/resume is happening on >>> loopback channel. >>> the idea is to have a switch_set_flag_recursive function, which set the >>> flag not only to the current channel, but to all connected channels. >>> we need to push CS_PAUSE_MEDIA flag to the sofia channel. >>> >>> Thanks in advance, >>> Tamas >>> >>> Here is the callflow again: >>> 1, originate a loopback channel via event socket >>> 2, loopback-b channel is hunting the dialplan, wich decide routing, >>> caller_id, the need for recordings and so forth, and bridge a sofia call >>> 3. the record_session is running on the sofia channel with >>> bridge_pre_execute magic vars >>> 4 loopback-a channel is pushed into the fifo >>> 5 a script get the fifo::info via event socket >>> 6 originate a call to the consumer with the proper strategy with &fifo >>> out application >>> 7 sofia channel is bridged to the consumer >>> 8 loopback channels die >>> >>> >>> >>> Anthony Minessale ?rta: >>> > yes some code was missing for some reason, try again >>> > >>> > >>> > On Mon, Jan 26, 2009 at 10:11 AM, Tamas Cseke < >>> cstomi.levlist at gmail.com>wrote: >>> > >>> > >>> >> Hello, >>> >> >>> >> I tested with the attached patch. >>> >> It is working fine in a normal case. >>> >> >>> >> I have only problems with the automatic calls, because in this case >>> the >>> >> loopback channel is in the fifo, but the record_session is running on >>> the >>> >> sofia channel. >>> >> Maybe it could be sort out with putting the bug pause/resume functions >>> into >>> >> api function, what I should turn on and off on demand? >>> >> Anyway, I quess this is a bit extreme circumstance, and it isn't so >>> >> important to us now. >>> >> >>> >> Thanks, >>> >> Tamas >>> >> >>> >> Tamas Cseke ?rta: >>> >> >>> >> Hello, >>> >> >>> >>> Thank you your help. >>> >>> >>> >>> I tested with r11489, but moh is still recorded in fifo. >>> >>> >>> >>> I quess you I should test the CF_PAUSE_BUGS in r11466. But I didn't >>> find >>> >>> where you check this flag. >>> >>> Is it maybe possible you forget to commit something? >>> >>> >>> >>> Thanks, >>> >>> Tamas >>> >>> >>> >>> >>> >>> I didn't find where you >>> >>> Anthony Minessale ?rta: >>> >>> >>> >>> >>> >>> >>> >>>> please test latest trunk. >>> >>>> Patch added to pause media bugs while not in a bridge which should >>> pause >>> >>>> recordings and cut out the moh. >>> >>>> >>> >>>> >>> >>>> On Fri, Jan 23, 2009 at 10:45 AM, Tamas Cseke < >>> cstomi.levlist at gmail.com >>> >>>> >>> >>>>> wrote: >>> >>>>> >>> >>>> >>> >>>> >>> >>>>> Hello, >>> >>>>> >>> >>>>> we would like to distribute calls with fifo and record these >>> sessions >>> >>>>> but we'd like to skip the recording while the caller is waiting. >>> >>>>> (we don't need to record the hold music, just the speech with the >>> fifo >>> >>>>> consumer.) >>> >>>>> >>> >>>>> I tried >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> but it doesn't work because the channel is answered immediately >>> when the >>> >>>>> caller is pushed into the fifo. >>> >>>>> (I don't know if there exists any other channel flag that could be >>> use >>> >>>>> here) >>> >>>>> >>> >>>>> I also tried fifo_record_template. >>> >>>>> but it records the session from the point of view of the consumer's >>> >>>>> session, and after the bridge the recording is stopped. >>> >>>>> we would like to record the whole session into a single file even >>> after >>> >>>>> calltransfers >>> >>>>> >>> >>>>> moreover we'd like to use some kind of predcitive dialing >>> >>>>> which >>> >>>>> 1, originate a loopback channel via event socket >>> >>>>> 2, loopback-b channel is hunting the dialplan, wich decide routing, >>> >>>>> caller_id, the need for recordings and so forth, and bridge a sofia >>> call >>> >>>>> 3. the record_session is running on the sofia channel with >>> >>>>> bridge_pre_execute magic vars >>> >>>>> 4 loopback-a channel is pushed into the fifo >>> >>>>> 5 a script get the fifo::info via event socket >>> >>>>> 6 originate a call to the consumer with the proper strategy with >>> &fifo >>> >>>>> out application >>> >>>>> 7 sofia channel is bridged to the consumer >>> >>>>> 8 loopback channels die >>> >>>>> >>> >>>>> after transfers everything is recorded into one file. >>> >>>>> but the problem here is again the unwanted recording in the fifo >>> while >>> >>>>> the caller is waiting >>> >>>>> >>> >>>>> Could you please advise me any solution, if there is? >>> >>>>> >>> >>>>> >>> >>>>> Thank you, >>> >>>>> Tamas >>> >>>>> >>> >>>>> >>> >>>>> _______________________________________________ >>> >>>>> Freeswitch-users mailing list >>> >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at 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 at 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 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 >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/80b98c6a/attachment-0001.html From anthony.minessale at gmail.com Thu Jan 29 15:39:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Jan 2009 17:39:34 -0600 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <1233270471.4757.74.camel@dev03.cal.highergear.com> References: <1233196879.4757.54.camel@dev03.cal.highergear.com> <45C07A45-AA15-4AC2-8ED0-73F4485CF190@freeswitch.org> <191c3a030901290559l69c5a2b3t2905fbbf3467b764@mail.gmail.com> <1233261066.4757.64.camel@dev03.cal.highergear.com> <191c3a030901291249t3159f33bg875b90da74564289@mail.gmail.com> <1233270471.4757.74.camel@dev03.cal.highergear.com> Message-ID: <191c3a030901291539w4071a84g3d0222e3a19ebcb0@mail.gmail.com> for good measure i would rebuild just mod_python make mod_python-clean make mod_python-install On Thu, Jan 29, 2009 at 5:07 PM, Brian Deacon wrote: > On Thu, 2009-01-29 at 14:49 -0600, Anthony Minessale wrote: > > We used to build python 2.5.1 and install it into the freeswitch > > prefix > > > > we did ./configure --prefix=/usr/local/freeswitch > > --enable-threads CFLAGSFORSHARED="-fPIC" > > > > But the makefile no longer will find that one so....... > > > > you could move your python out of the way on your box and try > > installing this as your system python instead. > > > > ./configure --prefix=/your/real/python/prefix > > --enable-threads CFLAGSFORSHARED="-fPIC" > > > > if you are really brave you could delete the Makefile in mod_python > > and put back the old one > > that builds and installs custom python for you into the FS prefix > > > > > http://fisheye.freeswitch.org/browse/~raw,r=6393/FreeSWITCH/src/mod/languages/mod_python/Makefile > > > > > > the important thing is the --enable-threads and > > CFLAGSFORSHARED="-fPIC" on whatever one you use. > > > > So this is the error that gdb shows me, do you think this is the kind of > thing that the "special" compile would clear up? > > 2009-01-29 15:57:12 [NOTICE] mod_python.c:107 eval_some_python() > Invoking py module: callcontrol.routing.DummyRouteHandler > 2009-01-29 15:57:12 [ERR] mod_python.c:121 eval_some_python() Error > importing module > Traceback (most recent call last): > File > "/usr/local/freeswitch/python/callcontrol/routing/DummyRouteHandler.py", > line 19, in ? > import sqlalchemy > File "/usr/lib/python2.4/site-packages/sqlalchemy/__init__.py", line > 8, in ? > from sqlalchemy.types import \ > File "/usr/lib/python2.4/site-packages/sqlalchemy/types.py", line 25, > in ? > import datetime as dt > ImportError: /usr/lib/python2.4/lib-dynload/datetime.so: undefined > symbol: _Py_ZeroStruct > > > > And I believe I understood the answer I got on IRC, but just want to > confirm... If I remove my standard build of python 2.4 and do the > special build of 2.5.4, I don't need to do anything more than bounce > freeswitch? Or do I need to rebuild freeswitch so it builds mod_python > against my new version of python? > > Thanks again, you guys have been quite helpful and responsive. I'm > doubling what I'm paying you right now. :) > > Brian > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/17e3a199/attachment.html From bdeacon at highergear.com Thu Jan 29 16:33:00 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Thu, 29 Jan 2009 16:33:00 -0800 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <191c3a030901291539w4071a84g3d0222e3a19ebcb0@mail.gmail.com> References: <1233196879.4757.54.camel@dev03.cal.highergear.com> <45C07A45-AA15-4AC2-8ED0-73F4485CF190@freeswitch.org> <191c3a030901290559l69c5a2b3t2905fbbf3467b764@mail.gmail.com> <1233261066.4757.64.camel@dev03.cal.highergear.com> <191c3a030901291249t3159f33bg875b90da74564289@mail.gmail.com> <1233270471.4757.74.camel@dev03.cal.highergear.com> <191c3a030901291539w4071a84g3d0222e3a19ebcb0@mail.gmail.com> Message-ID: <1233275580.4757.79.camel@dev03.cal.highergear.com> On Thu, 2009-01-29 at 17:39 -0600, Anthony Minessale wrote: > for good measure i would rebuild just mod_python > > make mod_python-clean > make mod_python-install > I've got the newly built python sitting in /usr/lib/python2.5, and the old one still in /usr/lib/python2.4. /usr/bin/python is now pointed to python2.5. After a ./configure in freeswitch, I notice that the mod_python Makefile has LOCAL_CFLAGS with a -I/usr/include/python2.5, but PYTHON_SITE_DIR is still set to /usr/lib/python2.4/site-packages I'm trying just altering that variable to 2.5, but worried that maybe other places are getting 2.4 set. How does the logic work for how that gets there? Is it just searching for the first /usr/lib/python*? Brian From tleyden at branchcut.com Thu Jan 29 17:05:23 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Fri, 30 Jan 2009 05:35:23 +0430 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python Message-ID: > > > 2009-01-29 15:57:12 [NOTICE] mod_python.c:107 eval_some_python() > Invoking py module: callcontrol.routing.DummyRouteHandler > 2009-01-29 15:57:12 [ERR] mod_python.c:121 eval_some_python() Error > importing module > Traceback (most recent call last): > File > "/usr/local/freeswitch/python/callcontrol/routing/DummyRouteHandler.py", > line 19, in ? > import sqlalchemy > File "/usr/lib/python2.4/site-packages/sqlalchemy/__init__.py", line > 8, in ? > from sqlalchemy.types import \ > File "/usr/lib/python2.4/site-packages/sqlalchemy/types.py", line 25, > in ? > import datetime as dt > ImportError: /usr/lib/python2.4/lib-dynload/datetime.so: undefined > symbol: _Py_ZeroStruct > Ahh, that looks like the "global symbol import" error. http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2Fusr.2Flib.2F...2Fdatetime.so:_undefinedsymbol:__Py_ZeroStruct Should be very easy to fix.. see the wiki page. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/23cf95b5/attachment.html From bdeacon at highergear.com Thu Jan 29 17:22:37 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Thu, 29 Jan 2009 17:22:37 -0800 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <191c3a030901291539w4071a84g3d0222e3a19ebcb0@mail.gmail.com> References: <1233196879.4757.54.camel@dev03.cal.highergear.com> <45C07A45-AA15-4AC2-8ED0-73F4485CF190@freeswitch.org> <191c3a030901290559l69c5a2b3t2905fbbf3467b764@mail.gmail.com> <1233261066.4757.64.camel@dev03.cal.highergear.com> <191c3a030901291249t3159f33bg875b90da74564289@mail.gmail.com> <1233270471.4757.74.camel@dev03.cal.highergear.com> <191c3a030901291539w4071a84g3d0222e3a19ebcb0@mail.gmail.com> Message-ID: <1233278557.4757.91.camel@dev03.cal.highergear.com> My apologies that this is starting to turn into "teach the n00b how to compile stuff". (Not that that's keeping me from asking for help...) So I downloaded the 2.5.4 source tarball from python.org and ./configure --prefix=/usr --enable-threads CFLAGSFORSHARED="-fPIC" make make install I have a /usr/lib/python2.5 that looks kosher. I think I may have had to alter some of the hardlinks in /usr/bin because /usr/bin/python2.4 is still there. (So is /usr/lib/python2.4) I have trunk code from freeswitch. ./configure acts happy, but the mod_python Makefile looks ganked: LOCAL_CFLAGS = -I/usr/include/python2.5 -I/usr/include/python2.5 -fno-strict-aliasing -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes LOCAL_LDFLAGS= -lpthread -ldl -lutil -lm -lpython2.5 LOCAL_OBJS=freeswitch_python.o mod_python_wrap.o include ../../../../build/modmake.rules LINK=$(CXXLINK) PYMOD=freeswitch PYTHON_SITE_DIR=/usr/lib/python2.4/site-packages That PYTHON_SITE_DIR is wrong, but the deal-breaker I get right now is that the -lpython2.5 for LDFLAGS is making it mad. I get an ld error "cannot find -lpython2.5" And a libpython2.5.so is conspicuously absent from my machine. This smells a little bit like I'm not compiling python correctly, for which I would not blame you as putting under the category of "not your problem" but I'll TRIPLE what I'm paying you if you can help lead me out of these woods. :) Brian On Thu, 2009-01-29 at 17:39 -0600, Anthony Minessale wrote: > for good measure i would rebuild just mod_python > > make mod_python-clean > make mod_python-install > > > On Thu, Jan 29, 2009 at 5:07 PM, Brian Deacon > wrote: > On Thu, 2009-01-29 at 14:49 -0600, Anthony Minessale wrote: > > We used to build python 2.5.1 and install it into the > freeswitch > > prefix > > > > we did ./configure --prefix=/usr/local/freeswitch > > --enable-threads CFLAGSFORSHARED="-fPIC" > > > > But the makefile no longer will find that one so....... > > > > you could move your python out of the way on your box and > try > > installing this as your system python instead. > > > > ./configure --prefix=/your/real/python/prefix > > --enable-threads CFLAGSFORSHARED="-fPIC" > > > > if you are really brave you could delete the Makefile in > mod_python > > and put back the old one > > that builds and installs custom python for you into the FS > prefix > > > > > http://fisheye.freeswitch.org/browse/~raw,r=6393/FreeSWITCH/src/mod/languages/mod_python/Makefile > > > > > > the important thing is the --enable-threads and > > CFLAGSFORSHARED="-fPIC" on whatever one you use. > > > > > So this is the error that gdb shows me, do you think this is > the kind of > thing that the "special" compile would clear up? > > 2009-01-29 15:57:12 [NOTICE] mod_python.c:107 > eval_some_python() > Invoking py module: callcontrol.routing.DummyRouteHandler > 2009-01-29 15:57:12 [ERR] mod_python.c:121 eval_some_python() > Error > importing module > Traceback (most recent call last): > File > "/usr/local/freeswitch/python/callcontrol/routing/DummyRouteHandler.py", > line 19, in ? > import sqlalchemy > File > "/usr/lib/python2.4/site-packages/sqlalchemy/__init__.py", > line > 8, in ? > from sqlalchemy.types import \ > File "/usr/lib/python2.4/site-packages/sqlalchemy/types.py", > line 25, > in ? > import datetime as dt > ImportError: /usr/lib/python2.4/lib-dynload/datetime.so: > undefined > symbol: _Py_ZeroStruct > > > > And I believe I understood the answer I got on IRC, but just > want to > confirm... If I remove my standard build of python 2.4 and do > the > special build of 2.5.4, I don't need to do anything more than > bounce > freeswitch? Or do I need to rebuild freeswitch so it builds > mod_python > against my new version of python? > > Thanks again, you guys have been quite helpful and > responsive. I'm > doubling what I'm paying you right now. :) > > > Brian > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Jan 29 17:22:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 29 Jan 2009 19:22:44 -0600 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: References: Message-ID: <191c3a030901291722y7da8d65fid824c966f91a61d5@mail.gmail.com> Traun, If you knew that, why didn't you commit it to trunk? I added it to r11560 Brian, Can you try freeswitch latest trunk? On Thu, Jan 29, 2009 at 7:05 PM, Traun Leyden wrote: > >> 2009-01-29 15:57:12 [NOTICE] mod_python.c:107 eval_some_python() >> Invoking py module: callcontrol.routing.DummyRouteHandler >> 2009-01-29 15:57:12 [ERR] mod_python.c:121 eval_some_python() Error >> importing module >> Traceback (most recent call last): >> File >> "/usr/local/freeswitch/python/callcontrol/routing/DummyRouteHandler.py", >> line 19, in ? >> import sqlalchemy >> File "/usr/lib/python2.4/site-packages/sqlalchemy/__init__.py", line >> 8, in ? >> from sqlalchemy.types import \ >> File "/usr/lib/python2.4/site-packages/sqlalchemy/types.py", line 25, >> in ? >> import datetime as dt >> ImportError: /usr/lib/python2.4/lib-dynload/datetime.so: undefined >> symbol: _Py_ZeroStruct >> > > > Ahh, that looks like the "global symbol import" error. > > > http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2Fusr.2Flib.2F...2Fdatetime.so:_undefinedsymbol:__Py_ZeroStruct > > Should be very easy to fix.. see the wiki page. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090129/2603fe7c/attachment-0001.html From bdeacon at highergear.com Thu Jan 29 18:06:02 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Thu, 29 Jan 2009 18:06:02 -0800 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <191c3a030901291722y7da8d65fid824c966f91a61d5@mail.gmail.com> References: <191c3a030901291722y7da8d65fid824c966f91a61d5@mail.gmail.com> Message-ID: <1233281162.4757.94.camel@dev03.cal.highergear.com> Building now, Dunno if this is a dork-up on my part or a problem with the svn version versus the tarball version, but after a bootstrap.sh, configure, make megaclean and make installall, it complained about the lack of a quiet_libtool in the root of the source directory. I just copied in the one I had from my tarball version and the build is chewing on it. I'll let you know if this does the trick. Brian On Thu, 2009-01-29 at 19:22 -0600, Anthony Minessale wrote: > Traun, > > If you knew that, why didn't you commit it to trunk? > > I added it to r11560 > > Brian, > > Can you try freeswitch latest trunk? > > > On Thu, Jan 29, 2009 at 7:05 PM, Traun Leyden > wrote: > > 2009-01-29 15:57:12 [NOTICE] mod_python.c:107 > eval_some_python() > Invoking py module: > callcontrol.routing.DummyRouteHandler > 2009-01-29 15:57:12 [ERR] mod_python.c:121 > eval_some_python() Error > importing module > Traceback (most recent call last): > File > "/usr/local/freeswitch/python/callcontrol/routing/DummyRouteHandler.py", > line 19, in ? > import sqlalchemy > File > "/usr/lib/python2.4/site-packages/sqlalchemy/__init__.py", line > 8, in ? > from sqlalchemy.types import \ > File > "/usr/lib/python2.4/site-packages/sqlalchemy/types.py", line 25, > in ? > import datetime as dt > ImportError: /usr/lib/python2.4/lib-dynload/datetime.so: undefined > symbol: _Py_ZeroStruct > > > > Ahh, that looks like the "global symbol import" error. > > http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2Fusr.2Flib.2F...2Fdatetime.so:_undefinedsymbol:__Py_ZeroStruct > > Should be very easy to fix.. see the wiki page. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Jan 29 18:14:41 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 29 Jan 2009 20:14:41 -0600 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <1233281162.4757.94.camel@dev03.cal.highergear.com> References: <191c3a030901291722y7da8d65fid824c966f91a61d5@mail.gmail.com> <1233281162.4757.94.camel@dev03.cal.highergear.com> Message-ID: <123B3667-0C52-4069-A981-A97CE81C2087@freeswitch.org> You don't use make installall anymore. Just use "make current" to bring it up to trunk. /b On Jan 29, 2009, at 8:06 PM, Brian Deacon wrote: > Building now, > Dunno if this is a dork-up on my part or a problem with the svn > version > versus the tarball version, but after a bootstrap.sh, configure, make > megaclean and make installall, it complained about the lack of a > quiet_libtool in the root of the source directory. I just copied in > the > one I had from my tarball version and the build is chewing on it. > > I'll let you know if this does the trick. > > Brian From bdeacon at highergear.com Thu Jan 29 18:27:26 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Thu, 29 Jan 2009 18:27:26 -0800 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <191c3a030901291722y7da8d65fid824c966f91a61d5@mail.gmail.com> References: <191c3a030901291722y7da8d65fid824c966f91a61d5@mail.gmail.com> Message-ID: <1233282446.4757.99.camel@dev03.cal.highergear.com> TA-DA! My python can now not only import the sqlalchemy module, but the code I had before that was actually doing some database interaction is working now. Thank you very much for all the help. It is much appreciated. It sounds like the contents on the wiki with the modules.conf.xml or the LD workarounds are superceded now. Should I change the comments on there to reflect that it should now be fixed? Brian On Thu, 2009-01-29 at 19:22 -0600, Anthony Minessale wrote: > Traun, > > If you knew that, why didn't you commit it to trunk? > > I added it to r11560 > > Brian, > > Can you try freeswitch latest trunk? > > > On Thu, Jan 29, 2009 at 7:05 PM, Traun Leyden > wrote: > > 2009-01-29 15:57:12 [NOTICE] mod_python.c:107 > eval_some_python() > Invoking py module: > callcontrol.routing.DummyRouteHandler > 2009-01-29 15:57:12 [ERR] mod_python.c:121 > eval_some_python() Error > importing module > Traceback (most recent call last): > File > "/usr/local/freeswitch/python/callcontrol/routing/DummyRouteHandler.py", > line 19, in ? > import sqlalchemy > File > "/usr/lib/python2.4/site-packages/sqlalchemy/__init__.py", line > 8, in ? > from sqlalchemy.types import \ > File > "/usr/lib/python2.4/site-packages/sqlalchemy/types.py", line 25, > in ? > import datetime as dt > ImportError: /usr/lib/python2.4/lib-dynload/datetime.so: undefined > symbol: _Py_ZeroStruct > > > > Ahh, that looks like the "global symbol import" error. > > http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2Fusr.2Flib.2F...2Fdatetime.so:_undefinedsymbol:__Py_ZeroStruct > > Should be very easy to fix.. see the wiki page. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Jan 29 18:34:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 29 Jan 2009 20:34:04 -0600 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <1233282446.4757.99.camel@dev03.cal.highergear.com> References: <191c3a030901291722y7da8d65fid824c966f91a61d5@mail.gmail.com> <1233282446.4757.99.camel@dev03.cal.highergear.com> Message-ID: <3065CCBA-1C35-4117-811A-0E9F6A5ABCAB@freeswitch.org> Can you do some examples and documentation on the wiki about what you're doing to maybe help others? /b On Jan 29, 2009, at 8:27 PM, Brian Deacon wrote: > TA-DA! > > My python can now not only import the sqlalchemy module, but the > code I > had before that was actually doing some database interaction is > working > now. > > Thank you very much for all the help. It is much appreciated. > > It sounds like the contents on the wiki with the modules.conf.xml or > the > LD workarounds are superceded now. Should I change the comments on > there to reflect that it should now be fixed? > > Brian From jmesquita at gmail.com Thu Jan 29 18:39:40 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 30 Jan 2009 00:39:40 -0200 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <3065CCBA-1C35-4117-811A-0E9F6A5ABCAB@freeswitch.org> References: <191c3a030901291722y7da8d65fid824c966f91a61d5@mail.gmail.com> <1233282446.4757.99.camel@dev03.cal.highergear.com> <3065CCBA-1C35-4117-811A-0E9F6A5ABCAB@freeswitch.org> Message-ID: <3C247D5E-96EA-48A0-8383-11A649330742@gmail.com> I have to ask to _please_ do add more info on the wiki about that cos python is a language that has been growing a lot on commercial and open source worlds. I have a personal interest on that as well since I intend to do some things with the mod_python module as well. Thanks, Mesquita On Jan 30, 2009, at 12:34 AM, Brian West wrote: > Can you do some examples and documentation on the wiki about what > you're doing to maybe help others? > > /b > > On Jan 29, 2009, at 8:27 PM, Brian Deacon wrote: > >> TA-DA! >> >> My python can now not only import the sqlalchemy module, but the >> code I >> had before that was actually doing some database interaction is >> working >> now. >> >> Thank you very much for all the help. It is much appreciated. >> >> It sounds like the contents on the wiki with the modules.conf.xml or >> the >> LD workarounds are superceded now. Should I change the comments on >> there to reflect that it should now be fixed? >> >> Brian > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peder at networkoblivion.com Thu Jan 29 19:16:49 2009 From: peder at networkoblivion.com (peder at networkoblivion.com) Date: Thu, 29 Jan 2009 21:16:49 -0600 Subject: [Freeswitch-users] mod_g729 In-Reply-To: <4ca506420901271348y1f5e4076ha6f13b7ad07e3b05@mail.gmail.com> References: <200901221537.57260.krivushinme@rn-inform.tomsk.ru> <4978A087.70301@freeswitch.org> <87f2f3b90901221134x5c64c1ct6214dc1e4db34693@mail.gmail.com> <001301c97cd6$02107b90$063172b0$@net> <4ca506420901270936x16eafee5if7370a6a755caadd@mail.gmail.com> <497F5DD1.8080507@networkoblivion.com> <4ca506420901271348y1f5e4076ha6f13b7ad07e3b05@mail.gmail.com> Message-ID: <49827121.9040508@networkoblivion.com> Do you just specify g722 for the codec prefs on FS? I added the items below and have FS set to use g722. When I call, I see that the phone offers g722 and g711u and FS chooses g722. On the phone I see g722 as inbound and outbound, but I don't hear anything. I've tried speaker and handset and get nothing on either one. Also, the phone doesn't show packets increasing even though I can see FS is playing files. It is an IP600. I switch the Polycom to offer 711u first and then g722 and FS chooses 711u and works fine. Jason Garland wrote: > Something like this might do it... ;) > > > > > > > > > > > > > > > > > > On Tue, Jan 27, 2009 at 2:17 PM, peder at networkoblivion.com > > wrote: > > > Codec Options* > > G.711 Codec, G.726, G.729AB, G.723.1A, G.722 wideband codec > > > > Speex is not listed, so Polycom can't do Speex. > > > > I should note that even the non-HD Polycom phones have this same chip > > and are capable of doing G.722 some some config tweaking. ;) > > Are you saying that you have actually gotten the non-HD versions to do > g.722? Or are you saying that one might be able to do this? If the > former, can you share what you did? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tleyden at branchcut.com Thu Jan 29 19:27:40 2009 From: tleyden at branchcut.com (Traun Leyden) Date: Fri, 30 Jan 2009 07:57:40 +0430 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python Message-ID: I don't know, I must have had a momentary lapse of reason .. but at least I documented it! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/a41b5873/attachment.html From sias at cpdata.co.za Fri Jan 30 02:39:44 2009 From: sias at cpdata.co.za (Sias Mey) Date: Fri, 30 Jan 2009 12:39:44 +0200 Subject: [Freeswitch-users] Conference dialing and uuid Message-ID: <20090130103944.GA23536@cpdata.co.za> Hi, Im trying to build a web based conference control system. Got most of it sorted with some help from the list but I seem to have run into some strangeness. I use a conference dial call to pull extra users into the conference. I couldent find a way of setting channel variables or executing javascript directly on the conference dial since it expects and endpoint and the {} syntax produced an error. So now I am using the Loopback inteface to register some values. One of the functions these script fullfill is to register the uuid of the new channel in the database, however after thinking for a while all of this is working fine upon further testing I found that going through loopback generated 3 channels. I was saving the uuid of the a leg of the call to loopback into the database. However to manipulate the call in the conference I need the uuid of the leg bridgeing to the conference. I have tried some queries against the core database in via javascript however there seems to be some delay as to when the needed leg gets inserted into the channels table. I have tried with execute_on_ring and execute_on_answer. But I suspect that the call only gets added after it is actually answered. Is there some way for me to find the uuid of this call? I can use api calls via conference list to find all the calls in the conference, however if two people get added to the same conference in rapid succesion it will be quite circumstantial as to which one is which since I dont have a way of directly relating the call back to the channels that originally spawned it without a lot of costly text comparisons. Any help from someone who understands this beter than I do will be greatly appreciated. Thanks in advance, Sias From brian at freeswitch.org Fri Jan 30 02:45:54 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 30 Jan 2009 04:45:54 -0600 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090130103944.GA23536@cpdata.co.za> References: <20090130103944.GA23536@cpdata.co.za> Message-ID: What wasn't working about this? The {} can be used everywhere without a problem... Maybe you can provide more details on this. /b On Jan 30, 2009, at 4:39 AM, Sias Mey wrote: > > I couldent find a way of setting channel variables or executing > javascript directly on the conference dial since it expects and > endpoint > and the {} syntax produced an error. So now I am using the Loopback > inteface to register some values. From pmhshz at gmail.com Fri Jan 30 03:11:20 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 30 Jan 2009 03:11:20 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> Message-ID: <21745312.post@talk.nabble.com> Thanks, Anthony In my previous test sps did not changed, but in recent test sps was dropped to 0 itself (as below). =============================================================== UP 0 years, 0 days, 5 hours, 1 minute, 53 seconds, 878 milliseconds, 190 microseconds 5474 session(s) since startup 75 session(s) 0/0 ============================================================= My system is 32 bit. CPU is Intel(R) Xeon(R) CPU X3220 @ 2.40GHz And RAM is 4GB Output of ulimit -a is: ulimit -a: (set after first test) core file size (blocks, -c) unlimited data seg size (kbytes, -d) unlimited max nice (-e) 20 file size (blocks, -f) unlimited pending signals (-i) unlimited max locked memory (kbytes, -l) unlimited max memory size (kbytes, -m) unlimited open files (-n) 999999 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) unlimited max rt priority (-r) unlimited stack size (kbytes, -s) 244 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited file locks (-x) unlimited =================================================== BTW using trunk on production system is safe? Warm thanks for kind responses... Anthony Minessale-2 wrote: > > When you get it in that state what do you see when you execute > > fsctl sps > > is the sps a very low number? > > Did the sps drop by itself from the value you originally set it to? > > Are you using 32 bit? > > if so try all of these commands in your shell before starting FS > > ulimit -c unlimited > ulimit -d unlimited > ulimit -f unlimited > ulimit -i unlimited > ulimit -n 999999 > ulimit -q unlimited > ulimit -u unlimited > ulimit -v unlimited > ulimit -x unlimited > ulimit -s 244 > ulimit -l unlimited > > > DO NOT put them in a script unless you source the script with . > . myscript or they will be undone instantly when the script exits > > BTW, I said to try latest trunk not 1.0.2 We can only debug the > development > code at this point. > > > > > > On Thu, Jan 29, 2009 at 10:06 AM, shehzad p wrote: > >> >> Hi Anthony, >> >> I found interesting result while testing Freeswitch, and it might be >> cause >> of freezing out of freeswitch, >> >> I updated my system (as you told) with latest stable version Freeswitch >> 1.0.2 >> First of all I set sps to 100, >> Then I sends call approximately 100 per seconds, Freeswitch works fine >> and >> handles all the calls very well. >> >> After that I send 130 calls per seconds, and magic happen now, Freeswitch >> handles first 100 calls only. >> all the preceding calls were failed (even not appeared in freeswitch >> console >> why?) >> >> When I put ngrep trace, System responds with 503 Maximum Calls In >> Progress. >> (as below) >> ########################################################### >> # >> U FSFSFSFSFS -> GWGWGWGWGW >> SIP/2.0 503 Maximum Calls In Progress. >> Via: SIP/2.0/UDP GWGWGWGWGW;branch=z9hG4bK53eafabb;rport=5060. >> From: "99999" ;tag=as2e10c170. >> To: ;tag=K3jSUFrDHpmmB. >> Call-ID: 0feb229e58afe7be17110deb361dc234 at GWGWGWGWGW. >> CSeq: 102 INVITE. >> Retry-After: 300. >> User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported. >> Accept: application/sdp. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, >> REFER, UPDATE, REGISTER, INFO, PUBLISH. >> Supported: timer, precondition, path, replaces. >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer. >> Content-Length: 0. >> . >> ##################################################################### >> >> >> Now another issue to note down is that, >> After all above happened and active calls comes to zero, >> I just make a single call which also fails with response 503 - Maximum >> Calls >> In Progress. >> >> >> Is this intended behaviour, should I increase SPS to overcome this. or >> something like bug. >> >> Please let me know what should be the resolution for this. >> >> Thanks, >> msp >> >> >> >> Anthony Minessale-2 wrote: >> > >> > Also remember, >> > Actually completely uninstall and erase /usr/local/freeswitch and the >> > 1.0.1 >> > source tree and freshly install the new one. >> > If you try to upgrade on top of a release with trunk it will cause more >> > problems for you. >> > >> > >> > On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice wrote: >> > >> >> Upgrade to trunk... Many many issues have been resolved since 1.0.1 >> was >> >> the >> >> current release >> >> >> >> >> >> > From: shehzad p >> >> > Reply-To: >> >> > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) >> >> > To: >> >> > Subject: [Freeswitch-users] Freeswitch freezes on increasing call >> >> traffic >> >> > >> >> > >> >> > Hi all, >> >> > >> >> > Yesterday my Freeswitch server faced a problem when call traffic >> >> increased >> >> > to more than 100. >> >> > >> >> > When I start Freeswitch, it works fine and then after some time >> >> > (approximately 15 to 20 minutes) it stops functioning (means no >> call >> >> is >> >> > being processed, no CLI command is working and it just freezes) >> until >> I >> >> > restart the freeswitch. >> >> > >> >> > I am using Freeswitch 1.0.1. >> >> > Debug (gdb) trace as on wiki page >> >> > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.sh is >> >> attached >> >> > http://www.nabble.com/file/p21701744/fs_debgu.txt fs_debgu.txt >> >> > -- >> >> > View this message in context: >> >> > >> >> >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 >> >> > p21701744.html >> >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> > >> >> > >> >> > _______________________________________________ >> >> > Freeswitch-users mailing list >> >> > Freeswitch-users at 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 at 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 at hotmail.com >> < >> MSN%3Aanthony_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 >> < >> sip%3A888 at conference.freeswitch.org >> > >> > iax:guest at conference.freeswitch.org/888 >> > >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> > pstn:213-799-1400 >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21729863.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21745312.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kawarod at laposte.net Fri Jan 30 03:42:33 2009 From: kawarod at laposte.net (rod) Date: Fri, 30 Jan 2009 15:42:33 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC Message-ID: <4982E7A9.4070103@laposte.net> Dear list, I've been playing with freeswitch for some time (2 months) and the fact is that I'm very pleased with the functionnalities of this software. I'd like to use FS as a SBC handling media and I'm doing some tests with sipp to load the machine but I'm unable to bridge more than 60 calls without seeing the CPU being loaded at 100%. I'm sure something is going wrong with my setup but I'm unable to see what. The test machine has the following specs: Athlon XP 3500+ with 2GB of memory (I know this is not a high end machine :p) Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 95 model name : AMD Athlon(tm) 64 Processor 3500+ stepping : 2 cpu MHz : 2199.973 cache size : 512 KB fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm extapic cr8_legacy bogomips : 4402.97 TLB size : 1024 4K pages clflush size : 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc I installed FS on a fresh debian 64: Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 UTC 2009 x86_64 GNU/Linux I set the ulimit parameters like those on the website: freeswitch at internal> ... Freeswitch:/opt/free-svn/bin# ulimit -a core file size (blocks, -c) unlimited data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) unlimited max locked memory (kbytes, -l) unlimited max memory size (kbytes, -m) unlimited open files (-n) 999999 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) unlimited real-time priority (-r) 0 stack size (kbytes, -s) 244 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited file locks (-x) unlimited My network setup is the following: SIPP machine (10.10.10.1/24)----------------vlan 55 ----------(10.10.10.254/24) FS (10.10.20.254/24)-------------- vlan56 -------------------(10.10.20.100/24) OTHER STOCK FS I launched sipp with: sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi 10.10.10.1 -i 10.10.10.1 -mp 25000 10.10.10.254:5060 The dialplan on FS is very simple: FreeSWITCH Version 1.0.trunk (11560M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[100] SQL [Enabled] The test is very simple: sipp dial 9999 that matches in my FS dialplan and this is bridged to an other FS machine playing music on hold. When I launch "top" I see after 30 to 40 s that FS consumes all the CPU ressources (with a mean of 50-60 % before), with 80 calls. When I set 70 calls, I have to wait 70-80 s before seeing the same issue. Presence is set to false on the 2 profile. I have the same issue with FS 1.0.2 that' s why I tried FS 11560. When I use the FS machine as a router to test the packet per second performance, I'm reaching 100Mbps with 8000pps in each direction (from vlan 55 to vlan56) with less than 12% CPU. So that I don't think there's an issue with the network. Here is an "mpstat -P ALL 1" to show you what's happening suddenly with 70 bridge calls: 12:31:26 CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 0,00 89,00 6241,00 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 0,00 89,00 6241,00 12:31:27 CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 0,00 22,22 6035,35 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 0,00 22,22 6035,35 12:31:28 CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 0,00 0,00 5483,17 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 0,00 0,00 5483,17 The CPU is going from 89% idle to 0% in less than 2 seconds. I know that I don't have to expect too much from this kind of hardware, but it seems strange that the CPU power vanished so suddenly. Thanks a lot for the guys that have read this long mail :p kind regards, rod From kawarod at laposte.net Fri Jan 30 03:56:05 2009 From: kawarod at laposte.net (rod) Date: Fri, 30 Jan 2009 15:56:05 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC Message-ID: <4982EAD5.1010200@laposte.net> Dear list, I've been playing with freeswitch for some time (2 months) and the fact is that I'm very pleased with the functionnalities of this software. I'd like to use FS as a SBC handling media and I'm doing some tests with sipp to load the machine but I'm unable to bridge more than 60 calls without seeing the CPU being loaded at 100%. I'm sure something is going wrong with my setup but I'm unable to see what. The test machine has the following specs: Athlon XP 3500+ with 2GB of memory (I know this is not a high end machine :p) Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 95 model name : AMD Athlon(tm) 64 Processor 3500+ stepping : 2 cpu MHz : 2199.973 cache size : 512 KB fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm extapic cr8_legacy bogomips : 4402.97 TLB size : 1024 4K pages clflush size : 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc I installed FS on a fresh debian 64: Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 UTC 2009 x86_64 GNU/Linux I set the ulimit parameters like those on the website: freeswitch at internal> ... Freeswitch:/opt/free-svn/bin# ulimit -a core file size (blocks, -c) unlimited data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) unlimited max locked memory (kbytes, -l) unlimited max memory size (kbytes, -m) unlimited open files (-n) 999999 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) unlimited real-time priority (-r) 0 stack size (kbytes, -s) 244 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited file locks (-x) unlimited My network setup is the following: SIPP machine (10.10.10.1/24)----------------vlan 55 ----------(10.10.10.254/24) FS (10.10.20.254/24)-------------- vlan56 -------------------(10.10.20.100/24) OTHER STOCK FS I launched sipp with: sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi 10.10.10.1 -i 10.10.10.1 -mp 25000 10.10.10.254:5060 The dialplan on FS is very simple: FreeSWITCH Version 1.0.trunk (11560M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[100] SQL [Enabled] The test is very simple: sipp dial 9999 that matches in my FS dialplan and this is bridged to an other FS machine playing music on hold. When I launch "top" I see after 30 to 40 s that FS consumes all the CPU ressources (with a mean of 50-60 % before), with 80 calls. When I set 70 calls, I have to wait 70-80 s before seeing the same issue. Presence is set to false on the 2 profile. I have the same issue with FS 1.0.2 that' s why I tried FS 11560. When I use the FS machine as a router to test the packet per second performance, I'm reaching 100Mbps with 8000pps in each direction (from vlan 55 to vlan56) with less than 12% CPU. So that I don't think there's an issue with the network. Here is an "mpstat -P ALL 1" to show you what's happening suddenly with 70 bridge calls: 12:31:26 CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 0,00 89,00 6241,00 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 0,00 89,00 6241,00 12:31:27 CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 0,00 22,22 6035,35 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 0,00 22,22 6035,35 12:31:28 CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 0,00 0,00 5483,17 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 0,00 0,00 5483,17 The CPU is going from 89% idle to 0% in less than 2 seconds. I know that I don't have to expect too much from this kind of hardware, but it seems strange that the CPU power vanished so suddenly. Thanks a lot for the guys that have read this long mail :p kind regards, rod From leon at scarlet-internet.nl Fri Jan 30 04:11:54 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Fri, 30 Jan 2009 13:11:54 +0100 Subject: [Freeswitch-users] LDAP Integration In-Reply-To: <4962D64D.3080809@skopis.com> References: <49417123.10709@ydeasolutions.com.br> <49417538.9040203@ydeasolutions.com.br> <200812120842.00808.hads@nice.net.nz> <49418790.60001@ydeasolutions.com.br> <87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com> <494198F3.10806@ydeasolutions.com.br> <4947A61F.6060806@ydeasolutions.com.br> <49485AD1.5070708@skopis.com> <4949025F.9040008@ydeasolutions.com.br> <49621532.5080003@ydeasolutions.com.br> <4962D64D.3080809@skopis.com> Message-ID: <2DCF79D8-E2B5-43EB-93D1-EED92506E8DF@scarlet-internet.nl> Hi John, I've been trying to get your mod_xml_ldap module running, but didn't get very far yet.. What is the official way to get the module built ? I tried modifying trunk/freeswitch.spec so that XML_INT_MODULES contains xml_int/mod_xml_ldap There's also a directories/mod_ldap in DISABLED_MODULES in the same file, but I don't suppose it's necessary to enable it, or is it ? The mod_xml_ldap doesn't get built by running make make or dpkg- buildpackage from trunk/ Also I tried building it from the module directory itself, but then I get the following error: fsbuilder at sv:~/trunk/src/mod/xml_int/mod_xml_ldap$ make Compiling mod_xml_ldap.c... cc1: warnings being treated as errors mod_xml_ldap.c: In function 'xml_ldap_search': mod_xml_ldap.c:356: warning: cast from pointer to integer of different size make[1]: *** [mod_xml_ldap.o] Error 1 make: *** [all] Error 1 (Also I had to apt-get install libsasl2 libsasl2-dev, otherwise make from this dir errored with missing sasl/sasl.h) Can you see what I'm doing wrong ? (I'm using svn rev 11560) thanks & regards, Leon On Jan 6, 2009, at 4:55 AM, John Skopis (Lists) wrote: > Vinicius Kobashi wrote: >> hi ppl. >> >> i tried hard to make it work, but still i couldnt find a complete >> openldap scheme that provides these information, and i still could't >> find out where to put these configuration... >> >> can anyone help me? >> >> thankz! >> >> vinicius escreveu: >>> thankz! >>> >>> ill set my openldap to provide these information.. >>> >>> but these about these binding settings... where should i set them? >>> >>> best regards >>> >>> John Skopis (Lists) wrote: >>>> vinicius wrote: >>>> >>>>> hi ppl.. i tried to find something at google, but i couldnt >>>>> manage to find >>>>> anything. >>>>> i still dont know what to do to make the mod_xml_ldap work. >>>>> i couldnt find information about how to build a config file for >>>>> the >>>>> module, and where to store it... >>>>> >>>>> can anyone give me a help? >>>>> >>>>> >>>> >>>> Be advised mod_xml_ldap is probably not production quality and will >>>> undoubtedly change, eventually at least. >>>> >>>> Here is what I used once: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> bindings="configuration"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> which should/probably/might work with ldap objects like these: >>>> >>>> dn: cn=John Skopis,ou=people,dc=example >>>> objectClass: person >>>> objectClass: inetOrgPerson >>>> objectClass: organizationalPerson >>>> objectClass: FreeSWITCH-Exten-Object >>>> objectClass: top >>>> cn: John Skopis >>>> sn: Skopis >>>> givenName: John >>>> FSid: 1001 >>>> FSmailbox: 1001 >>>> FSpassword: 1234 >>>> FSvm-password: 1001 >>>> FSemail-addr: john+fs at skopis.com >>>> FSvm-email-all-messages: TRUE >>>> FSvm-delete-file: TRUE >>>> FSvm-attach-file: TRUE >>>> >>>> dn: SIPIdentityUserName=1001,ou=h350,dc=example >>>> objectClass: person >>>> objectClass: SIPIdentity >>>> objectClass: top >>>> cn: 1001 >>>> sn: 1001 >>>> SIPIdentitySIPURI: sip:1001 at 172.16.75.129 >>>> SIPIdentityRegistrarAddress: 172.16.75.128 >>>> SIPIdentityProxyAddress: 172.16.75.128 >>>> SIPIdentityPassword: 1234 >>>> SIPIdentityUserName: 1001 >>>> SIPIdentityServiceLevel: premium >>>> >>>> > > Again, the module is not production quality. Hopefully I will conjurer > the time and know-how to put something decent together eventually. > > To load configuration for any fs module you need to define the XML > configuration element under the section "configuration". > > A good starting point is the file > $PREFIX/conf/freeswitch.xml > > http://wiki.freeswitch.org/wiki/Freeswitch.xml > > Also take a look at $PREFIX/logs/freeswitch.xml.fsxml > > to load mod_xml_ldap you would need to add something like this to > modules.conf.xml > > > > and create an xml_ldap.conf.xml in > $PREFIX/autoload_configs/xml_ldap.conf.xml > > > ... > > > The ITU is doing some work called h.350: > http://www.itu.int/ITU-T/studygroups/com16/h350/index.html > > Here is what I was working with: > attributetype ( 1.3.6.1.4.1.65535.2.1.1 NAME 'FSid' > DESC 'FreeSWITCH Extension ID' > EQUALITY caseIgnoreIA5Match > SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) > > attributetype ( 1.3.6.1.4.1.65535.2.1.2 NAME 'FSmailbox' > DESC 'FreeSWITCH Extension Mailbox' > EQUALITY caseIgnoreIA5Match > SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) > > attributetype ( 1.3.6.1.4.1.65535.2.1.3 NAME 'FSpassword' > DESC 'FreeSWITCH Password' > EQUALITY caseExactIA5Match > SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 > SINGLE-VALUE ) > > attributetype ( 1.3.6.1.4.1.65535.2.1.4 NAME 'FSa1hash' > DESC 'FreeSWITCH Crypted Password' > EQUALITY caseExactIA5Match > SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 > SINGLE-VALUE ) > > attributetype ( 1.3.6.1.4.1.65535.2.1.5 NAME 'FSvm-password' > DESC 'FreeSWITCH VoiceMail Password' > EQUALITY integerMatch > SYNTAX 1.3.6.1.4.1.1466.115.121.1.27 > SINGLE-VALUE ) > > attributetype ( 1.3.6.1.4.1.65535.2.1.6 NAME 'FSemail-addr' > DESC 'E-mail address to send voicemail' > EQUALITY caseIgnoreIA5Match > SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) > > attributetype ( 1.3.6.1.4.1.65535.2.1.7 NAME 'FSvm-email-all-messages' > DESC 'FreeSWITCH Email All Mesages' > EQUALITY booleanMatch > SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 > SINGLE-VALUE ) > > attributetype ( 1.3.6.1.4.1.65535.2.1.8 NAME 'FSvm-delete-file' > DESC 'FreeSWITCH VoiceMail Delete File' > EQUALITY booleanMatch > SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 > SINGLE-VALUE ) > > attributetype ( 1.3.6.1.4.1.65535.2.1.9 NAME 'FSvm-attach-file' > DESC 'FreeSWITCH VoiceMail Attach file' > EQUALITY booleanMatch > SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 > SINGLE-VALUE ) > > > > > > objectclass ( 1.3.6.1.4.1.65535.2.2.1 NAME 'FreeSWITCH-Exten-Object' > SUP top AUXILIARY > DESC '%obj_desc%' > MUST ( FSid $ FSpassword ) > MAY ( FSmailbox $ FSa1hash $ FSvm-password $ FSemail-addr $ > FSvm-email-all-messages $ FSvm-delete-file $ FSvm-attach-file ) ) > > hth > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sias at cpdata.co.za Fri Jan 30 05:33:15 2009 From: sias at cpdata.co.za (Sias Mey) Date: Fri, 30 Jan 2009 15:33:15 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: References: <20090130103944.GA23536@cpdata.co.za> Message-ID: <20090130133315.GB23536@cpdata.co.za> Hi Brian, Hmmm Ill do some more testing on it later. But I got a destination out of order when I tried. Right now Im busy implementing the string checking. Which seems like it will work out ok, but is clearly not ideal. Thanks for the replay On Fri, Jan 30, 2009 at 04:45:54AM -0600, Brian West wrote: > What wasn't working about this? The {} can be used everywhere without > a problem... Maybe you can provide more details on this. > > /b > > > > > On Jan 30, 2009, at 4:39 AM, Sias Mey wrote: > > > > > I couldent find a way of setting channel variables or executing > > javascript directly on the conference dial since it expects and > > endpoint > > and the {} syntax produced an error. So now I am using the Loopback > > inteface to register some values. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Fri Jan 30 05:49:42 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 30 Jan 2009 14:49:42 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> Message-ID: <49830576.6080907@ewetel.de> Hello, today I uploaded the Q931-To-Pcap patch into openzap's trunk (r628). So you can test it. How to start Q931 to pcap ? In FS just enter "oz q931_pcap on [pcapfilename without suffix]" to start logging q931 packets to pcap. It opens a file called "q931.pcap" or ".pcap". It is saved in FS's log directory. has currently not really an affect to the command. It is only used to make sure that you have at least one valid span configured. Further it is put into 802.1q vlan tag id which is displayed in wireshrak and tshark. Unfortunately I couldn't test it yet (On my side it's always zero). How to stop Q931 pcap? Simply enter "oz q931_pcap off" into FS console. must be valid, but has no affect. Second way is to unload openzap module or shutdown FS. How are the packets saved? All Q931 packets send or received by any span are saved into one file. To see from where to where the packets was send, the FreeSWITCH's side is always marked with ethernet address "02:00:01:AA:AA:AA" and IP address "1.1.1.1" Remote side is always marked with ethernet address "02:00:01:BB:BB:BB" and IP adresss "2.2.2.2" Span ID is intended to be put into VLAN ID, but this is currently not sure. Maybe it's spanid-1 or always zero - I don't know. The pcap timestamp starts with 0 and is increased by each q931 packet. (Maybe a real timestamp is better here) After each saved q931 packet data is flushed into pcap file. This is needed for the small perl script below. How to decode it with wireshark? Get the pcap file from FS log dir and send it via email, ftp or scp to where you have wireshark running. Open it in wireshark. Current wireshark decode the stuff by default as "TPKT - Unknown TPDU type (0x0)". Of course we have a TPKT packet, but wireshark is not able to detect the Q931 packet by default. So just do a right click on such a packet list entry, choose "decode as ..." and click on "do not decode". You can also click on "Decode" and then choose AIM or CFLOW protocol. Yes, AIM is not really Q931 or TPKT, but it works... After applying the packets are decoded as wanted. The black color in the packet list marks some little bugs in the TCP packet generated by this patch. E.g. tcp checksum is zero, but should be vaild. I have code to calculate it, but in my eyes it is an unescessary load for FS. How to decode it with tshark? tshark allows us to decode pcap files right on cli. To do so just enter this: tshark -d tcp.port==102,aim -Rq931 -Ttext -V -r aim is the protocol as what tshark should decode the tcp payload. Some other protocols are working to to get tcp's payload decoded as TPKT with q931 (she so called "Q931 over IP"). Unfortunately it decodes not just q931 but the whole overhead (ethernet,ip,tcp,tpkt) so I build a perl script, which extracts only Q931 packets. For this script I have to flush each Q931 packet into the pcap file, cause this allows to have some kind of real time decoding. You have to start Q931ToPcap logging in FS first, then start the script. You need to have tshark installed for this. The script has the pcap filename incl. path as an optional argument. If not given, it uses the default filename defined within the script. To stop the script press "ctrl+c". Here is the script: #!/usr/bin/perl $default_filename="/opt/app/voip/ippbx/log/q931.pcap"; $display=0; if($#ARGV<0){ $filename=$default_filename; } else{ $filename=$ARGV[0]; } $cmd="tail -n +0 -f ".$filename." | tshark -d tcp.port==102,aim -Rq931 -Ttext -V -i - 2>1|"; print "\n"; open(PCAP, $cmd); while ($line=) { chomp($line); if($line=~/^Frame ([0-9]+) \(/) { $number=$1; } if($line=~/Destination: 02:00:01:aa:aa:aa/i) { $direction=1; } elsif($line=~/Destination: 02:00:01:bb:bb:bb/i) { $direction=0 } elsif($line=~/802.1Q Virtual LAN, PRI: 7, CFI: 0, ID: ([0-9]+)/i) { $spanid=$1; } elsif($line eq "Q.931") { $display=1; $intro=1; next; } elsif(length($line)==0) { $display=0; $intro=0; print "\n\n"; next; } if($display == 1) { if($intro==1) { $mode=$direction?"RECEIVING -----": "SENDING -------"; printf("-- $mode Packet number: %05i --- SpanID: %i ----------------\n", $number, $spanid); $intro=0; } print "$line\n"; } } close(PCAP); regards Helmut From anthony.minessale at gmail.com Fri Jan 30 05:56:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 30 Jan 2009 07:56:55 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <4982EAD5.1010200@laposte.net> References: <4982EAD5.1010200@laposte.net> Message-ID: <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> Which of the 2 machines has the load issue? You said it was one box calling the other. You have 2 major things against you, single CPU and AMD, but you should at least be able to get in the vicinity of 800-1000 calls on a box like that. Are you calling the default 9999? It's not really an appropriate extension for load testing. On the terminating box you should set up a manual extension that is the first one in the dial plan to play a wav file from preferably a ram disk or /tmp If you do plan on using this in production accept nothing less than a multi-core intel machine with at least 4 cores, the more cores the better because that parallel processing is where FS gets it's atvantage. On Fri, Jan 30, 2009 at 5:56 AM, rod wrote: > Dear list, > > I've been playing with freeswitch for some time (2 months) and the fact > is that I'm very pleased with the functionnalities of this software. > > I'd like to use FS as a SBC handling media and I'm doing some tests with > sipp to load the machine but I'm unable to bridge more than 60 calls > without seeing the CPU being loaded at 100%. I'm sure something is going > wrong with my setup but I'm unable to see what. > > The test machine has the following specs: > Athlon XP 3500+ with 2GB of memory (I know this is not a high end > machine :p) > > Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > processor : 0 > vendor_id : AuthenticAMD > cpu family : 15 > model : 95 > model name : AMD Athlon(tm) 64 Processor 3500+ > stepping : 2 > cpu MHz : 2199.973 > cache size : 512 KB > fpu : yes > fpu_exception : yes > cpuid level : 1 > wp : yes > flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge > mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext fxsr_opt > rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm extapic > cr8_legacy > bogomips : 4402.97 > TLB size : 1024 4K pages > clflush size : 64 > cache_alignment : 64 > address sizes : 40 bits physical, 48 bits virtual > power management: ts fid vid ttp tm stc > > I installed FS on a fresh debian 64: > Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 UTC 2009 > x86_64 GNU/Linux > > I set the ulimit parameters like those on the website: > freeswitch at internal> ... > Freeswitch:/opt/free-svn/bin# ulimit -a > core file size (blocks, -c) unlimited > data seg size (kbytes, -d) unlimited > scheduling priority (-e) 0 > file size (blocks, -f) unlimited > pending signals (-i) unlimited > max locked memory (kbytes, -l) unlimited > max memory size (kbytes, -m) unlimited > open files (-n) 999999 > pipe size (512 bytes, -p) 8 > POSIX message queues (bytes, -q) unlimited > real-time priority (-r) 0 > stack size (kbytes, -s) 244 > cpu time (seconds, -t) unlimited > max user processes (-u) unlimited > virtual memory (kbytes, -v) unlimited > file locks (-x) unlimited > > > My network setup is the following: > > SIPP machine (10.10.10.1/24)----------------vlan55 > ----------(10.10.10.254/24) FS (10.10.20.254/24)--------------vlan56 > -------------------(10.10.20.100/24) OTHER STOCK FS > > > I launched sipp with: > sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi 10.10.10.1 -i > 10.10.10.1 -mp 25000 10.10.10.254:5060 > > The dialplan on FS is very simple: > > > > > > > > > data="sofia/external/9999 at 10.10.20.100"/> > > > > > > > FreeSWITCH Version 1.0.trunk (11560M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[100] > SQL [Enabled] > > > The test is very simple: sipp dial 9999 that matches in my FS dialplan > and this is bridged to an other FS machine playing music on hold. > When I launch "top" I see after 30 to 40 s that FS consumes all the CPU > ressources (with a mean of 50-60 % before), with 80 calls. > When I set 70 calls, I have to wait 70-80 s before seeing the same issue. > > Presence is set to false on the 2 profile. > > I have the same issue with FS 1.0.2 that' s why I tried FS 11560. > > When I use the FS machine as a router to test the packet per second > performance, I'm reaching 100Mbps with 8000pps in each direction (from > vlan 55 to vlan56) with less than 12% CPU. So that I don't think there's > an issue with the network. > > Here is an "mpstat -P ALL 1" to show you what's happening suddenly with > 70 bridge calls: > 12:31:26 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 > 0,00 89,00 6241,00 > 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 > 0,00 89,00 6241,00 > > 12:31:27 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 > 0,00 22,22 6035,35 > 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 > 0,00 22,22 6035,35 > > 12:31:28 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 > 0,00 0,00 5483,17 > 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 > 0,00 0,00 5483,17 > > > The CPU is going from 89% idle to 0% in less than 2 seconds. > > I know that I don't have to expect too much from this kind of hardware, > but it seems strange that the CPU power vanished so suddenly. > > Thanks a lot for the guys that have read this long mail :p > > kind regards, > rod > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/44500c78/attachment.html From anthony.minessale at gmail.com Fri Jan 30 06:02:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 30 Jan 2009 08:02:16 -0600 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: References: Message-ID: <191c3a030901300602h520b3f1fn71c9172cac18d5db@mail.gmail.com> The reason it works now is because I checked in the LD change into tree yesterday. It probably would have worked with the old copy of python too but I did not realize we did not make that change yet. But you still have a fresh new python to boot! On Thu, Jan 29, 2009 at 9:27 PM, Traun Leyden wrote: > > I don't know, I must have had a momentary lapse of reason .. but at least I > documented it! > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/96b7f8b9/attachment-0001.html From anthony.minessale at gmail.com Fri Jan 30 06:05:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 30 Jan 2009 08:05:07 -0600 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090130133315.GB23536@cpdata.co.za> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> Message-ID: <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> you should be able to use {} in the dial command you also should be able to do originate {...}sofia/profile/user at domain.comconference:@ inline to the api interface On Fri, Jan 30, 2009 at 7:33 AM, Sias Mey wrote: > Hi Brian, > > Hmmm Ill do some more testing on it later. But I got a destination out > of order when I tried. Right now Im busy implementing the string > checking. Which seems like it will work out ok, but is clearly not > ideal. > > Thanks for the replay > > On Fri, Jan 30, 2009 at 04:45:54AM -0600, Brian West wrote: > > What wasn't working about this? The {} can be used everywhere without > > a problem... Maybe you can provide more details on this. > > > > /b > > > > > > > > > > On Jan 30, 2009, at 4:39 AM, Sias Mey wrote: > > > > > > > > I couldent find a way of setting channel variables or executing > > > javascript directly on the conference dial since it expects and > > > endpoint > > > and the {} syntax produced an error. So now I am using the Loopback > > > inteface to register some values. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/22132945/attachment.html From pmhshz at gmail.com Fri Jan 30 07:29:09 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 30 Jan 2009 07:29:09 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21745312.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> Message-ID: <21749375.post@talk.nabble.com> When freeswitch freezes, we can't connect to it to check sps status, but once we were able to connect and at that time it was showing 0/0 sps. thanks... shehzad p wrote: > > Thanks, Anthony > > In my previous test sps did not changed, > but in recent test sps was dropped to 0 itself (as below). > =============================================================== > UP 0 years, 0 days, 5 hours, 1 minute, 53 seconds, 878 milliseconds, 190 > microseconds > 5474 session(s) since startup > 75 session(s) 0/0 > ============================================================= > > My system is 32 bit. > CPU is Intel(R) Xeon(R) CPU X3220 @ 2.40GHz > And RAM is 4GB > > Output of ulimit -a is: > ulimit -a: (set after first test) > core file size (blocks, -c) unlimited > data seg size (kbytes, -d) unlimited > max nice (-e) 20 > file size (blocks, -f) unlimited > pending signals (-i) unlimited > max locked memory (kbytes, -l) unlimited > max memory size (kbytes, -m) unlimited > open files (-n) 999999 > pipe size (512 bytes, -p) 8 > POSIX message queues (bytes, -q) unlimited > max rt priority (-r) unlimited > stack size (kbytes, -s) 244 > cpu time (seconds, -t) unlimited > max user processes (-u) unlimited > virtual memory (kbytes, -v) unlimited > file locks (-x) unlimited > =================================================== > > > BTW using trunk on production system is safe? > > Warm thanks for kind responses... > > > > Anthony Minessale-2 wrote: >> >> When you get it in that state what do you see when you execute >> >> fsctl sps >> >> is the sps a very low number? >> >> Did the sps drop by itself from the value you originally set it to? >> >> Are you using 32 bit? >> >> if so try all of these commands in your shell before starting FS >> >> ulimit -c unlimited >> ulimit -d unlimited >> ulimit -f unlimited >> ulimit -i unlimited >> ulimit -n 999999 >> ulimit -q unlimited >> ulimit -u unlimited >> ulimit -v unlimited >> ulimit -x unlimited >> ulimit -s 244 >> ulimit -l unlimited >> >> >> DO NOT put them in a script unless you source the script with . >> . myscript or they will be undone instantly when the script exits >> >> BTW, I said to try latest trunk not 1.0.2 We can only debug the >> development >> code at this point. >> >> >> >> >> >> On Thu, Jan 29, 2009 at 10:06 AM, shehzad p wrote: >> >>> >>> Hi Anthony, >>> >>> I found interesting result while testing Freeswitch, and it might be >>> cause >>> of freezing out of freeswitch, >>> >>> I updated my system (as you told) with latest stable version Freeswitch >>> 1.0.2 >>> First of all I set sps to 100, >>> Then I sends call approximately 100 per seconds, Freeswitch works fine >>> and >>> handles all the calls very well. >>> >>> After that I send 130 calls per seconds, and magic happen now, >>> Freeswitch >>> handles first 100 calls only. >>> all the preceding calls were failed (even not appeared in freeswitch >>> console >>> why?) >>> >>> When I put ngrep trace, System responds with 503 Maximum Calls In >>> Progress. >>> (as below) >>> ########################################################### >>> # >>> U FSFSFSFSFS -> GWGWGWGWGW >>> SIP/2.0 503 Maximum Calls In Progress. >>> Via: SIP/2.0/UDP GWGWGWGWGW;branch=z9hG4bK53eafabb;rport=5060. >>> From: "99999" ;tag=as2e10c170. >>> To: ;tag=K3jSUFrDHpmmB. >>> Call-ID: 0feb229e58afe7be17110deb361dc234 at GWGWGWGWGW. >>> CSeq: 102 INVITE. >>> Retry-After: 300. >>> User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported. >>> Accept: application/sdp. >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >>> NOTIFY, >>> REFER, UPDATE, REGISTER, INFO, PUBLISH. >>> Supported: timer, precondition, path, replaces. >>> Allow-Events: talk, presence, dialog, call-info, sla, >>> include-session-description, presence.winfo, message-summary, refer. >>> Content-Length: 0. >>> . >>> ##################################################################### >>> >>> >>> Now another issue to note down is that, >>> After all above happened and active calls comes to zero, >>> I just make a single call which also fails with response 503 - Maximum >>> Calls >>> In Progress. >>> >>> >>> Is this intended behaviour, should I increase SPS to overcome this. or >>> something like bug. >>> >>> Please let me know what should be the resolution for this. >>> >>> Thanks, >>> msp >>> >>> >>> >>> Anthony Minessale-2 wrote: >>> > >>> > Also remember, >>> > Actually completely uninstall and erase /usr/local/freeswitch and the >>> > 1.0.1 >>> > source tree and freshly install the new one. >>> > If you try to upgrade on top of a release with trunk it will cause >>> more >>> > problems for you. >>> > >>> > >>> > On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice >>> wrote: >>> > >>> >> Upgrade to trunk... Many many issues have been resolved since 1.0.1 >>> was >>> >> the >>> >> current release >>> >> >>> >> >>> >> > From: shehzad p >>> >> > Reply-To: >>> >> > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) >>> >> > To: >>> >> > Subject: [Freeswitch-users] Freeswitch freezes on increasing call >>> >> traffic >>> >> > >>> >> > >>> >> > Hi all, >>> >> > >>> >> > Yesterday my Freeswitch server faced a problem when call traffic >>> >> increased >>> >> > to more than 100. >>> >> > >>> >> > When I start Freeswitch, it works fine and then after some time >>> >> > (approximately 15 to 20 minutes) it stops functioning (means no >>> call >>> >> is >>> >> > being processed, no CLI command is working and it just freezes) >>> until >>> I >>> >> > restart the freeswitch. >>> >> > >>> >> > I am using Freeswitch 1.0.1. >>> >> > Debug (gdb) trace as on wiki page >>> >> > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.sh is >>> >> attached >>> >> > http://www.nabble.com/file/p21701744/fs_debgu.txt fs_debgu.txt >>> >> > -- >>> >> > View this message in context: >>> >> > >>> >> >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 >>> >> > p21701744.html >>> >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >> > >>> >> > >>> >> > _______________________________________________ >>> >> > Freeswitch-users mailing list >>> >> > Freeswitch-users at 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 at 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 at hotmail.com >>> < >>> MSN%3Aanthony_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 >>> < >>> sip%3A888 at conference.freeswitch.org >>> > >>> > iax:guest at conference.freeswitch.org/888 >>> > >>> googletalk:conf+888 at conference.freeswitch.org >>> >>> > >>> > pstn:213-799-1400 >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21729863.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 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 >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21749375.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Fri Jan 30 08:18:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 30 Jan 2009 10:18:13 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21749375.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> Message-ID: <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> if you are using unix you can use the supplied script scripts/freeswitch-gcore to capture a copy of the resident memory and I can have a look perhaps. Trunk is safe for production as we are in beta stage for a release of 1.0.3 at this time. On Fri, Jan 30, 2009 at 9:29 AM, shehzad p wrote: > > When freeswitch freezes, we can't connect to it to check sps status, > but once we were able to connect and at that time it was showing 0/0 sps. > > thanks... > > shehzad p wrote: > > > > Thanks, Anthony > > > > In my previous test sps did not changed, > > but in recent test sps was dropped to 0 itself (as below). > > =============================================================== > > UP 0 years, 0 days, 5 hours, 1 minute, 53 seconds, 878 milliseconds, 190 > > microseconds > > 5474 session(s) since startup > > 75 session(s) 0/0 > > ============================================================= > > > > My system is 32 bit. > > CPU is Intel(R) Xeon(R) CPU X3220 @ 2.40GHz > > And RAM is 4GB > > > > Output of ulimit -a is: > > ulimit -a: (set after first test) > > core file size (blocks, -c) unlimited > > data seg size (kbytes, -d) unlimited > > max nice (-e) 20 > > file size (blocks, -f) unlimited > > pending signals (-i) unlimited > > max locked memory (kbytes, -l) unlimited > > max memory size (kbytes, -m) unlimited > > open files (-n) 999999 > > pipe size (512 bytes, -p) 8 > > POSIX message queues (bytes, -q) unlimited > > max rt priority (-r) unlimited > > stack size (kbytes, -s) 244 > > cpu time (seconds, -t) unlimited > > max user processes (-u) unlimited > > virtual memory (kbytes, -v) unlimited > > file locks (-x) unlimited > > =================================================== > > > > > > BTW using trunk on production system is safe? > > > > Warm thanks for kind responses... > > > > > > > > Anthony Minessale-2 wrote: > >> > >> When you get it in that state what do you see when you execute > >> > >> fsctl sps > >> > >> is the sps a very low number? > >> > >> Did the sps drop by itself from the value you originally set it to? > >> > >> Are you using 32 bit? > >> > >> if so try all of these commands in your shell before starting FS > >> > >> ulimit -c unlimited > >> ulimit -d unlimited > >> ulimit -f unlimited > >> ulimit -i unlimited > >> ulimit -n 999999 > >> ulimit -q unlimited > >> ulimit -u unlimited > >> ulimit -v unlimited > >> ulimit -x unlimited > >> ulimit -s 244 > >> ulimit -l unlimited > >> > >> > >> DO NOT put them in a script unless you source the script with . > >> . myscript or they will be undone instantly when the script exits > >> > >> BTW, I said to try latest trunk not 1.0.2 We can only debug the > >> development > >> code at this point. > >> > >> > >> > >> > >> > >> On Thu, Jan 29, 2009 at 10:06 AM, shehzad p wrote: > >> > >>> > >>> Hi Anthony, > >>> > >>> I found interesting result while testing Freeswitch, and it might be > >>> cause > >>> of freezing out of freeswitch, > >>> > >>> I updated my system (as you told) with latest stable version Freeswitch > >>> 1.0.2 > >>> First of all I set sps to 100, > >>> Then I sends call approximately 100 per seconds, Freeswitch works fine > >>> and > >>> handles all the calls very well. > >>> > >>> After that I send 130 calls per seconds, and magic happen now, > >>> Freeswitch > >>> handles first 100 calls only. > >>> all the preceding calls were failed (even not appeared in freeswitch > >>> console > >>> why?) > >>> > >>> When I put ngrep trace, System responds with 503 Maximum Calls In > >>> Progress. > >>> (as below) > >>> ########################################################### > >>> # > >>> U FSFSFSFSFS -> GWGWGWGWGW > >>> SIP/2.0 503 Maximum Calls In Progress. > >>> Via: SIP/2.0/UDP GWGWGWGWGW;branch=z9hG4bK53eafabb;rport=5060. > >>> From: "99999" ;tag=as2e10c170. > >>> To: ;tag=K3jSUFrDHpmmB. > >>> Call-ID: 0feb229e58afe7be17110deb361dc234 at GWGWGWGWGW. > >>> CSeq: 102 INVITE. > >>> Retry-After: 300. > >>> User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported. > >>> Accept: application/sdp. > >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > >>> NOTIFY, > >>> REFER, UPDATE, REGISTER, INFO, PUBLISH. > >>> Supported: timer, precondition, path, replaces. > >>> Allow-Events: talk, presence, dialog, call-info, sla, > >>> include-session-description, presence.winfo, message-summary, refer. > >>> Content-Length: 0. > >>> . > >>> ##################################################################### > >>> > >>> > >>> Now another issue to note down is that, > >>> After all above happened and active calls comes to zero, > >>> I just make a single call which also fails with response 503 - Maximum > >>> Calls > >>> In Progress. > >>> > >>> > >>> Is this intended behaviour, should I increase SPS to overcome this. or > >>> something like bug. > >>> > >>> Please let me know what should be the resolution for this. > >>> > >>> Thanks, > >>> msp > >>> > >>> > >>> > >>> Anthony Minessale-2 wrote: > >>> > > >>> > Also remember, > >>> > Actually completely uninstall and erase /usr/local/freeswitch and the > >>> > 1.0.1 > >>> > source tree and freshly install the new one. > >>> > If you try to upgrade on top of a release with trunk it will cause > >>> more > >>> > problems for you. > >>> > > >>> > > >>> > On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice > >>> wrote: > >>> > > >>> >> Upgrade to trunk... Many many issues have been resolved since 1.0.1 > >>> was > >>> >> the > >>> >> current release > >>> >> > >>> >> > >>> >> > From: shehzad p > >>> >> > Reply-To: > >>> >> > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) > >>> >> > To: > >>> >> > Subject: [Freeswitch-users] Freeswitch freezes on increasing call > >>> >> traffic > >>> >> > > >>> >> > > >>> >> > Hi all, > >>> >> > > >>> >> > Yesterday my Freeswitch server faced a problem when call traffic > >>> >> increased > >>> >> > to more than 100. > >>> >> > > >>> >> > When I start Freeswitch, it works fine and then after some time > >>> >> > (approximately 15 to 20 minutes) it stops functioning (means no > >>> call > >>> >> is > >>> >> > being processed, no CLI command is working and it just freezes) > >>> until > >>> I > >>> >> > restart the freeswitch. > >>> >> > > >>> >> > I am using Freeswitch 1.0.1. > >>> >> > Debug (gdb) trace as on wiki page > >>> >> > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.shis > >>> >> attached > >>> >> > http://www.nabble.com/file/p21701744/fs_debgu.txt fs_debgu.txt > >>> >> > -- > >>> >> > View this message in context: > >>> >> > > >>> >> > >>> > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 > >>> >> > p21701744.html > >>> >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. > >>> >> > > >>> >> > > >>> >> > _______________________________________________ > >>> >> > Freeswitch-users mailing list > >>> >> > Freeswitch-users at 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 at 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 at hotmail.com > >>> > >< > >>> MSN%3Aanthony_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 > >>> > >< > >>> sip%3A888 at conference.freeswitch.org > > > > >>> > > >>> > iax:guest at conference.freeswitch.org/888 > >>> > > >>> googletalk:conf+888 at conference.freeswitch.org > > > > >>> > > > > >>> > > >>> > pstn:213-799-1400 > >>> > > >>> > _______________________________________________ > >>> > Freeswitch-users mailing list > >>> > Freeswitch-users at lists.freeswitch.org > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> > http://www.freeswitch.org > >>> > > >>> > > >>> > >>> -- > >>> View this message in context: > >>> > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21729863.html > >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at 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 at hotmail.com < > MSN%3Aanthony_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 < > sip%3A888 at conference.freeswitch.org > > > >> iax:guest at conference.freeswitch.org/888 > >> googletalk:conf+888 at conference.freeswitch.org > > > > >> pstn:213-799-1400 > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21749375.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/a2e93bfb/attachment-0001.html From peder at networkoblivion.com Fri Jan 30 09:05:29 2009 From: peder at networkoblivion.com (peder at networkoblivion.com) Date: Fri, 30 Jan 2009 11:05:29 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> References: <4982EAD5.1010200@laposte.net> <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> Message-ID: <49833359.6020603@networkoblivion.com> Given the choice between dual core and dual processor, would one work better than the other for FS? Assuming roughly the same speed. Or would it not really have a measurable difference assuming same speed and both Intel? Anthony Minessale wrote: > Which of the 2 machines has the load issue? You said it was one box > calling the other. > > You have 2 major things against you, single CPU and AMD, but you should > at least be able to get in the vicinity of 800-1000 calls on a box like > that. > > Are you calling the default 9999? It's not really an appropriate > extension for load testing. > On the terminating box you should set up a manual extension that is the > first one in the dial plan > to play a wav file from preferably a ram disk or /tmp > > If you do plan on using this in production accept nothing less than a > multi-core intel machine with at least 4 cores, the more cores the > better because that parallel processing is where FS gets it's atvantage. > > > > On Fri, Jan 30, 2009 at 5:56 AM, rod > wrote: > > Dear list, > > I've been playing with freeswitch for some time (2 months) and the fact > is that I'm very pleased with the functionnalities of this software. > > I'd like to use FS as a SBC handling media and I'm doing some tests with > sipp to load the machine but I'm unable to bridge more than 60 calls > without seeing the CPU being loaded at 100%. I'm sure something is going > wrong with my setup but I'm unable to see what. > > The test machine has the following specs: > Athlon XP 3500+ with 2GB of memory (I know this is not a high end > machine :p) > > Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > processor : 0 > vendor_id : AuthenticAMD > cpu family : 15 > model : 95 > model name : AMD Athlon(tm) 64 Processor 3500+ > stepping : 2 > cpu MHz : 2199.973 > cache size : 512 KB > fpu : yes > fpu_exception : yes > cpuid level : 1 > wp : yes > flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge > mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext fxsr_opt > rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm extapic > cr8_legacy > bogomips : 4402.97 > TLB size : 1024 4K pages > clflush size : 64 > cache_alignment : 64 > address sizes : 40 bits physical, 48 bits virtual > power management: ts fid vid ttp tm stc > > I installed FS on a fresh debian 64: > Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 UTC 2009 > x86_64 GNU/Linux > > I set the ulimit parameters like those on the website: > freeswitch at internal> ... > Freeswitch:/opt/free-svn/bin# ulimit -a > core file size (blocks, -c) unlimited > data seg size (kbytes, -d) unlimited > scheduling priority (-e) 0 > file size (blocks, -f) unlimited > pending signals (-i) unlimited > max locked memory (kbytes, -l) unlimited > max memory size (kbytes, -m) unlimited > open files (-n) 999999 > pipe size (512 bytes, -p) 8 > POSIX message queues (bytes, -q) unlimited > real-time priority (-r) 0 > stack size (kbytes, -s) 244 > cpu time (seconds, -t) unlimited > max user processes (-u) unlimited > virtual memory (kbytes, -v) unlimited > file locks (-x) unlimited > > > My network setup is the following: > > SIPP machine (10.10.10.1/24)----------------vlan > 55 > ----------(10.10.10.254/24 ) FS > (10.10.20.254/24)-------------- > vlan56 > -------------------(10.10.20.100/24 ) OTHER > STOCK FS > > > I launched sipp with: > sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi 10.10.10.1 -i > 10.10.10.1 -mp 25000 10.10.10.254:5060 > > The dialplan on FS is very simple: > > > > > > > > > data="sofia/external/9999 at 10.10.20.100 "/> > > > > > > > FreeSWITCH Version 1.0.trunk (11560M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[100] > SQL [Enabled] > > > The test is very simple: sipp dial 9999 that matches in my FS dialplan > and this is bridged to an other FS machine playing music on hold. > When I launch "top" I see after 30 to 40 s that FS consumes all the CPU > ressources (with a mean of 50-60 % before), with 80 calls. > When I set 70 calls, I have to wait 70-80 s before seeing the same > issue. > > Presence is set to false on the 2 profile. > > I have the same issue with FS 1.0.2 that' s why I tried FS 11560. > > When I use the FS machine as a router to test the packet per second > performance, I'm reaching 100Mbps with 8000pps in each direction (from > vlan 55 to vlan56) with less than 12% CPU. So that I don't think there's > an issue with the network. > > Here is an "mpstat -P ALL 1" to show you what's happening suddenly with > 70 bridge calls: > 12:31:26 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 > 0,00 89,00 6241,00 > 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 > 0,00 89,00 6241,00 > > 12:31:27 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 > 0,00 22,22 6035,35 > 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 > 0,00 22,22 6035,35 > > 12:31:28 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 > 0,00 0,00 5483,17 > 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 > 0,00 0,00 5483,17 > > > The CPU is going from 89% idle to 0% in less than 2 seconds. > > I know that I don't have to expect too much from this kind of hardware, > but it seems strange that the CPU power vanished so suddenly. > > Thanks a lot for the guys that have read this long mail :p > > kind regards, > rod > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Fri Jan 30 09:23:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 30 Jan 2009 11:23:46 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <49833359.6020603@networkoblivion.com> References: <4982EAD5.1010200@laposte.net> <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> <49833359.6020603@networkoblivion.com> Message-ID: <191c3a030901300923t409948ebjbf8f51ceb3d5bd65@mail.gmail.com> I have not compared them directly to be able to answer. I do know that the more cores the better (even slower ones) because each core divides the entire workload of the scheduler and increases the total threads that can work in parallel. also you can start FS with -hp (high performance) which activates the round-robin scheduler, increases some ulimits and forces all consumed memory to be permanently resident with mlockall (no swapping) On Fri, Jan 30, 2009 at 11:05 AM, peder at networkoblivion.com < peder at networkoblivion.com> wrote: > Given the choice between dual core and dual processor, would one work > better than the other for FS? Assuming roughly the same speed. Or > would it not really have a measurable difference assuming same speed and > both Intel? > > Anthony Minessale wrote: > > Which of the 2 machines has the load issue? You said it was one box > > calling the other. > > > > You have 2 major things against you, single CPU and AMD, but you should > > at least be able to get in the vicinity of 800-1000 calls on a box like > > that. > > > > Are you calling the default 9999? It's not really an appropriate > > extension for load testing. > > On the terminating box you should set up a manual extension that is the > > first one in the dial plan > > to play a wav file from preferably a ram disk or /tmp > > > > If you do plan on using this in production accept nothing less than a > > multi-core intel machine with at least 4 cores, the more cores the > > better because that parallel processing is where FS gets it's atvantage. > > > > > > > > On Fri, Jan 30, 2009 at 5:56 AM, rod > > wrote: > > > > Dear list, > > > > I've been playing with freeswitch for some time (2 months) and the > fact > > is that I'm very pleased with the functionnalities of this software. > > > > I'd like to use FS as a SBC handling media and I'm doing some tests > with > > sipp to load the machine but I'm unable to bridge more than 60 calls > > without seeing the CPU being loaded at 100%. I'm sure something is > going > > wrong with my setup but I'm unable to see what. > > > > The test machine has the following specs: > > Athlon XP 3500+ with 2GB of memory (I know this is not a high end > > machine :p) > > > > Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > > processor : 0 > > vendor_id : AuthenticAMD > > cpu family : 15 > > model : 95 > > model name : AMD Athlon(tm) 64 Processor 3500+ > > stepping : 2 > > cpu MHz : 2199.973 > > cache size : 512 KB > > fpu : yes > > fpu_exception : yes > > cpuid level : 1 > > wp : yes > > flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr > pge > > mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext > fxsr_opt > > rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm extapic > > cr8_legacy > > bogomips : 4402.97 > > TLB size : 1024 4K pages > > clflush size : 64 > > cache_alignment : 64 > > address sizes : 40 bits physical, 48 bits virtual > > power management: ts fid vid ttp tm stc > > > > I installed FS on a fresh debian 64: > > Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 UTC 2009 > > x86_64 GNU/Linux > > > > I set the ulimit parameters like those on the website: > > freeswitch at internal> ... > > Freeswitch:/opt/free-svn/bin# ulimit -a > > core file size (blocks, -c) unlimited > > data seg size (kbytes, -d) unlimited > > scheduling priority (-e) 0 > > file size (blocks, -f) unlimited > > pending signals (-i) unlimited > > max locked memory (kbytes, -l) unlimited > > max memory size (kbytes, -m) unlimited > > open files (-n) 999999 > > pipe size (512 bytes, -p) 8 > > POSIX message queues (bytes, -q) unlimited > > real-time priority (-r) 0 > > stack size (kbytes, -s) 244 > > cpu time (seconds, -t) unlimited > > max user processes (-u) unlimited > > virtual memory (kbytes, -v) unlimited > > file locks (-x) unlimited > > > > > > My network setup is the following: > > > > SIPP machine (10.10.10.1/24)----------------vlan > > 55 > > ----------(10.10.10.254/24 ) FS > > (10.10.20.254/24)-------------- > > vlan56 > > -------------------(10.10.20.100/24 ) OTHER > > STOCK FS > > > > > > I launched sipp with: > > sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi 10.10.10.1 -i > > 10.10.10.1 -mp 25000 10.10.10.254:5060 > > > > The dialplan on FS is very simple: > > > > > > > > > > > > > > > > > > > data="sofia/external/9999 at 10.10.20.100 "/> > > > > > > > > > > > > > > FreeSWITCH Version 1.0.trunk (11560M) Started. > > Crash Protection [Disabled] > > Max Sessions[1000] > > Session Rate[100] > > SQL [Enabled] > > > > > > The test is very simple: sipp dial 9999 that matches in my FS > dialplan > > and this is bridged to an other FS machine playing music on hold. > > When I launch "top" I see after 30 to 40 s that FS consumes all the > CPU > > ressources (with a mean of 50-60 % before), with 80 calls. > > When I set 70 calls, I have to wait 70-80 s before seeing the same > > issue. > > > > Presence is set to false on the 2 profile. > > > > I have the same issue with FS 1.0.2 that' s why I tried FS 11560. > > > > When I use the FS machine as a router to test the packet per second > > performance, I'm reaching 100Mbps with 8000pps in each direction > (from > > vlan 55 to vlan56) with less than 12% CPU. So that I don't think > there's > > an issue with the network. > > > > Here is an "mpstat -P ALL 1" to show you what's happening suddenly > with > > 70 bridge calls: > > 12:31:26 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 > > 0,00 89,00 6241,00 > > 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 > > 0,00 89,00 6241,00 > > > > 12:31:27 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 > > 0,00 22,22 6035,35 > > 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 > > 0,00 22,22 6035,35 > > > > 12:31:28 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 > > 0,00 0,00 5483,17 > > 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 > > 0,00 0,00 5483,17 > > > > > > The CPU is going from 89% idle to 0% in less than 2 seconds. > > > > I know that I don't have to expect too much from this kind of > hardware, > > but it seems strange that the CPU power vanished so suddenly. > > > > Thanks a lot for the guys that have read this long mail :p > > > > kind regards, > > rod > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 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 > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/bcb25a21/attachment-0001.html From bdeacon at highergear.com Fri Jan 30 12:34:04 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Fri, 30 Jan 2009 12:34:04 -0800 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <3065CCBA-1C35-4117-811A-0E9F6A5ABCAB@freeswitch.org> References: <191c3a030901291722y7da8d65fid824c966f91a61d5@mail.gmail.com> <1233282446.4757.99.camel@dev03.cal.highergear.com> <3065CCBA-1C35-4117-811A-0E9F6A5ABCAB@freeswitch.org> Message-ID: <1233347644.4757.112.camel@dev03.cal.highergear.com> I would love to give back in return for the help you guys have given me, but I'm not sure what would be valuable. In terms of what it took to get my stuff to work correctly, it just came down to Anthony checking in the code fix. My other (failed) attempts to work against a differently compiled Python weren't necessary. Moreover, it now works with the python 2.4 I have on that machine as opposed to an "official" 2.5 python package I was going to replace it with. So I've updated the part in the wiki about this bug to basically just say "Fixed in the code now": http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2F...2Fdatetime.so:_undefined_symbol:_PyExc_IOError and http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2Fusr.2Flib.2F...2Fdatetime.so:_undefinedsymbol:__Py_ZeroStruct What maybe isn't on the wiki (or at least I didn't see it) is that the only way I was able to see the error output was to run freeswitch from gdb. Would I have seen that log output by bumping up the loglevel from fs_cli? But otherwise what I'm doing so far is pretty generic to the other examples: And mymodule is just a standard fsapi(session, stream, env, args) The only "tricky" thing was that the code called from fsapi was calling out to a postgresql database via sqlalchemy. But from python's point of view, there's nothing clever about talking to databases, and from freeswitch's point of view there's nothing clever about calling a python module's fsapi function. I'll poke around and see if there's anything in the mod_python area that I could improve on, but if you could point out anything worthwhile that I just accomplished (or rather, that y'all accomplished for me) I'd be more than happy to document it. But near as I can tell, all I've really done is help you guys fix the bug for which I was the loudest complainer. :) Brian On Thu, 2009-01-29 at 20:34 -0600, Brian West wrote: > Can you do some examples and documentation on the wiki about what > you're doing to maybe help others? > > /b > > On Jan 29, 2009, at 8:27 PM, Brian Deacon wrote: > > > TA-DA! > > > > My python can now not only import the sqlalchemy module, but the > > code I > > had before that was actually doing some database interaction is > > working > > now. > > > > Thank you very much for all the help. It is much appreciated. > > > > It sounds like the contents on the wiki with the modules.conf.xml or > > the > > LD workarounds are superceded now. Should I change the comments on > > there to reflect that it should now be fixed? > > > > Brian > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdeacon at highergear.com Fri Jan 30 12:54:14 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Fri, 30 Jan 2009 12:54:14 -0800 Subject: [Freeswitch-users] Is pyrun available from mod_commands (and thus to mod_xml_rpc?) Message-ID: <1233348854.4757.116.camel@dev03.cal.highergear.com> Maybe this is just missing documentation for mod_commands. I can run pyrun from fs_cli. Does that mean I can run it from mod_xml_rpc? Brian From bdeacon at highergear.com Fri Jan 30 13:01:10 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Fri, 30 Jan 2009 13:01:10 -0800 Subject: [Freeswitch-users] Answering my own question (Was: Is pyrun available from mod_commands (and thus to mod_xml_rpc?)) In-Reply-To: <1233348854.4757.116.camel@dev03.cal.highergear.com> References: <1233348854.4757.116.camel@dev03.cal.highergear.com> Message-ID: <1233349270.4757.119.camel@dev03.cal.highergear.com> http://freeswitch:works at myfreeswitchbox:8080/webapi/help On Fri, 2009-01-30 at 12:54 -0800, Brian Deacon wrote: > Maybe this is just missing documentation for mod_commands. I can run > pyrun from fs_cli. Does that mean I can run it from mod_xml_rpc? > > Brian From bdeacon at highergear.com Fri Jan 30 13:05:49 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Fri, 30 Jan 2009 13:05:49 -0800 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <191c3a030901300602h520b3f1fn71c9172cac18d5db@mail.gmail.com> References: <191c3a030901300602h520b3f1fn71c9172cac18d5db@mail.gmail.com> Message-ID: <1233349549.4757.121.camel@dev03.cal.highergear.com> Actually, I punted and it's working with a generic python 2.4.something. But having just found out that installall is the old lameness, maybe that was the reason for my linking problems when trying to compile against a niftier python. I'll try that trick when next I hit a roadblock. On Fri, 2009-01-30 at 08:02 -0600, Anthony Minessale wrote: > The reason it works now is because I checked in the LD change into > tree yesterday. > It probably would have worked with the old copy of python too but I > did not realize we did not make that change yet. > > But you still have a fresh new python to boot! > > On Thu, Jan 29, 2009 at 9:27 PM, Traun Leyden > wrote: > > I don't know, I must have had a momentary lapse of reason .. > but at least I documented it! > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch-users at digitaldan.com Fri Jan 30 13:35:17 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Fri, 30 Jan 2009 14:35:17 -0700 (MST) Subject: [Freeswitch-users] compiling mod_openzap Message-ID: <29266376.201233351317867.JavaMail.root@zimbra> Hi guys I pulled FS from trunk today to build some debian packages and ran into this error when building mod_openzap: mod_openzap.c:2197: warning: assignment discards qualifiers from pointer target type I went ahead and changed the line at 2178 in mod_openzap.c from char *pcapfn = NULL; to const char *pcapfn = NULL; Do you see any problems with this change? Dan- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/64488697/attachment.html From anthony.minessale at gmail.com Fri Jan 30 13:54:49 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 30 Jan 2009 15:54:49 -0600 Subject: [Freeswitch-users] Is pyrun available from mod_commands (and thus to mod_xml_rpc?) In-Reply-To: <1233348854.4757.116.camel@dev03.cal.highergear.com> References: <1233348854.4757.116.camel@dev03.cal.highergear.com> Message-ID: <191c3a030901301354u235ee33br842ed953813a25cb@mail.gmail.com> yes you can On Fri, Jan 30, 2009 at 2:54 PM, Brian Deacon wrote: > Maybe this is just missing documentation for mod_commands. I can run > pyrun from fs_cli. Does that mean I can run it from mod_xml_rpc? > > Brian > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/9be0d3c0/attachment.html From anthony.minessale at gmail.com Fri Jan 30 13:56:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 30 Jan 2009 15:56:05 -0600 Subject: [Freeswitch-users] More troubles with SQLAlchemy and mod_python In-Reply-To: <1233349549.4757.121.camel@dev03.cal.highergear.com> References: <191c3a030901300602h520b3f1fn71c9172cac18d5db@mail.gmail.com> <1233349549.4757.121.camel@dev03.cal.highergear.com> Message-ID: <191c3a030901301356u377665f2lf8e936b71de1bad7@mail.gmail.com> just wikify what you learn and drop by irc and say hi and we can find ways for you to help if you are interested. On Fri, Jan 30, 2009 at 3:05 PM, Brian Deacon wrote: > Actually, I punted and it's working with a generic python 2.4.something. > But having just found out that installall is the old lameness, maybe > that was the reason for my linking problems when trying to compile > against a niftier python. > > I'll try that trick when next I hit a roadblock. > > On Fri, 2009-01-30 at 08:02 -0600, Anthony Minessale wrote: > > The reason it works now is because I checked in the LD change into > > tree yesterday. > > It probably would have worked with the old copy of python too but I > > did not realize we did not make that change yet. > > > > But you still have a fresh new python to boot! > > > > On Thu, Jan 29, 2009 at 9:27 PM, Traun Leyden > > wrote: > > > > I don't know, I must have had a momentary lapse of reason .. > > but at least I documented it! > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 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 > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/6779fced/attachment-0001.html From mike at jerris.com Fri Jan 30 14:31:24 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 30 Jan 2009 17:31:24 -0500 Subject: [Freeswitch-users] compiling mod_openzap In-Reply-To: <29266376.201233351317867.JavaMail.root@zimbra> References: <29266376.201233351317867.JavaMail.root@zimbra> Message-ID: <35AA95A0-1CEF-4ABD-AB50-3B6FF8EE9628@jerris.com> This is fixed in svn trunk now, I will have to do a little bit of autoconf work to get the build w/o libpcap working again but expect that this weekend. Mike On Jan 30, 2009, at 4:35 PM, Dan wrote: > Hi guys I pulled FS from trunk today to build some debian packages > and ran into this error when building mod_openzap: > > mod_openzap.c:2197: warning: assignment discards qualifiers from > pointer target type > > I went ahead and changed the line at 2178 in mod_openzap.c from > > char *pcapfn = NULL; > to > const char *pcapfn = NULL; > > Do you see any problems with this change? > > Dan- > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/19f50521/attachment.html From bdeacon at highergear.com Fri Jan 30 16:55:41 2009 From: bdeacon at highergear.com (Brian Deacon) Date: Fri, 30 Jan 2009 16:55:41 -0800 Subject: [Freeswitch-users] Is Session.uuid the same as caller_profile/uuid in the cdr.xml? Message-ID: <1233363341.6872.4.camel@dev03.cal.highergear.com> Hiya, So my xml cdr has just one uuid in it. (I guess the shape and contents of this are configurable? But I'm using however it defaults.) cdr/callflow/caller_profile/uuid I wouldn't have thought caller_profile would be where I'd find a session.uuid, but since that seems to be the only uuid in the cdr, I'm guessing that's it if it's there. Is that it? I'm going to be logging the start of a phone call using the session uuid as the PK for a log record and need to be able to update that record later with call duration information when the CDR comes through after hangup. Brian From Daniell at airg.com Fri Jan 30 17:17:49 2009 From: Daniell at airg.com (Daniel Liang) Date: Fri, 30 Jan 2009 17:17:49 -0800 Subject: [Freeswitch-users] How to break a playback with loops In-Reply-To: <191c3a030901291354t5cc16a22h32b5defb755c1d3c@mail.gmail.com> References: <0B02E756F603CC409EB553879B090CC80A1B6DC2@HPEXCHVS01.exchange.airg><87f2f3b90901291334i27052568yc90a83d250cc1b76@mail.gmail.com> <191c3a030901291354t5cc16a22h32b5defb755c1d3c@mail.gmail.com> Message-ID: <0B02E756F603CC409EB553879B090CC80A1B72BA@HPEXCHVS01.exchange.airg> It works. Thank you. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: January 29, 2009 1:54 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How to break a playback with loops The break function with the all app is better for this api break all This lets you still break even if the playback is in event lock I did add to trunk a patch so you can do it with the app you can now add "all" as the app args to break completely. On Thu, Jan 29, 2009 at 3:34 PM, Michael Collins wrote: Could you please file a bug report on this? jira.freeswitch.org. Also, follow the bug reporting guidelines here: http://wiki.freeswitch.org/wiki/Reporting_Bugs Thanks, MC On Thu, Jan 29, 2009 at 11:05 AM, Daniel Liang wrote: > Hi, > > I am trying to playing back a wav file a few times, and on receiving a dtmf > event, it stops playing the file. The command I sent is: > > sendmsg > call-command: execute > execute-app-name: playback > execute-app-arg: > loops: 4 > > On receiving a dtmf event, I send a break command: > > sendmsg > call-command: execute > execute-app-name: break > > However, that only breaks one loop of the playback. And I have to press a > dtmf for 4 times to stop the file. > > I tried sending 4 break commands all at once. But it didn't work. And even > worse, my program stops receiving dtmfs anymore. My guess is that the first > break stops the first loop, but before the second loop starts, the following > 3 breaks stop something else that I don't know what they are. > > Any help is appreciated. > > Thanks. > Daniel Liang > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/17ca7b75/attachment.html From marc at kasteris.com Fri Jan 30 17:41:15 2009 From: marc at kasteris.com (Marc Orenberg) Date: Fri, 30 Jan 2009 17:41:15 -0800 (PST) Subject: [Freeswitch-users] Spanish prompts for mod_say Message-ID: <131558.29126.qm@web50802.mail.re2.yahoo.com> Does anybody know where I can find Spanish prompts for mod_say? I see the English ones in http://files.freeswitch.org, but no other languages. Thanks, Marc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/7c49d1ca/attachment.html From msc at freeswitch.org Fri Jan 30 18:08:15 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 30 Jan 2009 18:08:15 -0800 Subject: [Freeswitch-users] Spanish prompts for mod_say In-Reply-To: <131558.29126.qm@web50802.mail.re2.yahoo.com> References: <131558.29126.qm@web50802.mail.re2.yahoo.com> Message-ID: We don't have any yet but we'd love to get some. If someone is willing to donate the money then we could have GM Voices do them. Barring that, if someone has a voice talent who can record the Spanish prompts and donate them to the project then I'm sure they would be graciously accepted and added to the repo. -MC Sent from my iPhone On Jan 30, 2009, at 5:41 PM, Marc Orenberg wrote: > Does anybody know where I can find Spanish prompts for mod_say? > I see the English ones in http://files.freeswitch.org, but no other > languages. > Thanks, > Marc > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/6b14385c/attachment.html From mike at jerris.com Fri Jan 30 18:10:37 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 30 Jan 2009 21:10:37 -0500 Subject: [Freeswitch-users] Spanish prompts for mod_say In-Reply-To: <131558.29126.qm@web50802.mail.re2.yahoo.com> References: <131558.29126.qm@web50802.mail.re2.yahoo.com> Message-ID: <0C7930AF-8A60-488D-8E0D-289B8D22A8BC@jerris.com> No one has spent the time to even update the translations on this or to come up with the money to have them recorded. If you can get a group together to fund the recordings, we would be happy to host them. We work with a company called gmvoices for our recordings and would like to continue working with them for other languages as well. If you can pick out a preferred voice from http://gmvoices.com/listen and get some people to pool in the cash, we would love to get this done. I full set of recordings for all that we require will probably cost $2000-$3000. If anyone is interested in funding this, please contact me offlist so we can discuss. Mike On Jan 30, 2009, at 8:41 PM, Marc Orenberg wrote: > Does anybody know where I can find Spanish prompts for mod_say? > I see the English ones in http://files.freeswitch.org, but no other > languages. > Thanks, > Marc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090130/d49c619b/attachment-0001.html From msc at freeswitch.org Fri Jan 30 18:11:10 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 30 Jan 2009 18:11:10 -0800 Subject: [Freeswitch-users] Is Session.uuid the same as caller_profile/uuid in the cdr.xml? In-Reply-To: <1233363341.6872.4.camel@dev03.cal.highergear.com> References: <1233363341.6872.4.camel@dev03.cal.highergear.com> Message-ID: <71541DBC-DA0E-402E-A097-1F1F17976496@freeswitch.org> Turn on XML CDR and try that. Compare the variables in an XML record to the same CSV record. You'll be able to see them names of the variables you want to add. Or just use XML :) -MC Sent from my iPhone On Jan 30, 2009, at 4:55 PM, Brian Deacon wrote: > Hiya, > > So my xml cdr has just one uuid in it. (I guess the shape and > contents > of this are configurable? But I'm using however it defaults.) > > cdr/callflow/caller_profile/uuid > > I wouldn't have thought caller_profile would be where I'd find a > session.uuid, but since that seems to be the only uuid in the cdr, I'm > guessing that's it if it's there. Is that it? > > I'm going to be logging the start of a phone call using the session > uuid > as the PK for a log record and need to be able to update that record > later with call duration information when the CDR comes through after > hangup. > > Brian > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jlists at skopis.com Fri Jan 30 19:15:36 2009 From: jlists at skopis.com (John Skopis (Lists)) Date: Fri, 30 Jan 2009 21:15:36 -0600 Subject: [Freeswitch-users] LDAP Integration In-Reply-To: <2DCF79D8-E2B5-43EB-93D1-EED92506E8DF@scarlet-internet.nl> References: <49417123.10709@ydeasolutions.com.br> <49417538.9040203@ydeasolutions.com.br> <200812120842.00808.hads@nice.net.nz> <49418790.60001@ydeasolutions.com.br> <87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com> <494198F3.10806@ydeasolutions.com.br> <4947A61F.6060806@ydeasolutions.com.br> <49485AD1.5070708@skopis.com> <4949025F.9040008@ydeasolutions.com.br> <49621532.5080003@ydeasolutions.com.br> <4962D64D.3080809@skopis.com> <2DCF79D8-E2B5-43EB-93D1-EED92506E8DF@scarlet-internet.nl> Message-ID: <4983C258.6080705@skopis.com> Leon de Rooij wrote: > Hi John, > > I've been trying to get your mod_xml_ldap module running, but didn't > get very far yet.. > > What is the official way to get the module built ? > The official way to build all fs modules is to uncomment the entry in modules.conf. If you want to build a specific module there are targets make mod_name-clean make mod_name-install as for mod_xml_ldap, I really do not feel that it is as quality as I would expect a production quality module to be. > I tried modifying trunk/freeswitch.spec so that > > XML_INT_MODULES contains xml_int/mod_xml_ldap > > There's also a directories/mod_ldap in DISABLED_MODULES in the same > file, but I don't suppose it's necessary to enable it, or is it ? > mod_ldap is a separate module, implementing the directory interface, not to be confused with the "directory", which is queried for user + domain configuration (e.g., conf/directory/default.xml). perhaps it should be renamed to mod_dbi? > The mod_xml_ldap doesn't get built by running make make or dpkg- > buildpackage from trunk/ > > Also I tried building it from the module directory itself, but then I > get the following error: > > fsbuilder at sv:~/trunk/src/mod/xml_int/mod_xml_ldap$ make > Compiling mod_xml_ldap.c... > cc1: warnings being treated as errors > mod_xml_ldap.c: In function 'xml_ldap_search': > mod_xml_ldap.c:356: warning: cast from pointer to integer of different > size > make[1]: *** [mod_xml_ldap.o] Error 1 > make: *** [all] Error 1 > I have been working on a new module called mod_entity that works off a simple description of an xml entitiy (domain, user, extension, condition, action, anti-action currently) querying a db backend via the directory interface for fields used to build the entity. It still needs a bit of work but I am hoping to get a patch together this weekend. I will post it to the freeswitch-dev list asking for comments. Off the top of my head at least the wishlist TODO is: implement connection pooling for mod_directory implement a cache either as a module used by an xml_int mod or in switch_xml to cache a switch_xml_t > (Also I had to apt-get install libsasl2 libsasl2-dev, otherwise make > from this dir errored with missing sasl/sasl.h) > > Can you see what I'm doing wrong ? > > (I'm using svn rev 11560) > > thanks & regards, > > Leon > > On Jan 6, 2009, at 4:55 AM, John Skopis (Lists) wrote: > >> Vinicius Kobashi wrote: >>> hi ppl. >>> >>> i tried hard to make it work, but still i couldnt find a complete >>> openldap scheme that provides these information, and i still could't >>> find out where to put these configuration... >>> >>> can anyone help me? >>> >>> thankz! >>> >>> vinicius escreveu: >>>> thankz! >>>> >>>> ill set my openldap to provide these information.. >>>> >>>> but these about these binding settings... where should i set them? >>>> >>>> best regards >>>> >>>> John Skopis (Lists) wrote: >>>>> vinicius wrote: >>>>> >>>>>> hi ppl.. i tried to find something at google, but i couldnt >>>>>> manage to find >>>>>> anything. >>>>>> i still dont know what to do to make the mod_xml_ldap work. >>>>>> i couldnt find information about how to build a config file for >>>>>> the >>>>>> module, and where to store it... >>>>>> >>>>>> can anyone give me a help? >>>>>> >>>>>> >>>>> Be advised mod_xml_ldap is probably not production quality and will >>>>> undoubtedly change, eventually at least. >>>>> >>>>> Here is what I used once: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> bindings="configuration"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> which should/probably/might work with ldap objects like these: >>>>> >>>>> dn: cn=John Skopis,ou=people,dc=example >>>>> objectClass: person >>>>> objectClass: inetOrgPerson >>>>> objectClass: organizationalPerson >>>>> objectClass: FreeSWITCH-Exten-Object >>>>> objectClass: top >>>>> cn: John Skopis >>>>> sn: Skopis >>>>> givenName: John >>>>> FSid: 1001 >>>>> FSmailbox: 1001 >>>>> FSpassword: 1234 >>>>> FSvm-password: 1001 >>>>> FSemail-addr: john+fs at skopis.com >>>>> FSvm-email-all-messages: TRUE >>>>> FSvm-delete-file: TRUE >>>>> FSvm-attach-file: TRUE >>>>> >>>>> dn: SIPIdentityUserName=1001,ou=h350,dc=example >>>>> objectClass: person >>>>> objectClass: SIPIdentity >>>>> objectClass: top >>>>> cn: 1001 >>>>> sn: 1001 >>>>> SIPIdentitySIPURI: sip:1001 at 172.16.75.129 >>>>> SIPIdentityRegistrarAddress: 172.16.75.128 >>>>> SIPIdentityProxyAddress: 172.16.75.128 >>>>> SIPIdentityPassword: 1234 >>>>> SIPIdentityUserName: 1001 >>>>> SIPIdentityServiceLevel: premium >>>>> >>>>> >> Again, the module is not production quality. Hopefully I will conjurer >> the time and know-how to put something decent together eventually. >> >> To load configuration for any fs module you need to define the XML >> configuration element under the section "configuration". >> >> A good starting point is the file >> $PREFIX/conf/freeswitch.xml >> >> http://wiki.freeswitch.org/wiki/Freeswitch.xml >> >> Also take a look at $PREFIX/logs/freeswitch.xml.fsxml >> >> to load mod_xml_ldap you would need to add something like this to >> modules.conf.xml >> >> >> >> and create an xml_ldap.conf.xml in >> $PREFIX/autoload_configs/xml_ldap.conf.xml >> >> >> ... >> >> >> The ITU is doing some work called h.350: >> http://www.itu.int/ITU-T/studygroups/com16/h350/index.html >> >> Here is what I was working with: >> attributetype ( 1.3.6.1.4.1.65535.2.1.1 NAME 'FSid' >> DESC 'FreeSWITCH Extension ID' >> EQUALITY caseIgnoreIA5Match >> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) >> >> attributetype ( 1.3.6.1.4.1.65535.2.1.2 NAME 'FSmailbox' >> DESC 'FreeSWITCH Extension Mailbox' >> EQUALITY caseIgnoreIA5Match >> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) >> >> attributetype ( 1.3.6.1.4.1.65535.2.1.3 NAME 'FSpassword' >> DESC 'FreeSWITCH Password' >> EQUALITY caseExactIA5Match >> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 >> SINGLE-VALUE ) >> >> attributetype ( 1.3.6.1.4.1.65535.2.1.4 NAME 'FSa1hash' >> DESC 'FreeSWITCH Crypted Password' >> EQUALITY caseExactIA5Match >> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 >> SINGLE-VALUE ) >> >> attributetype ( 1.3.6.1.4.1.65535.2.1.5 NAME 'FSvm-password' >> DESC 'FreeSWITCH VoiceMail Password' >> EQUALITY integerMatch >> SYNTAX 1.3.6.1.4.1.1466.115.121.1.27 >> SINGLE-VALUE ) >> >> attributetype ( 1.3.6.1.4.1.65535.2.1.6 NAME 'FSemail-addr' >> DESC 'E-mail address to send voicemail' >> EQUALITY caseIgnoreIA5Match >> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) >> >> attributetype ( 1.3.6.1.4.1.65535.2.1.7 NAME 'FSvm-email-all-messages' >> DESC 'FreeSWITCH Email All Mesages' >> EQUALITY booleanMatch >> SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 >> SINGLE-VALUE ) >> >> attributetype ( 1.3.6.1.4.1.65535.2.1.8 NAME 'FSvm-delete-file' >> DESC 'FreeSWITCH VoiceMail Delete File' >> EQUALITY booleanMatch >> SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 >> SINGLE-VALUE ) >> >> attributetype ( 1.3.6.1.4.1.65535.2.1.9 NAME 'FSvm-attach-file' >> DESC 'FreeSWITCH VoiceMail Attach file' >> EQUALITY booleanMatch >> SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 >> SINGLE-VALUE ) >> >> >> >> >> >> objectclass ( 1.3.6.1.4.1.65535.2.2.1 NAME 'FreeSWITCH-Exten-Object' >> SUP top AUXILIARY >> DESC '%obj_desc%' >> MUST ( FSid $ FSpassword ) >> MAY ( FSmailbox $ FSa1hash $ FSvm-password $ FSemail-addr $ >> FSvm-email-all-messages $ FSvm-delete-file $ FSvm-attach-file ) ) >> >> hth >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pmhshz at gmail.com Sat Jan 31 01:44:26 2009 From: pmhshz at gmail.com (shehzad p) Date: Sat, 31 Jan 2009 01:44:26 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> Message-ID: <21761523.post@talk.nabble.com> Hi Anthony, Freeswitch 1.0.2 was crashed on last test again... BT is on http://pastebin.freeswitch.org/6979. I tried to use scripts/freeswitch-gcore, to capture resident memory, but before the command complete the process system hanged up so only half output was captured.< http://www.nabble.com/file/p21761523/gcore-fs.txt gcore-fs.txt > Now I have checkout from trunk and will post back if any new thing found... Thanks, msp Anthony Minessale-2 wrote: > > if you are using unix you can use the supplied script > > scripts/freeswitch-gcore > > to capture a copy of the resident memory and I can have a look perhaps. > > Trunk is safe for production as we are in beta stage for a release of > 1.0.3 > at this time. > > > > On Fri, Jan 30, 2009 at 9:29 AM, shehzad p wrote: > >> >> When freeswitch freezes, we can't connect to it to check sps status, >> but once we were able to connect and at that time it was showing 0/0 sps. >> >> thanks... >> >> shehzad p wrote: >> > >> > Thanks, Anthony >> > >> > In my previous test sps did not changed, >> > but in recent test sps was dropped to 0 itself (as below). >> > =============================================================== >> > UP 0 years, 0 days, 5 hours, 1 minute, 53 seconds, 878 milliseconds, >> 190 >> > microseconds >> > 5474 session(s) since startup >> > 75 session(s) 0/0 >> > ============================================================= >> > >> > My system is 32 bit. >> > CPU is Intel(R) Xeon(R) CPU X3220 @ 2.40GHz >> > And RAM is 4GB >> > >> > Output of ulimit -a is: >> > ulimit -a: (set after first test) >> > core file size (blocks, -c) unlimited >> > data seg size (kbytes, -d) unlimited >> > max nice (-e) 20 >> > file size (blocks, -f) unlimited >> > pending signals (-i) unlimited >> > max locked memory (kbytes, -l) unlimited >> > max memory size (kbytes, -m) unlimited >> > open files (-n) 999999 >> > pipe size (512 bytes, -p) 8 >> > POSIX message queues (bytes, -q) unlimited >> > max rt priority (-r) unlimited >> > stack size (kbytes, -s) 244 >> > cpu time (seconds, -t) unlimited >> > max user processes (-u) unlimited >> > virtual memory (kbytes, -v) unlimited >> > file locks (-x) unlimited >> > =================================================== >> > >> > >> > BTW using trunk on production system is safe? >> > >> > Warm thanks for kind responses... >> > >> > >> > >> > Anthony Minessale-2 wrote: >> >> >> >> When you get it in that state what do you see when you execute >> >> >> >> fsctl sps >> >> >> >> is the sps a very low number? >> >> >> >> Did the sps drop by itself from the value you originally set it to? >> >> >> >> Are you using 32 bit? >> >> >> >> if so try all of these commands in your shell before starting FS >> >> >> >> ulimit -c unlimited >> >> ulimit -d unlimited >> >> ulimit -f unlimited >> >> ulimit -i unlimited >> >> ulimit -n 999999 >> >> ulimit -q unlimited >> >> ulimit -u unlimited >> >> ulimit -v unlimited >> >> ulimit -x unlimited >> >> ulimit -s 244 >> >> ulimit -l unlimited >> >> >> >> >> >> DO NOT put them in a script unless you source the script with . >> >> . myscript or they will be undone instantly when the script exits >> >> >> >> BTW, I said to try latest trunk not 1.0.2 We can only debug the >> >> development >> >> code at this point. >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Jan 29, 2009 at 10:06 AM, shehzad p wrote: >> >> >> >>> >> >>> Hi Anthony, >> >>> >> >>> I found interesting result while testing Freeswitch, and it might be >> >>> cause >> >>> of freezing out of freeswitch, >> >>> >> >>> I updated my system (as you told) with latest stable version >> Freeswitch >> >>> 1.0.2 >> >>> First of all I set sps to 100, >> >>> Then I sends call approximately 100 per seconds, Freeswitch works >> fine >> >>> and >> >>> handles all the calls very well. >> >>> >> >>> After that I send 130 calls per seconds, and magic happen now, >> >>> Freeswitch >> >>> handles first 100 calls only. >> >>> all the preceding calls were failed (even not appeared in freeswitch >> >>> console >> >>> why?) >> >>> >> >>> When I put ngrep trace, System responds with 503 Maximum Calls In >> >>> Progress. >> >>> (as below) >> >>> ########################################################### >> >>> # >> >>> U FSFSFSFSFS -> GWGWGWGWGW >> >>> SIP/2.0 503 Maximum Calls In Progress. >> >>> Via: SIP/2.0/UDP GWGWGWGWGW;branch=z9hG4bK53eafabb;rport=5060. >> >>> From: "99999" ;tag=as2e10c170. >> >>> To: ;tag=K3jSUFrDHpmmB. >> >>> Call-ID: 0feb229e58afe7be17110deb361dc234 at GWGWGWGWGW. >> >>> CSeq: 102 INVITE. >> >>> Retry-After: 300. >> >>> User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported. >> >>> Accept: application/sdp. >> >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> >>> NOTIFY, >> >>> REFER, UPDATE, REGISTER, INFO, PUBLISH. >> >>> Supported: timer, precondition, path, replaces. >> >>> Allow-Events: talk, presence, dialog, call-info, sla, >> >>> include-session-description, presence.winfo, message-summary, refer. >> >>> Content-Length: 0. >> >>> . >> >>> ##################################################################### >> >>> >> >>> >> >>> Now another issue to note down is that, >> >>> After all above happened and active calls comes to zero, >> >>> I just make a single call which also fails with response 503 - >> Maximum >> >>> Calls >> >>> In Progress. >> >>> >> >>> >> >>> Is this intended behaviour, should I increase SPS to overcome this. >> or >> >>> something like bug. >> >>> >> >>> Please let me know what should be the resolution for this. >> >>> >> >>> Thanks, >> >>> msp >> >>> >> >>> >> >>> >> >>> Anthony Minessale-2 wrote: >> >>> > >> >>> > Also remember, >> >>> > Actually completely uninstall and erase /usr/local/freeswitch and >> the >> >>> > 1.0.1 >> >>> > source tree and freshly install the new one. >> >>> > If you try to upgrade on top of a release with trunk it will cause >> >>> more >> >>> > problems for you. >> >>> > >> >>> > >> >>> > On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice >> >>> wrote: >> >>> > >> >>> >> Upgrade to trunk... Many many issues have been resolved since >> 1.0.1 >> >>> was >> >>> >> the >> >>> >> current release >> >>> >> >> >>> >> >> >>> >> > From: shehzad p >> >>> >> > Reply-To: >> >>> >> > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) >> >>> >> > To: >> >>> >> > Subject: [Freeswitch-users] Freeswitch freezes on increasing >> call >> >>> >> traffic >> >>> >> > >> >>> >> > >> >>> >> > Hi all, >> >>> >> > >> >>> >> > Yesterday my Freeswitch server faced a problem when call traffic >> >>> >> increased >> >>> >> > to more than 100. >> >>> >> > >> >>> >> > When I start Freeswitch, it works fine and then after some time >> >>> >> > (approximately 15 to 20 minutes) it stops functioning (means no >> >>> call >> >>> >> is >> >>> >> > being processed, no CLI command is working and it just freezes) >> >>> until >> >>> I >> >>> >> > restart the freeswitch. >> >>> >> > >> >>> >> > I am using Freeswitch 1.0.1. >> >>> >> > Debug (gdb) trace as on wiki page >> >>> >> > >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.shis >> >>> >> attached >> >>> >> > http://www.nabble.com/file/p21701744/fs_debgu.txt fs_debgu.txt >> >>> >> > -- >> >>> >> > View this message in context: >> >>> >> > >> >>> >> >> >>> >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 >> >>> >> > p21701744.html >> >>> >> > Sent from the Freeswitch-users mailing list archive at >> Nabble.com. >> >>> >> > >> >>> >> > >> >>> >> > _______________________________________________ >> >>> >> > Freeswitch-users mailing list >> >>> >> > Freeswitch-users at 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 at 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 at hotmail.com >> >>> >> >> >< >> >>> >> MSN%3Aanthony_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 >> >>> >> >> >< >> >>> >> sip%3A888 at conference.freeswitch.org >> >> > >> >>> > >> >>> > iax:guest at conference.freeswitch.org/888 >> >>> > >> >>> >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> >>> >> >> >> > >> >>> > >> >>> > pstn:213-799-1400 >> >>> > >> >>> > _______________________________________________ >> >>> > Freeswitch-users mailing list >> >>> > Freeswitch-users at lists.freeswitch.org >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> > >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> > http://www.freeswitch.org >> >>> > >> >>> > >> >>> >> >>> -- >> >>> View this message in context: >> >>> >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21729863.html >> >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >>> >> >>> >> >>> _______________________________________________ >> >>> Freeswitch-users mailing list >> >>> Freeswitch-users at 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 at hotmail.com >> < >> MSN%3Aanthony_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 >> < >> sip%3A888 at conference.freeswitch.org >> > >> >> iax:guest at conference.freeswitch.org/888 >> >> >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> >> pstn:213-799-1400 >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21749375.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21761523.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From saigop at gmail.com Sat Jan 31 04:12:25 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Sat, 31 Jan 2009 17:42:25 +0530 Subject: [Freeswitch-users] Javascript Dial and Print the UUID in event socket Message-ID: <2ea4d47e0901310412h5b2563can4344b7db4d21383a@mail.gmail.com> Hi, I am using event socket. I am trying to dial a outbound number in Javascript (api jsrun