[Freeswitch-users] Is mod vmd working?
I followed these instructions for Mod_vmd except for a Windows box: http://wiki.freeswitch.org/wiki/Mod_vmd I tried testing to see if it's working by dialing the following extension: ??? !-- mod_vmd test extension (new mod)-- ??? extension name=vmdtest ??? ??? condition field=destination_number expression=^$ ??? ??? ??? action application=answer/ ??? ??? ??? action application=info/ ??? ??? ??? action application=vmd/ ??? ??? ??? action application=sleep data=25000/ ??? ??? ??? action application=info/ !-- Look for chan var vmd_detect here -- ??? ??? ??? action application=hangup/ ??? ??? /condition ??? /extension However, I didn't see channel variable vmd_detect in the FreeSwitch console. ?? Mark. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is mod vmd working?
Mark, It does work ... but I can't really attest to how well ... especially compared to other things out there. I started capturing this in CDR's to see and it didn't seem like it worked very well. If this is really critical to you, you might want to ping Ken Rice. I know he might have a better option. SDR ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Feedback on Freeswitch for Windows?
Hello For single-host settings, getting customers to buy a separate server just to run Freeswitch is overkill, so I'm thinking about selling just the IVR application to run on Windows. Unless a PCI card is available, the FXO connection will be provided by Sangoma's USB device. I'd like some feedback on running Freeswitch on XP and Vista: Is it ready for production use? Does it require beefy hardware? Thank you for any hint. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Mod_limit stuck when hitting limit value
thanks Mathieu. I setup an IRC account to give it a try. Comme ça je pourrais t'embeter avec mes pbms :p rod Mathieu Rene wrote: limit_hash uses a faster data structure then limit but works the same way for tne end-user. viens sur IRC si t'as des questions en francais =) Math On 17-Mar-09, at 3:06 AM, rod wrote: Hi, not too hard :p but it's just a bad habit when I write in my native language (french). I guess that this spelling is not too common for english speaker. I'll do my best next time to write it correctly. @tamas you are right, we could use limit_hash the same way as limit when not specifying the /rate @Mathieu did you suggest limit_hash is more scalable than limit? But I don't understand why limit_hash is not suitable for data replication (DB lookup for limit and memory for limit_hash??), even if I don't know how to do it with limit. regards. Raymond Chandler wrote: Tamas wrote: My guess is: pbm = problem :) sure, but is it really that hard to spell all the way out? -Ray ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] echo cancellation on PRI cards
To share my experience: I had issues with echo with many E1 trunks in Serbia, especially when voice in between telco's network went to well known bad analog lines. I used OSLEC and I was fortunate to have Steve to complain to, he helped patching it further after my beta testing. Not many people would do that imho. I now switched to Sangoma cards with Octasic chips and occasionally would still hear certain echo. My view is that here some echo cancelling solution is very necessary, otherwise whole VoIP business comes up to bad reputation People would just not listen to themselves speaking, even using $400 phone. Regards, Ognjen 2009/3/18 David Knell d...@3c.co.uk Steve Underwood wrote: [whopping big snip] The first bit of that's a tad patronising, isn't it, You are the one who started out being offensive. I'm sorry if you find disagreement offensive; you might not wish to read beyond this point if so. and, in the case of the decade-old Aculab cards which which I'm most familiar, is also untrue. I can't find too much about the old cards on the web now, but I found http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html which is pretty much a copy and paste from the old Dialogic web pages, and you'll see it says Cut through : Local echo cancellation permits 100% detection with a 4.5 dB return loss line. The Aculabs did the same thing for sure. They just couldn't work without cancellation. There were some very early Dialogic cards, using DTMF receiver chips and OKI ADPCM chips, and had no general purpose DSPs. They performed really badly because of the lack of cancellation, and were quickly replaced with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms into a Motorola 56k DSP chips. The same document, under the bit which you've quoted, says: (E-1) Digital trunks use separate transmit and receive paths to network. Performance dependent on far end handset's match to local analog loop. - i.e. the card does no echo cancellation. Aculab didn't even offer echo cancellation on Prosody for years and, when they did, it consumed prodigious amounts of DSP. Nonetheless, the DTMF detection worked perfectly well, even across 120 channels per 40MHz SHARC - there's just no way that those DSPs had enough horsepower to do echo cancellation across that many channels. An Asterisk box with an el-cheapo quad E1 card in that I use for TDM-SIP gatewaying detects DTMF perfectly well with no echo cancellation. You just don't need echo cancellation to achieve perfectly acceptable DTMF detection. ASR - yes, maybe, but surely only in the case where the application requires barge-in; even then, I'd be interested to see some test results, particuarly where the outbound prompt is killed the moment the ASR reports start of speech. I'm afraid that your original bald claim - that IVRs badly need echo cancellation is simply wrong, misleading and irresponsible: those believing it will end up spending large sums of money on technology which they probably do not need. --Dave ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] openZAP disconnect cause wrong?
Hello, ok found it ... was a configuration issue due to the continue on fail = true variable in my dialplan. Hangup application fixed this :) Sorry for the post. regards helmut On 18.03.2009 10:20, Helmut Kuper wrote: Hello, I'm not sure whether the following is a bug or a config issue: ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] openZAP disconnect cause wrong?
Hello, I'm not sure whether the following is a bug or a config issue: I found this in my log file: 2009-03-18 10:07:00 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: USER_BUSY 2009-03-18 10:07:00 [DEBUG] mod_dptools.c:2025 audio_bridge_function() Continue on fail [true]: Cause: USER_BUSY 2009-03-18 10:07:00 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:5/2850 [CS_EXECUTE] [NORMAL_CLEARING] FS obviously doesn't pass through the disconnect cause from Bridge app to openzap module. Analyzing the corresponding q931.pcap trace confirms this. Do I have to configure it somewhere e.g. a mapping or so, or is this a bug? regrads helmut ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Losing Gateway registration
Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is mod vmd working?
SDR?
I'm wondering why there was nothing in the console showing the channel
variable ${vmd_detect} as the wiki says there should be:
action application=info/ !-- Look for chan var vmd_detect here --
Mark
-Original Message-
From: Shelby Ramsey sicfsl...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 18 Mar 2009 12:11 am
Subject: Re: [Freeswitch-users] Is mod vmd working?
Mark,
It does work ... but I can't really attest to how well ... especially compared
to other things out there.? I started capturing this in CDR's to see and it
didn't seem like it worked very well.
If this is really critical to you, you might want to ping Ken Rice.? I know he
might have a better option.
SDR
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Re: [Freeswitch-users] Is mod vmd working?
Mark, Because it didn't detect a beep. It will be be there as vmd_detect=true if it does. I'm not sure exactly how reliable it's beep detection is. SDR ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] echo cancellation on PRI cards
OK, one last go and I give up. Lets look at the documentation for Dialogic springware. This is the DSP package that loads in their cards or runs on the host in HMP applications. It does things like DTMF generation and detection for all Dialogic cards except the DM3 series. The documentation says: *PerfectDigit DTMF Signaling* • DSP-based DTMF (touchtone) detection algorithm optimized for lowest talk-off and play-off susceptibility in the industry. The system will not easily be fooled by mistaking human speech for DTMF tones. • Minimum tone duration and interdigit delay times accurately handle speed dialing typical of power users • Utilizes echo cancellation which results in superior cut through for accurate DTMF tone interpretation during voice file playback within a broad range of network/switch environments • DTMF outbound dialing generated by DSP for accuracy and flexibility (dialing levels are adjustable to meet a variety of global PTT requirements) Detecting supervisory tones on phone lines is trivial. Not falsely detecting them is where things get interesting. The standard test for DTMF receivers is a set of cassette tapes from Bellcore containing about 3 hours of snippets from real telephone calls in North America. Most DTMF receiver chips get a few hundred false DTMF hits in those 3 hours. Dialogic get 20 something. My DTMF receiver gets 19. The reason its hard to detect these things reliably is voice doesn't sit there nicely at one level. Its level and its spectrum bounce all over the place, and a real DTMF digit is only there for 40ms or so. I defy anyone to visually identify a 40ms DTMF digit amongst real dynamic speech if it isn't *way* above the voice in amplitude. This is why your phone has to mute your voice when you press a digit. The DTMF receiver has no chance of reliable detection with speech and digits mixed. In the few special cases where concurrent speech and signaling tone are present on the PSTN (e.g. 2280Hz signaling in .eu and 2600Hz in .us) the signaling sequence is very carefully constructed to avoid confusing the system. DTMF is never used in that way. There is one obvious special case where all DTMF receivers need to tolerate spillback. They need to differentiate between dialing tone and DTMF on the first digit you dial. They do this very simply. Dialing tone was chosen to be pretty low frequencies - 350Hz + 440Hz, 425Hz + 475Hz and similar pairings. The lowest DTMF tone is well above this. An aggressive low pass filter in the DTMF receiver removes the dial tone spillback, while barely affecting the lowest DTMF tone. This was the original design of DTMF, but.. IVRs changed all that. Their DTMF receivers are expected to work amidst outgoing prompts, which may be going to phones with an awful match to the line. The spillback can be huge. The good IVR hardware suppliers, like Dialogic, very quickly added echo cancellation to their cards. I can say a lot of negative things about Dialogic, but one thing they did really well was their DTMF cut-through. When people get used to an IVR they expect to hammer in digit sequences as fast as they can, in the face of a machine desperately trying to play voice prompts to them. Dialogic cards do this really well, on lines of all types, and on networks of varying quality. This would be impossible without echo cancellation. David Knell wrote: Steve Underwood wrote: David Knell wrote: Steve Underwood wrote: [whopping big snip] The first bit of that's a tad patronising, isn't it, You are the one who started out being offensive. I'm sorry if you find disagreement offensive; you might not wish to read beyond this point if so. and, in the case of the decade-old Aculab cards which which I'm most familiar, is also untrue. I can't find too much about the old cards on the web now, but I found http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html which is pretty much a copy and paste from the old Dialogic web pages, and you'll see it says Cut through : Local echo cancellation permits 100% detection with a 4.5 dB return loss line. The Aculabs did the same thing for sure. They just couldn't work without cancellation. There were some very early Dialogic cards, using DTMF receiver chips and OKI ADPCM chips, and had no general purpose DSPs. They performed really badly because of the lack of cancellation, and were quickly replaced with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms into a Motorola 56k DSP chips. The same document, under the bit which you've quoted, says: (E-1) Digital trunks use separate transmit and receive paths to network. Performance dependent on far end handset's match to local analog loop. - i.e. the card does no echo cancellation. Your messages are starting to looked deranged. Why would they only
Re: [Freeswitch-users] Is mod vmd working?
Another issue with this module is the resources it consumes. We had it running on 50 calls yesterday and the cpu's all went to 90+% Regards, From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Shelby Ramsey Sent: 18 March 2009 13:08 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Is mod vmd working? Mark, Because it didn't detect a beep. It will be be there as vmd_detect=true if it does. I'm not sure exactly how reliable it's beep detection is. SDR ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is mod vmd working?
Nik Middleton wrote: Another issue with this module is the resources it consumes. We had it running on 50 calls yesterday and the cpu’s all went to 90+% That's odd. Something must be fouling up, as the algorithm he used should be very lightweight. Steve ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Losing Gateway registration
Upgrade to 1.03 or SVN Trunk /b On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Losing Gateway registration
if you are behind NAT it is possible that your router forgot the mapping betweeen FS and your provider, try addingparam name=ping value=30 / to your gateway. Math On 18-Mar-09, at 10:07 AM, Brian West wrote: Upgrade to 1.03 or SVN Trunk /b On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] TLS support in Debian build
On Mar 17, 2009, at 10:31 PM, Jason White wrote: Brian West br...@freeswitch.org wrote: if you installed the ssl devel stuff AFTER you configured you'll need to reconfigure. I'm reasonably sure it was installed already, unless it was pulled in recently by a package upgrade. The configure script needs to look in /usr/include/openssl for the headers. I'll have a look at config.log and try to work out what it looked for and why it didn't find it. you will have to look in the config.log in libs/sofia-sip Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Feedback on Freeswitch for Windows?
There is currently no openzap (sangoma, etc) support on windows, we hope this will be coming soon. Mike On Mar 17, 2009, at 5:20 AM, Gilles wrote: Hello For single-host settings, getting customers to buy a separate server just to run Freeswitch is overkill, so I'm thinking about selling just the IVR application to run on Windows. Unless a PCI card is available, the FXO connection will be provided by Sangoma's USB device. I'd like some feedback on running Freeswitch on XP and Vista: Is it ready for production use? Does it require beefy hardware? Thank you for any hint. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is mod vmd working?
I added a voicemail tag in to a default extension 1001, I hear the voicemail beep but still don't see vmd_detect. Mark -Original Message- From: Shelby Ramsey sicfsl...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, 18 Mar 2009 6:07 am Subject: Re: [Freeswitch-users] Is mod vmd working? Mark, Because it didn't detect a beep.? It will be be there as vmd_detect=true if it does.? I'm not sure exactly how reliable it's beep detection is.? SDR ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem dialing out via E1
We're a couple more steps forward from yesterday. Turned out some of my regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra space before one of the closing brackets in the default.xml example. After staring at the screen all day it's funny how you miss these things! Situation now is I can get the call into FS but, it rings the extension for a fraction of a second then the call drops. Here's the contents of the public and default dialplans I'm using (as per example in the wiki) and the debug - http://pastebin.freeswitch.org/7819 http://pastebin.freeswitch.org/7819 I'm also seeing another issue when placing subsequent inbound calls, they bounce if hitting the same channel the first call came in to (typically /1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818 http://pastebin.freeswitch.org/7818 Getting there (slowly) Mark. mercutioviz wrote: On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron mark.tab...@rnid-typetalk.org.uk wrote: Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, We didn't do anything. I think that's a euphemism for we did a reset and prayed. However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Problem-dialing-out-via-E1-tp22479047p22582281.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem dialing out via E1
We're a couple more steps forward from yesterday. Turned out some of my regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra space before one of the closing brackets in the default.xml example. After staring at the screen all day it's funny how you miss these things! Situation now is I can get the call into FS but, it rings the extension for a fraction of a second then the call drops. Here's the contents of the public and default dialplans I'm using (as per example in the wiki) and the debug - http://pastebin.freeswitch.org/7819 I'm also seeing another issue when placing subsequent inbound calls, they bounce if hitting the same channel the first call came in to (typically /1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818 Getting there (slowly) Mark. quote author=mercutioviz On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron mark.tab...@rnid-typetalk.org.uk wrote: Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, We didn't do anything. I think that's a euphemism for we did a reset and prayed. However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org /quote -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 March 2009 15:48 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron mark.tab...@rnid-typetalk.org.uk wrote: Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, We didn't do anything. I think that's a euphemism for we did a reset and prayed. However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup
I upgraded to FreeSWITCH Version 1.0.trunk (12654M) but caller is still being hungup (and not continuing on with dialplan) after agent disconnect with hangup_after_bridge=false Is there a separate patch I need to apply? Thanks. --matt On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong mattdf...@gmail.com wrote: Hi Anthony, thanks for the reply. I've searched thru jira, and didn't find anything when searching for fifo that was recently updated or related, except http://jira.freeswitch.org/browse/MODAPP-189 and I'm not sure if this does what I need. Was this what you were referring to? Thanks. --matt 2009/3/17 Anthony Minessale anthony.miness...@gmail.com there is a patch in jira that will implement this feature about to be added 2009/3/17 Matthew Fong mattdf...@gmail.com I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the fifo in execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem dialing out via E1
2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch there was a timer problem which was not solved yet. This caused channels to be busy in my case. I am not sure whether this is solved yet. Can anybody confirm? Best regards Peter Mark Tabron schrieb: We're a couple more steps forward from yesterday. Turned out some of my regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra space before one of the closing brackets in the default.xml example. After staring at the screen all day it's funny how you miss these things! Situation now is I can get the call into FS but, it rings the extension for a fraction of a second then the call drops. Here's the contents of the public and default dialplans I'm using (as per example in the wiki) and the debug - http://pastebin.freeswitch.org/7819 I'm also seeing another issue when placing subsequent inbound calls, they bounce if hitting the same channel the first call came in to (typically /1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818 Getting there (slowly) Mark. quote author=mercutioviz On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron mark.tab...@rnid-typetalk.org.uk wrote: Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, We didn't do anything. I think that's a euphemism for we did a reset and prayed. However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org /quote -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 March 2009 15:48 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron mark.tab...@rnid-typetalk.org.uk wrote: Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, We didn't do anything. I think that's a euphemism for we did a reset and prayed. However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org
Re: [Freeswitch-users] Is mod vmd working?
2009/3/18 mszla...@aol.com: I added a voicemail tag in to a default extension 1001, I hear the voicemail beep but still don't see vmd_detect. Mark FYI, I've used mod_vmd but only in a TDM environment on outbound calls via a PRI. It worked very well on for detecting answering machine beeps and vm beeps on cell phone voice mails. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem dialing out via E1
On Wed, Mar 18, 2009 at 10:18 AM, Peter P GMX prometheus...@gmx.net wrote: 2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch there was a timer problem which was not solved yet. This caused channels to be busy in my case. I am not sure whether this is solved yet. Can anybody confirm? We're using ozmod_libpri which has it's own PRI handling. So far, so good... -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup
This is the patch http://jira.freeswitch.org/browse/MODAPP-237 it's not added yet. 2009/3/18 Matthew Fong mattdf...@gmail.com I upgraded to FreeSWITCH Version 1.0.trunk (12654M) but caller is still being hungup (and not continuing on with dialplan) after agent disconnect with hangup_after_bridge=false Is there a separate patch I need to apply? Thanks. --matt On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong mattdf...@gmail.comwrote: Hi Anthony, thanks for the reply. I've searched thru jira, and didn't find anything when searching for fifo that was recently updated or related, except http://jira.freeswitch.org/browse/MODAPP-189 and I'm not sure if this does what I need. Was this what you were referring to? Thanks. --matt 2009/3/17 Anthony Minessale anthony.miness...@gmail.com there is a patch in jira that will implement this feature about to be added 2009/3/17 Matthew Fong mattdf...@gmail.com I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the fifo in execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is mod vmd working?
Hi MC,
With trunk 12638M, I tried checking vmd internally and externally to my cell.
No luck at all in detecting a voicemail (beep).
I used the following extensions to test this, maybe they are in error.
If not then how else can I detect from FS that I got voicemail in a phone
agnostic way (i.e, pots sip).
extension name=110
condition field=destination_number expression=^110$
action application=answer/
action application=vmd/
!--action application=voicemail data=default 10.0.0.3 1000/--
action application=bridge
data=sofia/gateway/spa3102PSTN/12223334...@10.0.0.5:5061/
action application=transfer data=111 XML default/
/condition
/extension
extension name=111
condition field=destination_number expression=^111$/
condition field=${vmd_detect} expression=^TRUE
action application=answer/
action application=speak data=flite|kal|voicemail detected/
action application=hangup/
anti-action application=answer/
anti-action application=speak data=flite|kal|no voicemail detected/
anti-action application=hangup/??? ??? ???
/condition
/extension
Mark.
-Original Message-
From: Michael Collins m...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 18 Mar 2009 10:24 am
Subject: Re: [Freeswitch-users] Is mod vmd working?
2009/3/18 mszla...@aol.com:
I added a voicemail tag in to a default extension 1001, I hear the
voicemail beep but still don't see vmd_detect.
Mark
FYI, I've used mod_vmd but only in a TDM environment on outbound calls
via a PRI. It worked very well on for detecting answering machine
beeps and vm beeps on cell phone voice mails.
-MC
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Re: [Freeswitch-users] TLS support in Debian build
Was this ever resolved? If we're missing something in the documentation, I'd like to make sure it's in there. Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3239 x0 On Mar 18, 2009, at 11:34 AM, Michael Jerris wrote: On Mar 17, 2009, at 10:31 PM, Jason White wrote: Brian West br...@freeswitch.org wrote: if you installed the ssl devel stuff AFTER you configured you'll need to reconfigure. I'm reasonably sure it was installed already, unless it was pulled in recently by a package upgrade. The configure script needs to look in /usr/include/openssl for the headers. I'll have a look at config.log and try to work out what it looked for and why it didn't find it. you will have to look in the config.log in libs/sofia-sip Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org PGP.sig Description: This is a digitally signed message part ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] TLS support in Debian build
I thought we had... hrm. /b On Mar 18, 2009, at 5:39 PM, Karl Vesterling wrote: Was this ever resolved? If we're missing something in the documentation, I'd like to make sure it's in there. Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3239 x0 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is mod vmd working?
Hmm, Well We're connected direct to E1's and it doesn't work reliably here. That said, DTMF detect does recognise the beeps most of the time. Perhaps there's a regional variation. I wonder if it's country specific. The code looks logical. When I get some time I'll have a look at it and see how it can be improved. The concept is great and is much better that sniffing out human voice as that's prone to false positives. Much better to assume human and machine. Nothing worse than a silent call. Regards, -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 March 2009 17:24 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Is mod vmd working? 2009/3/18 mszla...@aol.com: I added a voicemail tag in to a default extension 1001, I hear the voicemail beep but still don't see vmd_detect. Mark FYI, I've used mod_vmd but only in a TDM environment on outbound calls via a PRI. It worked very well on for detecting answering machine beeps and vm beeps on cell phone voice mails. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is mod vmd working?
Ironically, I've used tone_detect to try and trap SIT tones and I found that answering machines in the USA seem to all send a beep in the same freq range as American SIT tones... :) -MC On Wed, Mar 18, 2009 at 4:22 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hmm, Well We're connected direct to E1's and it doesn't work reliably here. That said, DTMF detect does recognise the beeps most of the time. Perhaps there's a regional variation. I wonder if it's country specific. The code looks logical. When I get some time I'll have a look at it and see how it can be improved. The concept is great and is much better that sniffing out human voice as that's prone to false positives. Much better to assume human and machine. Nothing worse than a silent call. Regards, -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 March 2009 17:24 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Is mod vmd working? 2009/3/18 mszla...@aol.com: I added a voicemail tag in to a default extension 1001, I hear the voicemail beep but still don't see vmd_detect. Mark FYI, I've used mod_vmd but only in a TDM environment on outbound calls via a PRI. It worked very well on for detecting answering machine beeps and vm beeps on cell phone voice mails. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Is there a way to automatically re-login gtalk account
Hi all, mod_dingaling in client mode works well for me, but disconnected yesterday. 2009-03-18 16:57:32 [DEBUG] libdingaling.c:1545 xmpp_connect() io error 2 7 I use dl_login profile=gmail.com, and it re-login successfully. Is their a way to auto re-login after fail? Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is mod vmd working?
tone_detect! sounds good. BTW, was there any errors in those extensions I posted. I modified something you posted MC. -Original Message- From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Wed, 18 Mar 2009 5:15 pm Subject: Re: [Freeswitch-users] Is mod vmd working? Ironically, I've used tone_detect to try and trap SIT tones and I found that answering machines in the USA seem to all send a beep in the same freq range as American SIT tones... :) -MC On Wed, Mar 18, 2009 at 4:22 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hmm, Well We're connected direct to E1's and it doesn't work reliably here. That said, DTMF detect does recognise the beeps most of the time. Perhaps there's a regional variation. ?I wonder if it's country specific. ?The code looks logical. ?When I get some time I'll have a look at it and see how it can be improved. The concept is great and is much better that sniffing out human voice as that's prone to false positives. ?Much better to assume human and machine. ?Nothing worse than a silent call. Regards, -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 March 2009 17:24 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Is mod vmd working? 2009/3/18 ?mszla...@aol.com: I added a voicemail tag in to a default extension 1001, I hear the voicemail beep but still don't see vmd_detect. Mark FYI, I've used mod_vmd but only in a TDM environment on outbound calls via a PRI. It worked very well on for detecting answering ?machine beeps and vm beeps on cell phone voice mails. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Issue relating mod_nibblebill
Is this issue still open? I just noticed this. The error you are receiving indicates UnixODBC is installed, but not configured properly (most likely anyway). The UnixODBC drivers are kind of a pain to setup on some systems, especially CentOS, but this article may help you get it working - http://webaj.com/how-setup-mysql-dsn-datasbase-source-centos-myodbc-and-unix odbc-command-line.htm. I strongly recommend making sure the test commands they list work before trying to get UnixODBC working within FreeSWITCH. Also, it looks like you may have failed to copy the sample mod_nibblebill XML config file to /usr/local/freeswitch/conf/autoload_configs/ . You may want to give that a try. Within that file is the name of the ODBC driver being referenced - make sure that driver exists (see link above). - Darren -Original Message- From: JayaPrakash [mailto:jp.man...@gmail.com] Sent: Saturday, March 14, 2009 4:49 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Issue relating mod_nibblebill Hi All, I am newbie to Freeswitch. I installed freeswitch-1.0.3 in Debian machine. I am able to make call, check presence, retrieve CDRs. I followed the installation steps given in mod_nibblebill for rating. While, installing mysql-connector-odbc, it has thrown errors related to mysql-config file, that it does not exist. Coming to mysql, mysql-client-5 and mysql-server-5 are installed. So I installed libmyodbc which is used for the same functionality. Rest of the steps are done, as given in mod-nibblebill installation. When the freeradius server is restarted, it has given the following error. 2009-03-14 14:55:47 [ERR] switch_odbc.c:164 switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2009-03-14 14:55:47 [CRIT] mod_nibblebill.c:233 load_config() Cannot connect to ODBC driver/database freeswitch (user: root / pass dev)! Will you please have a look in solving this issue ? , how the issue can be solved? Thanks Regards, JP. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Is there a way to automatically re-login gtalk account
I do have auto-login enabled in jingle_profile: param name=auto-login value=true/ param name=auto-reply value=Press *Call* to join my conference/ param name=sasl value=plain/ param name=tls value=true/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] FS in Solaris
Hi list, Any experience building FS in Solaris using Sun Studio? Thanks Pablo ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
