Re: [asterisk-users] help with Cepstral 6 and Asterisk 11
Wouldn't it be easier in your case to pay somebody to do the job? I doubt that it would take more than a couple of minutes to compile, install, and configure the package. Maybe some things need to get adjusted as the author has abandoned the project (at least there is no longer a project web page) and the latest sources are about 2 years old. Building from sources is not that difficult, but if you don't have a proper configure script you are responsible that all prerequisites are met, which can be time consuming if you don't know your distro well enough. Here, there is no configure script and some things, which might be invalid for your machine, are hand coded inside the Makefile. Nothing spectacular, but you could end up asking a lot more questions that have nothing to do with asterisk. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CTI
Hi people I'm just mailing to see what people are using for CTI solutions with asterisk. Aslos, has anyone managed to integrate asterisk with Salesforce? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CTI
http://camrivox.com/products/flexor-cti-salesforce/ We've used this for a few clients. On Fri, Jan 10, 2014 at 6:33 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi people I'm just mailing to see what people are using for CTI solutions with asterisk. Aslos, has anyone managed to integrate asterisk with Salesforce? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- www.ringfree.biz 828-575-0030 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Structure of asterisk follow me table
Hi everybody, I'am developping an interface to manage asterisk follow me module.I want to do this in realtime and I didn't find the structure of the asterisk follow me module. I'am working with asterisk 1.8.5 and Postgresql 9.2 on CentOS 6.2 Can any body help me ! Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CTI
- Original Message - http://camrivox.com/products/flexor-cti-salesforce/ We've used this for a few clients. How were your experiences with it? I have a customer that will want this type of integration in the near future, and would love to hear how installation, operation, and support has been. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 11.7.0: Delayed audio
On connection to an incoming call via PSTN where asterisk [11.7.0] is Dialing an internal extension on answering the call there is about 6-7 seconds before audio is heard on either side. When looking at the CLI traces when I answer the incoming call that asterisk extensions were dialing, I see immediately upon answering 0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:portnumber then not until about 6 seconds later I see this 0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:diffportnumber and immediately hear audio what appears to be an issue is that the RTP link(audio) setup is delayed. Anyone have suggestions on how to fix this issue? pc -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk API
Hello Folks; I have an Asterisk server Asterisk 11.7.0 built by root @xxx on a x86_64 running Linux on 2013-12-27 18:47:44 UTC No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. Is there an API out there that anyone knows of that I can pass commands, etc to Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, etc? I'm kind of clueless. Is there something available? Thanks - Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CTI
No major issues. They're always very responsive. I'd get a demo from them for the client and make sure that the feature set is a match. But I always say that with 3rd party apps. On Fri, Jan 10, 2014 at 10:39 AM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - http://camrivox.com/products/flexor-cti-salesforce/ We've used this for a few clients. How were your experiences with it? I have a customer that will want this type of integration in the near future, and would love to hear how installation, operation, and support has been. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- www.ringfree.biz 828-575-0030 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to read IRQs and timing slips values
2014/1/9 Shaun Ruffell sruff...@digium.com On Thu, Jan 09, 2014 at 06:01:34PM +0100, Olivier wrote: Hi, On a Asterisk 1.8.12 system working OK for months (100k calls proceed), users are complaining for bad audio. My setup is: PSTN --E1/PRI --- Asterisk --- E1/PRI--- Siemens HiPath ---E1/PRI --- PSTN asterisk -rx dahdi show version DAHDI Version: SVN-trunk-r10414 Echo Canceller: HWEC asterisk -rx pri show version libpri version: 1.4.12 A quick glance at Asterisk logs shows lines like this: [2014-01-09 17:19:34] NOTICE[26034]: chan_dahdi.c:3099 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 1 [2014-01-09 17:19:35] NOTICE[26035]: chan_dahdi.c:3099 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 2 [2014-01-09 17:19:49] NOTICE[26035]: chan_dahdi.c:3099 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 2 I read an old thread inviting an admin to check for shared IRQs and timing slips. My questions are: 1. cat /proc/interrupts 's output is: # cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 0: 90147 0 0 0 00 0 0 IO-APIC-edge timer 1: 2 0 0 0 00 0 0 IO-APIC-edge i8042 8: 0 0 1 0 00 0 0 IO-APIC-edge rtc0 9: 0 0 0 0 00 0 0 IO-APIC-fasteoi acpi 12: 4 0 0 0 00 0 0 IO-APIC-edge i8042 14: 93 0 0 0 00 0 0 IO-APIC-edge ata_piix 15: 0 0 0 0 00 0 0 IO-APIC-edge ata_piix 16: 3378646209 3378695076 3378691115 3378697362 3378691116 3378706831 3378710635 3378702358 IO-APIC-fasteoi wct2xxp Can I positively conclude that my Dahdi PRI board IS NOT sharing IRQ (which is good) ? Yes, there is not IRQ sharing going on. 2. What would you suggest reading the following output ? cat /proc/dahdi/2 Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 (MASTER) HDB3/CCS Timing slips: 175319 32 TE2/0/2/1 Clear (In use) (EC: VPMOCT064 - INACTIVE) 33 TE2/0/2/2 Clear (In use) (EC: VPMOCT064 - INACTIVE) 34 TE2/0/2/3 Clear (In use) (EC: VPMOCT064 - INACTIVE) 35 TE2/0/2/4 Clear (In use) (EC: VPMOCT064 - INACTIVE) 36 TE2/0/2/5 Clear (In use) (EC: VPMOCT064 - INACTIVE) 37 TE2/0/2/6 Clear (In use) (EC: VPMOCT064 - INACTIVE) 38 TE2/0/2/7 Clear (In use) (EC: VPMOCT064 - INACTIVE) 39 TE2/0/2/8 Clear (In use) (EC: VPMOCT064 - INACTIVE) 40 TE2/0/2/9 Clear (In use) (EC: VPMOCT064 - INACTIVE) 41 TE2/0/2/10 Clear (In use) (EC: VPMOCT064 - INACTIVE) 42 TE2/0/2/11 Clear (In use) (EC: VPMOCT064 - INACTIVE) 43 TE2/0/2/12 Clear (In use) (EC: VPMOCT064 - INACTIVE) 44 TE2/0/2/13 Clear (In use) (EC: VPMOCT064 - INACTIVE) 45 TE2/0/2/14 Clear (In use) (EC: VPMOCT064 - INACTIVE) 46 TE2/0/2/15 Clear (In use) (EC: VPMOCT064 - INACTIVE) 47 TE2/0/2/16 HDLCFCS (In use) (EC: VPMOCT064 - INACTIVE) 48 TE2/0/2/17 Clear (In use) (EC: VPMOCT064 - INACTIVE) 49 TE2/0/2/18 Clear (In use) (EC: VPMOCT064 - INACTIVE) 50 TE2/0/2/19 Clear (In use) (EC: VPMOCT064 - INACTIVE) 51 TE2/0/2/20 Clear (In use) (EC: VPMOCT064 - INACTIVE) 52 TE2/0/2/21 Clear (In use) (EC: VPMOCT064 - INACTIVE) 53 TE2/0/2/22 Clear (In use) (EC: VPMOCT064 - INACTIVE) 54 TE2/0/2/23 Clear (In use) (EC: VPMOCT064 - INACTIVE) 55 TE2/0/2/24 Clear (In use) (EC: VPMOCT064 - INACTIVE) 56 TE2/0/2/25 Clear (In use) (EC: VPMOCT064 - INACTIVE) 57 TE2/0/2/26 Clear (In use) (EC: VPMOCT064 - INACTIVE) 58 TE2/0/2/27 Clear (In use) (EC: VPMOCT064 - INACTIVE) 59 TE2/0/2/28 Clear (In use) (EC: VPMOCT064 - INACTIVE) 60 TE2/0/2/29 Clear (In use) (EC: VPMOCT064 - INACTIVE) 61 TE2/0/2/30 Clear (In use) (EC: VPMOCT064 - INACTIVE) 62 TE2/0/2/31 Clear (In use) (EC: VPMOCT064 - INACTIVE) So span 2 has slips, which could explain the HDLC aborts you saw previously. 3. As shown above, my box has two connections with PSTN (same provider for both): one direct, one through an HiPath PBX. So you've probably nailed it here. It *seems* like the HiPath PBX is regenerating the clock on the downstream port based on the other information. How can I double check timing slips don't come from inconsistency between both clock sources ? My first thought would be to unplug the link between
Re: [asterisk-users] How to read IRQs and timing slips values
On Fri, Jan 10, 2014 at 07:48:21PM +0100, Olivier wrote: With a single span directly connected to PSTN I'm still getting timing slips (140 slips/hour). Would you agree to qualify this rate as excessive ? Yes, this is excessive. Given these figures, may I also exclude an hardware failure inside my card or on the hosting machine ? In other words, how to detect a timing slip, Dahdi must use some inner clock as a reference, doesn't it ? Could this inner clock be presently broken ? You've configured the card to recover timing from the provider? If so, do your slips follow the actual cable that you were using to connect to provider or PBX? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to read IRQs and timing slips values
2014/1/10 Shaun Ruffell sruff...@digium.com On Fri, Jan 10, 2014 at 07:48:21PM +0100, Olivier wrote: With a single span directly connected to PSTN I'm still getting timing slips (140 slips/hour). Would you agree to qualify this rate as excessive ? Yes, this is excessive. Given these figures, may I also exclude an hardware failure inside my card or on the hosting machine ? In other words, how to detect a timing slip, Dahdi must use some inner clock as a reference, doesn't it ? Could this inner clock be presently broken ? You've configured the card to recover timing from the provider? I'm not sure but I don't think so as I've just configured the card with: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 span=2,2,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 echocanceller=oslec,32-46,48-62 Span1 is the one direct to provider equipement. Span2 is thh one that was connected to HiPath and which is simply unplugged If so, do your slips follow the actual cable that you were using to connect to provider or PBX? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to read IRQs and timing slips values
On Fri, Jan 10, 2014 at 08:34:57PM +0100, Olivier wrote: 2014/1/10 Shaun Ruffell sruff...@digium.com You've configured the card to recover timing from the provider? I'm not sure but I don't think so as I've just configured the card with: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 span=2,2,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 echocanceller=oslec,32-46,48-62 Span1 is the one direct to provider equipement. Span2 is thh one that was connected to HiPath and which is simply unplugged That looks correct. You might want to check your cables next. Do you only get timing slips when connected to the provider, or to the HiPath as well? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Text to Speech Engine
Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk API
Search google for Asterisk Manager Interface. For the most part, if you have raw Asterisk installed then that's what you get and have to build what you want on top of it or hire a developer to do it. Date: Fri, 10 Jan 2014 12:12:47 -0500 From: szilvertho...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk API Hello Folks; I have an Asterisk server Asterisk 11.7.0 built by root @xxx on a x86_64 running Linux on 2013-12-27 18:47:44 UTC No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. Is there an API out there that anyone knows of that I can pass commands, etc to Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, etc? I'm kind of clueless. Is there something available? Thanks - Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text to Speech Engine
Luminvox is one.. There are others out there.. Here's an article by Ward Mundy that might help:http://nerdvittles.com/?p=7448 From: jpra...@gmail.com Date: Fri, 10 Jan 2014 12:16:43 -0800 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Text to Speech Engine Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text to Speech Engine
Actually, scratch that.. Luminvox is not text to speech it's speech recognition software. Got this mixed up and turned around :-) Anyhow, see the link I posted earlier, it's got some good info to get you started. From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jan 2014 14:42:27 -0600 Subject: Re: [asterisk-users] Text to Speech Engine Luminvox is one.. There are others out there.. Here's an article by Ward Mundy that might help:http://nerdvittles.com/?p=7448 From: jpra...@gmail.com Date: Fri, 10 Jan 2014 12:16:43 -0800 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Text to Speech Engine Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 11.7.0: Delayed audio
On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@curtissystemssoftware.com wrote: On connection to an incoming call via PSTN where asterisk [11.7.0] is Dialing an internal extension on answering the call there is about 6-7 seconds before audio is heard on either side. When looking at the CLI traces when I answer the incoming call that asterisk extensions were dialing, I see immediately upon answering 0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:portnumber then not until about 6 seconds later I see this 0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:diffportnumber and immediately hear audio what appears to be an issue is that the RTP link(audio) setup is delayed. Anyone have suggestions on how to fix this issue? If the RTP source address/port is changing, then Asterisk is receiving RTP packets from two different sources and is waiting for one of them to stabilize before it picks the actual source of the media stream. That's by design, as the locking in of the RTP source prevents a media injection attack. You can tweak how Asterisk does this using two settings in rtp.conf: ; Enable strict RTP protection. This will drop RTP packets that ; do not come from the source of the RTP stream. This option is ; enabled by default. ; strictrtp=yes ; Number of packets containing consecutive sequence values needed ; to change the RTP source socket address. This option only comes ; into play while using strictrtp=yes. Consider changing this value ; if rtp packets are dropped from one or both ends after a call is ; connected. This option is set to 4 by default. ; probation=8 Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text to Speech Engine
Lumenvox is actually both... but hard to justify for TTS given all the freebies... On 01/10/2014 02:50 PM, Todd R. wrote: Actually, scratch that.. Luminvox is not text to speech it's speech recognition software. Got this mixed up and turned around :-) Anyhow, see the link I posted earlier, it's got some good info to get you started. From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jan 2014 14:42:27 -0600 Subject: Re: [asterisk-users] Text to Speech Engine Luminvox is one.. There are others out there.. Here's an article by Ward Mundy that might help: http://nerdvittles.com/?p=7448 From: jpra...@gmail.com Date: Fri, 10 Jan 2014 12:16:43 -0800 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Text to Speech Engine Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text to Speech Engine
On 10/1/14 8:16 pm, Jai Rangi wrote: Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. We recently used Ivona for a fairly complex IVR project (multi-lingual, including pronunciation of foreign names). http://www.ivona.com Not free, but we found the sound quality considerably better than we were able to get from either Festival or Cepstral. Worth bearing in mind that we are based in the UK, so our primary concern was for good quality British English voices. I cannot comment on other variants such as Australian or American. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text to Speech Engine
http://translate.google.com/translate_tts?tl=enq=i always find google translate works well http://translate.google.com/translate_tts?tl=frq=je trouve toujours google translate fonctionne bien On Jan 10, 2014 3:17 PM, Jai Rangi jpra...@gmail.com wrote: Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text to Speech Engine
Thank you every one, Yes google's translate is really good. http://zaf.github.io/asterisk-googletts/ But I dont like the fact that have to go over the wire every time. Looking for some thing to install on local server. -Jai On Fri, Jan 10, 2014 at 5:15 PM, Darryl Moore dar...@moores.ca wrote: http://translate.google.com/translate_tts?tl=enq=i always find google translate works well http://translate.google.com/translate_tts?tl=frq=je trouve toujours google translate fonctionne bien On Jan 10, 2014 3:17 PM, Jai Rangi jpra...@gmail.com wrote: Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 11.7.0: Delayed audio
On 01/10/2014 04:01 PM, Matthew Jordan wrote: On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@curtissystemssoftware.com wrote: On connection to an incoming call via PSTN where asterisk [11.7.0] is Dialing an internal extension on answering the call there is about 6-7 seconds before audio is heard on either side. When looking at the CLI traces when I answer the incoming call that asterisk extensions were dialing, I see immediately upon answering 0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:portnumber then not until about 6 seconds later I see this 0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:diffportnumber and immediately hear audio what appears to be an issue is that the RTP link(audio) setup is delayed. Anyone have suggestions on how to fix this issue? If the RTP source address/port is changing, then Asterisk is receiving RTP packets from two different sources and is waiting for one of them to stabilize before it picks the actual source of the media stream. That's by design, as the locking in of the RTP source prevents a media injection attack. You can tweak how Asterisk does this using two settings in rtp.conf: ; Enable strict RTP protection. This will drop RTP packets that ; do not come from the source of the RTP stream. This option is ; enabled by default. ; strictrtp=yes ; Number of packets containing consecutive sequence values needed ; to change the RTP source socket address. This option only comes ; into play while using strictrtp=yes. Consider changing this value ; if rtp packets are dropped from one or both ends after a call is ; connected. This option is set to 4 by default. ; probation=8 Matt Matt, What if any risk do i have with setting strictrtp=no with nat=no on a local network i.e.: 192.168.1.x ? pc -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
