Re: [asterisk-users] Asterisk 1.2 + Zaptel 1.4 + HPEC = Crash?
Trevor Peirce wrote: Has anyone tried the combination of asterisk 1.2.24, zaptel 1.4.5 and HPEC 9.00.003? [snip] The result is a kernel panic followed by an automatic reboot. Nothing is written to log files so I cannot provide any debug information. As mentioned this has happened on multiple production machines and I do not have any other wctdm cards to test with. Although I can't say for certain that it's not Asterisk 1.2 vs. Zaptel 1.4 causing your issues, I've had crashes with HPEC on Zaptel 1.2 as well, and finally tracked these down to the HPEC binary overwriting bits of memory it shouldn't be touching. It may be the case that it's doing this for you on Zaptel 1.2, but due to code differences, isn't overwriting anything critical (for me, the crashes only started once we moved from kernel 2.6.15.7 to 2.6.20.x - the memory layout changed, HPEC would quickly corrupt the free list, and things would go boom upon the second deallocation of an HPEC EC). I've been working with Digium to get this found and fixed, so with luck there'll be an updated HPEC sometime reasonably soon. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Needed: Can't make local calls on a brand new PRI
Mark Engelhardt wrote: I just installed a PRI and when I make a local (seven digit) call, I get Code 28 back from the telco, (I believe code 28 means Invalid Number) and I hear a fast busy on the phone. Here is the output: -- Executing Dial(SIP/marke-17b1, ZAP/G1/4967171) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/4967171 -- Zap/23-1 is proceeding passing it to SIP/marke-17b1 -- PROGRESS with cause code 28 received -- Zap/23-1 is making progress passing it to SIP/marke-17b1 As you can see, asterisk is reporting 4967171 as the dialed number (which is valid) Cause code 28 means Invalid Number Format, and, if you're sure the number is correct, is often a case of you sending the wrong Type Of Number (TON) when setting up the call. Take a look at your pridialplan setting in zapata.conf - try unknown first, then other options until you find what works. A very very handy list I keep close to hand: http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Needed: Can't make local calls on a brand new PRI
Matt wrote: right and the asterisk debug is showing it going out as 7 digits whch the telco says is the way local should be dialed but yet the telco is seeing extra zeros on the end. we already know the ton is wrong...the question is where are the extra digits coming from. localprefix or similar perhaps? Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Wireless wrote: looks good. 9 says type [EMAIL PROTECTED] ~]# modprobe zaptel which returns nothing... when I run 10 At this point, if you run dmesg, do you find the following in your kernel log? Digium High-Performance Echo Canceller, version 8.20 Optimized for i386 CPU architecture Coypright (C) 2006 Digium, Inc. and Adaptive Digital Technologies, Inc. This module is supplied under a commercial license granted by Digium, Inc. Please see the full license text supplied by the accompanying register utility, or ask for a copy from Digium. If not, you've probably not got Zaptel built with HPEC properly. I'm truely stuck now, I cannot get HPEC to register with my Sangoma A200 card. I'm using Asterisk 1.2.15 Zaptel 1.2.13 Wanpipe drivers / util 2.3.4-7 I'm just not seeing any mention of HPEC in dmesg and I have tried different versions of the HPEC i386, i586, i686 and pentium3m the physical proc is a P3 650Mhz running CentOS 4.4 (Trixbox 2) I've rebuilt this box over the weekend from a fully patched CentOS 4.4 (yum update) as the hard drive failed! when I run ./register all seems ok then when I run ./zaphpec_enable it reports: No valid licenses for HPEC found. Any suggestions as to how I can debug what is not happening much appreciated Before building Zaptel, you are grabbing the correct version of hpec_x86_32.o_shipped for your CPU and putting it in zaptel-1.2.13/hpec/ right? It sounds to me like you've either not done that correctly, or something with the Sangoma build process is stopping the HPEC build working. After building zaptel, run strings zaptel.ko | grep 'High-Performance Echo Canceller' and see if you get a line like: Digium High-Performance Echo Canceller, version %s If not, you're going to need to dig into the way your Zaptel is being built to see why the HPEC module is not being included. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Andrew Kohlsmith wrote: On Thursday 15 February 2007 6:51 am, Steve Underwood wrote: It looks like octasic have started supplying their echo canceller as host software for zaptel now. I expect either canceller would work with the Sangoma cards, as they currently sit in the zaptel framework too. Out of curiosity, why do you suppose that it is the Octasic algorithm which is used in Digium's HPEC? I have no reasons to suspect otherwise, but I'm curious as to your reasons for suspecting that is indeed the case. I think Steve meant Octasic are _also_ now supplying their EC as host software for Zaptel. The HPEC canceller is from Adaptive Digital. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Wireless wrote: Does anyone know if the HPEC will work on a Sangoma A200 / 2 port FXO card? (I'm assuming so as it still uses Zapel) I've 2 PSTN lines one of which I cannot get rid of the echo, I've tried a 2GHz machine as apposed to my normal P3 650MHz and this made no difference. Would the 650Mhz be enough to run HPEC on one line (I assume only needing one licence) It should work, providing all the Wanpipe stuff is ready to work with Zaptel 1.2.13. As far as performance, you should be able to get one, maybe two channels of 1024 tap cancellation on the P3, but I'd advise careful testing, perhaps even using oprofile for a while to keep an eye on what's using what. You also have to watch out extra carefully due to the following: HPEC works in sparse mode, meaning it can cover 1024 taps, but just cancels echo in the parts where there is echo - hence CPU usage will likely change quite a bit with different echo paths - ie. a simple single reflection path will use less CPU than a complicated path with more than one reflection. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Dean Collins wrote: How do you fake echo for testing purposes then? All my tests have been done using sound files in userspace - I've written a few bits of code for doing this. Basically, the idea is you start with two sound files - speaker-A and speaker-B. Take speaker-A file, and run it through a finite impulse response filter (FIR) that has been preloaded with an echo path (ie. line echo characteristics). I use the various echo path models from the ITU G.168 specification for this, set at various pure delay offsets, and sometimes mixed together (ie. multiple paths at different offsets and amplitudes to simulate a variety of really nasty echo paths). The output is the speaker-A returned echo, and is saved to a file, then mixed with speaker-B so as to simulate doubletalk scenarios - resulting file called speaker-A-rx. My other tools wrap the various Zaptel echo cancellers into a userspace program, read .wav files of speaker-A and speaker-A-rx, run the echo canceller over them, and save the echo cancelled output to another file, which can then be listened to, spectrum analysed, etc. Testing the HPEC stuff was a bit more complicated, since it's a binary blob that requires licensing - I whacked up a quick'n'dirty Zaptel ioctl that takes bits of audio, feeds it through it, and passes it back, and uses rdtsc to keep track of CPU time used. Cheers, Nic. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nic Bellamy Sent: Thursday, 15 February 2007 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Wireless wrote: Does anyone know if the HPEC will work on a Sangoma A200 / 2 port FXO card? (I'm assuming so as it still uses Zapel) I've 2 PSTN lines one of which I cannot get rid of the echo, I've tried a 2GHz machine as apposed to my normal P3 650MHz and this made no difference. Would the 650Mhz be enough to run HPEC on one line (I assume only needing one licence) It should work, providing all the Wanpipe stuff is ready to work with Zaptel 1.2.13. As far as performance, you should be able to get one, maybe two channels of 1024 tap cancellation on the P3, but I'd advise careful testing, perhaps even using oprofile for a while to keep an eye on what's using what. You also have to watch out extra carefully due to the following: HPEC works in sparse mode, meaning it can cover 1024 taps, but just cancels echo in the parts where there is echo - hence CPU usage will likely change quite a bit with different echo paths - ie. a simple single reflection path will use less CPU than a complicated path with more than one reflection. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Wireless wrote: Thanks Nic, I have bought a couple of HPEC channel licences from Digium and been trying to get them working, all seems fine until I get to 9 and 10 of this doc ftp://ftp.digium.com/pub/telephony/hpec/README - at which point Asterisk is not running and I've issued a: wanrouter start command and all looks good. 9 says type [EMAIL PROTECTED] ~]# modprobe zaptel which returns nothing... when I run 10 At this point, if you run dmesg, do you find the following in your kernel log? Digium High-Performance Echo Canceller, version 8.20 Optimized for i386 CPU architecture Coypright (C) 2006 Digium, Inc. and Adaptive Digital Technologies, Inc. This module is supplied under a commercial license granted by Digium, Inc. Please see the full license text supplied by the accompanying register utility, or ask for a copy from Digium. If not, you've probably not got Zaptel built with HPEC properly. [EMAIL PROTECTED] ~]# ./zaphpec_enable I get - No valid licenses for HPEC found. If anyone can shed I bit of light on how to register my licence I'd be very greatful, I've checked in /var/lib/digium/licenses and there is a licence there. Hmm... not run into this myself - after registering my key, it worked first pop for me, giving the following output: # ./zaphpec_enable Digium High-Performance Echo Canceller Enabler Copyright (C) 2006, Digium, Inc. Version 1.0.0 Use the '-l' option to see license information for software included in this program. Found key 'HPEC-' for 4 channels. Found valid HPEC licenses for 4 channels. Successfully enabled 4 channels. After this, the follow line is spat out by the kernel: hpec_license_check: License granted for 4 channels Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
shadowym wrote: I gotta take issue with your comments that a HWEC is just software running on a DSP. In the case of Octasic, it's an ASIC. How it does EC is VERY different because.it's done completely in hardware, not firmware loaded into memory and run on a specialized CPU! Yes, the ASIC does contain an DSP but it is customized for EC. You cannot think of it as a CPU. You've obviously missed the fact that the Octasic chips have loadable firmware. Without being privy to any internal Octasic information, all I can guess is that their ASIC is a customised DSP core with perhaps more on-chip fast memory for FIR coefficient storage, and perhaps custom instructions/custom logic blocks specifically designed to improve the performance for the type of mathematical operations required for echo cancellation. Nobody in their right mind is going to do this entirely in custom circuitry - if you find a bug in your algorithm, or a way to improve things, what then - spend a few million on getting your chip rebatched and tell your users get their soldering irons out? Anyway, I'm going to shut up now before I get carried away and start a flame war :-) Cheers, Nic. -Original Message- From: Nic Bellamy [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 13, 2007 5:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) shadowym wrote: Interesting, Is this just a more advanced software echo canceller or software with hardware hooks or software with hardware assisted processing? A more advanced software canceller (there's no magical thing that makes hardware echo cancellers better, it's still software, but it's running on a DSP so it has more grunt available to it). It's licensed from Adaptive Digital Technologies - G.168 compliant, and supports up to 1024 taps (128ms) of tail coverage. Comes as a binary blob, but such is life. How would it compare to a true hardware echo canceller like the one Sangoma uses. Besides the extra CPU cycles required. Quite comparable - not sure if Octasic (as used by Sangoma and the latest Digium cards) or ADT would win in a shootout, but they're both in the same quality class. The main issue is going to be CPU usage - getting this going at 1024 taps on a full T1/E1 span would likely require two fast CPUs with the interrupts distributed evenly between them... and even then, *shrug* Cheers, Nic. -Original Message- From: Nic Bellamy [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 13, 2007 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Larry Shields wrote: I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? It's out now, and I've tried it - the difference between HPEC and MG2 from trunk is stunning - in situations with bad echo where MG2 can take ten or more seconds to converge to a reasonable degree, HPEC does it in perhaps 300ms - converging on my intake of breath before I say hello, and absolutely no echo after that unless I purposefully go out of my way to screw it up (whistling/blowing into the handpiece for instance - even then, the malfunction is minimal). You can now buy it from the Digium website (US$10 per channel), or if you have an in-warranty Digium card, email through the serial numbers to Digium support and they'll give you a key (this is what I did). You'll need Zaptel 1.2.13 to make it go. It does take quite a bit of CPU though - perhaps 70% more compared to MG2-trunk for the same number of taps from my rough measurements. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Larry Shields wrote: I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? It's out now, and I've tried it - the difference between HPEC and MG2 from trunk is stunning - in situations with bad echo where MG2 can take ten or more seconds to converge to a reasonable degree, HPEC does it in perhaps 300ms - converging on my intake of breath before I say hello, and absolutely no echo after that unless I purposefully go out of my way to screw it up (whistling/blowing into the handpiece for instance - even then, the malfunction is minimal). You can now buy it from the Digium website (US$10 per channel), or if you have an in-warranty Digium card, email through the serial numbers to Digium support and they'll give you a key (this is what I did). You'll need Zaptel 1.2.13 to make it go. It does take quite a bit of CPU though - perhaps 70% more compared to MG2-trunk for the same number of taps from my rough measurements. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Bill Gibbs wrote: Will this work with SIP channels? I get zero echo out the PRI but I do get it occasionally on a LD provider (SIP) we use. The stock * install doesn't appear to be doing anything stopping echo on those channels. Nope, it won't help - echo cancellation needs to be performed as close to the source of the echo as possible. When you've got SIP in the mix, you've got variable network delays, packet loss, jitter buffer interpolation and various other things to think about, and this would make an echo cancellers job orders of magnitude harder (and it's already a pretty hard problem). While it would be technically possible to echo-cancel SIP channels, it'd be _extremely_ CPU intensive (you'd need massive tail coverage) and probably not do a very good job. If you're getting echo from a VoIP-PSTN provider, they need to do something about it themselves - by the time it gets to you, it's too late. Cheers, Nic. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nic Bellamy Sent: Tuesday, February 13, 2007 3:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Larry Shields wrote: I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? It's out now, and I've tried it - the difference between HPEC and MG2 from trunk is stunning - in situations with bad echo where MG2 can take ten or more seconds to converge to a reasonable degree, HPEC does it in perhaps 300ms - converging on my intake of breath before I say hello, and absolutely no echo after that unless I purposefully go out of my way to screw it up (whistling/blowing into the handpiece for instance - even then, the malfunction is minimal). You can now buy it from the Digium website (US$10 per channel), or if you have an in-warranty Digium card, email through the serial numbers to Digium support and they'll give you a key (this is what I did). You'll need Zaptel 1.2.13 to make it go. It does take quite a bit of CPU though - perhaps 70% more compared to MG2-trunk for the same number of taps from my rough measurements. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customisable In-band ringing?
Ray Jackson wrote: Using SIP with progressinband=yes I get Asterisk to generate the ringing sound for callers. However, I was wondering if it is possible to configure what is 'played back' to the calling party? i.e. instead of just 'ring ring' could I potentially play back a song from an MP3, WAV or GSM file? I'm thinking it would be quite cool to offer a customised 'ring' sound while the caller is waiting for you to pick up? How can I do this with Asterisk or some external module perhaps? Any advice welcome! Hi Ray - LTNS - got the VoIP bug too now then? :-) The 'm' option has been mentioned, however that requires a separate music-on-hold class for each different sound, which would quickly become unmanageable. Another chap was working on a similar problem - have a look at the asterisk-dev thread titled app_dial.c modification started by Darren Nay on the 24th Jan 2007. I don't know if he's got it working yet, but it might be worth getting in touch with him. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
shadowym wrote: Interesting, Is this just a more advanced software echo canceller or software with hardware hooks or software with hardware assisted processing? A more advanced software canceller (there's no magical thing that makes hardware echo cancellers better, it's still software, but it's running on a DSP so it has more grunt available to it). It's licensed from Adaptive Digital Technologies - G.168 compliant, and supports up to 1024 taps (128ms) of tail coverage. Comes as a binary blob, but such is life. How would it compare to a true hardware echo canceller like the one Sangoma uses. Besides the extra CPU cycles required. Quite comparable - not sure if Octasic (as used by Sangoma and the latest Digium cards) or ADT would win in a shootout, but they're both in the same quality class. The main issue is going to be CPU usage - getting this going at 1024 taps on a full T1/E1 span would likely require two fast CPUs with the interrupts distributed evenly between them... and even then, *shrug* Cheers, Nic. -Original Message- From: Nic Bellamy [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 13, 2007 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Larry Shields wrote: I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? It's out now, and I've tried it - the difference between HPEC and MG2 from trunk is stunning - in situations with bad echo where MG2 can take ten or more seconds to converge to a reasonable degree, HPEC does it in perhaps 300ms - converging on my intake of breath before I say hello, and absolutely no echo after that unless I purposefully go out of my way to screw it up (whistling/blowing into the handpiece for instance - even then, the malfunction is minimal). You can now buy it from the Digium website (US$10 per channel), or if you have an in-warranty Digium card, email through the serial numbers to Digium support and they'll give you a key (this is what I did). You'll need Zaptel 1.2.13 to make it go. It does take quite a bit of CPU though - perhaps 70% more compared to MG2-trunk for the same number of taps from my rough measurements. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Only one out of 10 remote extensions expiring registry
Zeeshan Zakaria wrote: I have about 20+ phones on a server, all set for registry expiry 1 min. But only this one, with 2 accounts, keeps re-registerting itself. All the time this is what I see on asterisk CLI and it is kind of annoying. What only this phone does this and no other. Its on a remote location. All phones are Grandstream GXP-2000. -- Registered SIP '502' at 64.101.221.250 http://64.101.221.250 port 18639 expires 60 -- Registered SIP '7052823582' at 64.101.221.250 http://64.101.221.250 port 18641 expires 60 -- Registered SIP '502' at 64.101.221.250 http://64.101.221.250 port 18643 expires 60 -- Registered SIP '502' at 64.101.221.250 http://64.101.221.250 port 18647 expires 60 -- Registered SIP '7052823582' at 64.101.221.250 http://64.101.221.250 port 18649 expires 60 Notice how the port number keeps changing? Take a look at whatever is providing the Internet connection for the remote site - it probably has a dodgy NAT implementation. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13
Erick Perez wrote: PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme server for 8 more calls. Everything works fine in terms of the asterisk/meetme. The issue arises when the calls comes in via the ATA286 box and in any part of the meeting the CALLER hangs up but the ata286 does not realize the caller hung up so the channels remains open and everyone in the room hears a busy signal. After 30 seconds the ATA286 hangs up (I guess due to timeout) and then the tdm04b hungs the channel and then the meetme room gets back to normal. The ATA will be getting the hangup - it'll be what's generating the busy tone you hear when the SIP session between the ATA and your VoIP provider is terminated. If you can get your provider to enable the P205 Polarity Reversal setting on the ATA, the ATA will reverse the polarity of the voltage on it's FXS port when and outgoing call is answered (outbound calls), and when the remote end hangs up (for calls in either direction). You'll then be able to set hanguponpolarityswitch=yes in zapata.conf, and hangups should then be detected almost immediately (with luck, before any tones are heard). HTH, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lots and lots of log files
Ejay Hire wrote: Hello all, and good morning In my /var/log/asterisk directory I have 492,018 log files, most of which are empty. event_log.XXX queue_log.XXX messages.XXX where XXX is an integer. I removed them all and restarted asterisk a few days ago, but they came back. Take a look at the following bug, and see if it applies to you (Monitor() creating 2GB files, causing perpetual SIGXFSZ). http://bugs.digium.com/view.php?id=5984 If so, you should be able to figure out a way to mitigate the problem. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: SIP stuck channel soft hangup?
Martin Joseph wrote: On 2006-10-08 21:28:08 -0700, Nic Bellamy [EMAIL PROTECTED] said: Martin Joseph wrote: I am seeing occasional stuck SIP channels that seem to occur when the fricking Nokia E60 drifts out of WIFI range in the midst of a call. This is particularly annoying when the stuck channels include my PSTN gateway (wellgate 3701a), which leaves incoming and outgoing calls a busy signal. I see by googling that soft hangup is a good way to kill these channels and that works fine for me. I wonder if there is some way to automatically soft hangup these channels when the qualify fails? Take a look at rtptimeout in sip.conf - that might do what you need. Thanks again for the idea Nic! This does seem like a great way to do what I need, but it doesn't seem to work! I have added the statement rtptimeout=60 Into my extension for the Nokia E60. Then I reloaded asterisk. I tried just now to call through my gateway and then walk out of wifi range. The console continues to show me 2 active channels 1 active call, even after the minute (or several minutes) have passed? Any thoughts on why this doesn't work in 1.2.12? Hmm, this should work in 1.2.12 (I think it has for me). I'd recommend watching with tcpdump while you try this, as it's possible that your AP is picking up packets from your E60, but the E60 isn't getting them from the AP - in this case, as Asterisk will still be seeing the RTP, it won't time it out - even though it's dead from a users perpective. Can the other end still hear you at this point? There was a patch added a couple of months back, but this made it into 1.2.11: http://bugs.digium.com/view.php?id=7459 Depending on the state of the call, it won't always do the job - for instance if you're dialing but not connected, and the other end sends perpetual call progress tones. Asterisk isn't expecting any RTP at this point, so won't be able to do anything about it at this level. Even with this, if even one RTP packet gets through in that 60 seconds, it'll reset the timeout. Trying to make this more robust would get tricky, as we don't necessarily know what packetization interval the peer is using, so working on a % lost basis would be quite tricky. /braindump ;-) HTH, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP stuck channel soft hangup?
Martin Joseph wrote: I am seeing occasional stuck SIP channels that seem to occur when the fricking Nokia E60 drifts out of WIFI range in the midst of a call. This is particularly annoying when the stuck channels include my PSTN gateway (wellgate 3701a), which leaves incoming and outgoing calls a busy signal. I see by googling that soft hangup is a good way to kill these channels and that works fine for me. I wonder if there is some way to automatically soft hangup these channels when the qualify fails? Take a look at rtptimeout in sip.conf - that might do what you need. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
H.264 basic backport (was Re: [asterisk-users] Grandstream and H.264 !)
Carlos Chavez wrote: On Mon, 2006-09-04 at 16:26 -0300, Sergio (Red) wrote: hi, I´ need some help to implement the Grandstream GXV-3000 in my * platform. Someone know the state of H.264 Video Codec for Asterrisk?? This has been answered multiple times in the last month. Search the list before posting. You have to use the SVN version of Asterisk if you wish to use H264. I've been sitting on this for a while, but just got around to adding it to asterisk-backports.org: basic H.264 passthrough support for 1.2.11. It's at the Works for me with a pair of GXV3000s stage. http://asterisk-backports.org/wiki/index.php/Passthrough-h264 Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Grandstream GX-2000 phones for paging
Zeeshan Zakaria wrote: My client has all Grandstream GX-2000 phones in his office and he wants receptionist to use them for paging as well. Currently they are using Nortel and receptionist can easily do paging. He said that he had somebody setup their old Asterisk system in a way, that receptionist could dial an extension, after which her voice was heard on all grandstream phones' speaker phones. I want to know how to setup this type of feature on grandstream phones, i.e. dialing an extension will activate all phones' speaker phones. http://www.grandstream.com/FAQ/Asterisk.htm There's a PDF there that tells you (a) what settings to put on the phone, and (b) how to configure Asterisk to sent the SIP header that tells the phone to auto-answer. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager API: matching an Originate to the Newchannel event
Hi, I'm having a bit of trouble matching up Newchannel (and Newexten, etc. etc.) events with the Originate that created them. Basically, what I want to do is have software automatically initiate a call, and then track the status of that call through to completion. I can match to some degree with the Channel name in later events, but I can't see a way to do this that isn't inherently racey - ie. the person dials out, or someone calls in, at the same time as I'm doing my Originate, I'm not going to be able to match the events with any degree of certainty. Am I missing the obvious somewhere? Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue announcements when using ringback
Urban wrote: Hi, queue announcements works when we use music on hold in the queue, but if we use ringback e.g. queue(myqueue|r|) the announcments and hold time are not working, it seems that * is not even trying to read the queueu-announcment files. Is this by design, or is there a work around? Appears to be by-design, as the code specifically avoids playing announcements when the r flag is passed. Workaround that we use successfully: create/capture your standard ringing tone, and set up a musiconhold class that just loops it, and use this class for your queue. This way, the caller will hear ringing, but with the announcements breaking in as they should. HTH. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hitting # to Transfer out of a Queue
Andres wrote: Douglas Garstang wrote: I have dialled into a Queue, and an agent has answered the call with AgentcallbackLogin(). The agent hits #1, to transfer the call. Asterisk responds with 'Transfer', followed by dial tone. As soon as I enter a digit, Asterisk responds with 'I am sorry. That is not a valid extension' This is working for regular user-user dialling, not going through Queues. The queue app has Tt passed to it. You have to figure out what context the transfer is trying to use. In your case, the context does not have the extension your are dialing. First look at what context the agent is in, maybe thats the one being used. ... and when you find the correct context (a 'set verbose 99' before attempting the transfer should help you track it down - if it doesn't show you anything in the console, check the logs), you'll need to tell it what the correct context is. You do this by setting the TRANSFER_CONTEXT channel variable before going into the queue, eg. [somequeue] exten = s,1,Set(__TRANSFER_CONTEXT=from-internal) exten = s,n,Queue(foo,rt) (Whether the '__' prefix to make the variable inheritable is required will depend on your dialplan, but it Works For Me(tm).) Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hitting # to Transfer out of a Queue
Douglas Garstang wrote: Why? I don't want the variable to be global, or inherit to other channels. I only want it to be persistent for the current call in progress. It'll only inherit to channels *created from this one*, eg. Agent channels, Local channels and the like. It doesn't make it a global variable - see doc/README.variables for further information. Cheers, Nic. -Original Message- From: Andres [mailto:[EMAIL PROTECTED] Sent: Mon 7/17/2006 5:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Hitting # to Transfer out of a Queue exten = oe_ccare,1,NoOp(Queue oe_ccare called) exten = oe_ccare,n,Set(TIMEOUT(response)=5) exten = oe_ccare,n, GotoIfTime(8:00-17:30|mon-fri|*|*?one_queue_acd,oe_ccare-open,1) exten = oe_ccare,n,Goto(oe_ccare-shut,1) exten = oe_ccare-open,1, Answer exten = oe_ccare-open,n, Set(TRANSFER_CONTEXT=one_start) exten = oe_ccare-open,n(queue1), Queue(oe_custcare30) and so on... Anyway, when I press #1, and get 'pbx-transfer' followed by dial tone, as soon as I enter a digit, Asterisk still logs this to the console: Jul 17 17:07:53 VERBOSE[16439] logger.c: -- Unable to find extension '2' in context '' It seems that the TRANSFER_CONTEXT variable is not being set eventhough I am setting it... You need the 2 underscores infront of it__TRANSFER_CONTEXT -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad T1 Card
Michael Collins wrote: But the high dollars don't generally get you the high processing power, or a solid quality product (cough, Dialogic, cough). [snip] I'm looking forward to OSS telephony software (and the little guys with their little cards) taking the next step. The possibilities are intriguing, to say the least. [delurk] What I'd love to see is a reasonably grunty DSP available on the cards that is _user programmable_. There's some stuff a host processor isn't particularly good at (at least at present... most CPUs have an inbuilt FPU, but when do we get an inbuilt DSP?), and wouldn't it be nice to have 100% stable fax reception because all the time-critical processing is done in the DSP? Lots of fun stuff could be made to happen :-) Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
