[asterisk-users] e164 Format Numbers
This is probably a very simple question, but I can't for the life of me work it out. I'm trying to use Asterisk as a PTSN gateway to OCS (and believe I have all the SIP issues sorted), but OCS wants to dial in e164 format (+613blahblah). Because Asterisk sees the + in the SIP URI, it doesn't want to match anything in my dial plan, not even the S extension in the nominated context. Am I missing something completely obvious? ___ -- 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
[asterisk-users] Best phone for easy provisioning
Does anyone have any recommendations for a phone that has easy to understand/implement central provisioning? I've used CISCO 79XX phones, and they're great (but too expensive). I like Grandstream phones, but their provisioning sucks. What is everybody else using in large environments where individual config is not an option? Rod Bacon Technical Manager JASCO Consulting Pty. Ltd. http://www.jasco.net.au http://www.jasco.net.au/ Ph. 03 9432 6376 Fax: 03 9432 6378 ___ --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] AGI and Video
I've been using Asterisk for over a year now, and think I've pretty much got it nailed form a voice perspective. We have just purchased a couple of Video phones to start experimenting with IP video, with a view to eventually building an IP media platform, such as Intel's HMP. I have record/playback working in Voicemail, but am wondering what status Asterisk's AGI (specifically the record/playback fuctions) are at. A couple of other things that I'd like to be able to do with video in the short term; 1. Build a video menu system to overlay an IVR (eg. press 1 for blah, press 2 for something else, etc.) 2. Connect to an H.264 (or other codec) stream (eg. take a streaming feed from security camera, and attach it to an extension in the dialplan). I currently do this with audio, so I don't see video as being a huge extension to this. Has anyone got any useful links or documentation on any of the above? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --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 GXV-3000
I just got my 1st batch of GXV3000's. I can attest that the speakerphone is
every bit as bad as the GXP2000, perhaps even a little worse. Nowhere near as
good as Cisco. The other phone I personally found to be good for speakerphone
use us SNOM.
On Wednesday 31 May 2006 11:53, Paul C wrote:
Can you, or anyone else comment on the speakerphone ability of the GVX-3000
? We run the GXP-2000's and for the most part are happy with them, but
for upper management we're looking at phones with better speakerphone.
These would be ideal if the speakerphone isn't as terrible as the GXP-2000.
- Original Message -
From: The VoIP Connection [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, May 09, 2006 11:32 PM
Subject: RE: [Asterisk-Users] grandstream GXV-3000
Marek,
We have tested that that scenario and it works fine with the dev version
of
Asterisk. -Mike
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
-Original Message-
From: marek cervenka [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 09, 2006 8:46 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] grandstream GXV-3000
hi,
do you someone test this http://www.grandstream.com/y-gxv3000.htm?
video works? (it's have H264 video codec)
i want this topology
gxv-3000 - nat -{Internet}- Asterisk -{Internet}- nat - gxv-3000
---
Marek Cervenka
LCNA - http://lcna.slu.cz
===
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==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600 Fax: +613 99401650
FWD: 512237 ICQ: 5662270
==
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[Asterisk-Users] Dialogic Hardware
In case anyone is interested, I have a Dialogic D/600JCT-2E1-120 that we paid about A$15K for not so long ago. I am open to any serious offers. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --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] Simple windows / web Asterisk user software?
Our windows users are looking for a simple application to permit dialling and transfer from Windows desktop (or web page). I've looked at everything mentioned in the WIKI, and most are either not appropriate, or are not maintained any longer. I've used Flash Operator Panel, and quite like it, but I don't believe there is a way to have a per-user view (so people can only manage their own extension) so It's not really appropriate. ADM (Asterisk Desktop Manager) is close to what I'm after, but is still a little BETA for my liking. Any suggestions would be appreciated. ___ --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] Simple windows / web Asterisk user software?
I liked the look of it, but the documentation didn't mention transfer capability. Does it do transfers? On Tuesday 30 May 2006 10:27, Paul Hales wrote: Have you given SNAP a go? http://www.snapanumber.com/Home/tabid/53/Default.aspx ___ --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] SNOM 190 Daylight Savings
I've posted this to SNOM, but was wondering wheter anyone here has issues with SNOM 190 phones not showing the correct DST adjusted time (using the latest firmware). -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --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] Distinctive ring?
Has anyone found a solution to this? On Sun, 27 Nov 2005 01:46 am, Kristof Hardy wrote: Kerry Garrison wrote: pain to configure) have 4 ring types. I am guessing that I would need to figure out how to tell this particular phone to use a different ring tone unless there is a way to send a stutter type ring to the phones. Hi Kerry, I'm also using grandstreams on a few places, have the 'same' issue/question. Afaik it can't be done with the current Grandsteam firmware. (at least, you can't command the phone to use tone X, like you can do with Cisco's) You can use the phone's built-in Distinctive Ring Tone: setting (Advanced settings), but I'm not aware of any 'wildcard' you can fill in there, I only got it to work when filling in an 'exact' number. It could be that the next firmware (should have arrived end of oct) gives us distinctive ring tones and working hint leds.. Let's hope.. If you do find a way to get any working, please report back to the list, meanwhile, i'm eagerly waiting for the firmware :) cheers! ___ --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 -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --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] Grandstream NTP
All my BT101's and GXP2000's are failing NTP update. My NTP server is on my local LAN (and I've tried external ones), DNS is OK (and I've used IP address instead of DNS name). tcpdump on NTP server shows valid request, AND a valid response, yet the phones still display 02-01-1900. I have tried latest (and BETA firmware). Does anyone have any ideas? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --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 NTP
It now appears to be server specific. The shipped default, time.nist.gov, appears to work OK. Does anyone know of anything specific required by these grandstream phones as far as NTP server support goes? On Tue, 6 Dec 2005 10:34 am, Rod Bacon wrote: All my BT101's and GXP2000's are failing NTP update. My NTP server is on my local LAN (and I've tried external ones), DNS is OK (and I've used IP address instead of DNS name). tcpdump on NTP server shows valid request, AND a valid response, yet the phones still display 02-01-1900. I have tried latest (and BETA firmware). Does anyone have any ideas? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --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] Upgrading 1.0.9 to 1.2 beta
I have personally done this recently, and my advice is definately DO IT. In my situation, I noticed a marked improvement in echo and general audio quality. I too had gain settings that were out of whack compared to what others had experienced, but as long as your following the documented methods (ztmonitor, etc.) then whatever works for you is fine. One word of warning though, when I went to 1.2Beta2, my gain settings were all out of kilter again. Calls were suddenly far too quiet. I ended up setting them all back to 0 (you'll need to perform your tests all over again). This may be because of the different default echo canceller in the Zaptel drivers? Anyway... good luck. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Chris Bagnall wrote: Hello all, I'm contemplating upgrading a client's asterisk system from 1.0.9 to 1.2 beta to take advantage of some of the new echo cancellers in the later zaptel packages. Problem is, I'll be doing it without physical access to the box and without being able to personally test the new echo cancellation for them, so I'll be relying on information they provide me with. Their setup involves a Rev. I TDM400 card with 3 FXO modules connected to standard BT analogue lines. They've been complaining about echo for some time, despite the multitude of options I've tried in zapata.conf to limit the echo problem. Here are the current zapata.conf settings: echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=12.0 txgain=8.0 (rxgain and txgain calculated by running ztmonitor on a number of different calls over a period of a few days, aiming to keep the levels in the middle) 1) Is an upgrade to 1.2 likely to help at all? 2) If yes, which echo canceller is most likely to yield favourable results, and are there any changes I should make in the conf file? Thanks in advance. Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DSL router with QOS
What's the point? You can prioritise as much as you like at your own end, but as soon as it leaves your premises and enters the 'net, all bets are off! Even the contention ratio of the DSL circuit (as provided by your ISP) can kill you. QOS is really only useful in a point-to-point scenario, or in a meshed network that honors QOS on all links. If you really want to experiment, grab an old PIII for $50 off e-bay, and setup a linux box as a router behind your DSL modem. You can play with QOS as much as you like then, without forking out $500. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Keith Schmidt wrote: Any recommendations on an ADSL router with QOS for VOIP built in? Anything sub $500 would be great. Thank you Keith Schmidt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Changes from 1.2beta2 to 1.2RC-1
I've spent some time clicking my way around the digium website, but can't seem to locate a list of changes from * 1.2beta2 to 1.2RC-1. Can anyone point me in the right direction? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection in TE406P ??
Kevin P. Fleming wrote: There are other steps that can be taken if necessary first. Can you please elaborate on this? It may just save a lot of calls to Digium support about the same issue. (I have noticed this sporadically). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HDLC errors on PRI
The blanket statement that it's a motherbord or card problem is thrown around far too readily IMO. This is often just another way of saying that I have two (or more) pieces of hardware that don't play partucilarly nicely together in their default configuration, but I'm too lazy/busy/scared (I'm not accusing anyone here!) to sort it out. Linux offers a great deal of fine-tuning of the PCI bus and the interrupt subsystem, and you can get asterisk/zaptel to work well on most hardware with a little effort. Over the last six months, I have been through this extensively, and lost a lot of sleep in the process. I now have systems that I'd call solid. Start with digium's recommendations with regard to IDE tuning (HDPARM), framebuffer support, ACPI, and APIC. There is a document on their website about optimising your system to support their hardware. Then, look into SMP IRQ Affinity (if you have an SMP system, and wish to leave APIC enabled). You can also search for PCI latency, and see if this helps. Also, different versions of asterisk and zaptel can make a HUGE difference. I have had most success with 1.2Beta2 on my PSTN gateways, whilst still running STABLE on my IVR and VOIP servers. If you have a card that can be upgraded to the new firmware, do it. I've had all 3 of my 405P cards upgraded, and have noticed dramatic improvements in both performance and reliability. Use zttest every time you make a change, and also make sure that you unload/reload zaptel, the card drivers and asterisk each time you make a change. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Eric ManxPower Wieling wrote: Matthew Fredrickson wrote: Yeah, post your relevant portions of zaptel.conf. Usually it's a timing problem if you have HDLC abort errors. Actually, it's usually a motherboard or card problem that causes HDLC errors. Frequent causes are SATA controllers, IDE controllers, graphics modes, RAID controllers, GIGE controllers, etc. See the mailing list archives for the extensive discussions on this issue. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-488 FXO
I tried this unsuccessfully with an early (pre-release) version of the 488 firmware. I haven't tried it recently though. I'll have a play later in the week and let you know... == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Bill Michaelson wrote: Is anyone using a Grandstream ATA-488 FXO port to connect a PSTN trunk to their Asterisk box (via SIP, of course)? Is it possible to have such a beast operate reasonably? If so, is it also possible to use the FXS port concurrently and independently? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for Digium Boards
I have come across things like this before, but it's generally not an issue if you simply disable the onboard stuff that you don't need and select the PCI slot(s) wisely. I had the situation where my mobo allocated fixed IRQs to each slot, and shared IRQs between some of them (I can't remember the exact IRQs, but for arguments sake; 10, 11, 3, 5, 10, 11 in slot order - 6 slots, 2 sharing IRQs.) In most cases, IRQ 5 will be unused by anything on the mobo, giving one fixed, unused interrupt. This is where I placed my zaptel card. By disabling my COM ports, I was able to free IRQ3. You can also disable stuff like USB, Parallel, Audio, secondary IDE, etc. etc, which can all free-up IRQs. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == George Pajari wrote: FYI: We're trying to standardise on a tier one motherboard for the Asterisk boxes we build for customers and thought we'd try to use a low-end Intel Desktop Board since even a low-end Celeron has more than enough horsepower to handle a typical 8x32 PBX. To make a long story short, according to Intel Dealer Technical Support (we became Intel dealers in order to get answers to our questions) there is no Intel motherboard that permits the IRQs to be configured uniquely. They are all hardwired and shared. This information applies to both the Intel Desktop Board and Server Board product lines. Please let me know if your experience differs from what I've been told by Intel. Otherwise, you've been warned -- Intel mobos appear to be unsuitable for use with Digium hardware. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Missing audio from Zaptel channels - SOLVED!?
For those who are interested, the problem appears to NOT exist in 1.2Beta2. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Rod Bacon wrote: I have cross-posted this all over the place, and sent a copy directly to digium support, in the hope of getting to the bottom of a problem that has me pulling my hair out. I currently have 2 production PSTN gateway servers, running asterisk 1.2beta and TE406P cards (upgraded 405 cards, with hardware echo cancelers that we recently purchased on recommendation). We went to the beta version after installing the cancelers, as 1.0.9 kept segfaulting with the cancelers installed. Our PRIs terminate on a DMS100, at the same premises where our servers are co-located. Also in my farm, I have a dedicated IVR server, a VOIP gateway (SIP/IAX/H.323) and clustered MySQL servers running as FastAGI servers, to remove processor load from the PSTN servers. All servers are connected via gigabit Ethernet, and use IAX trunking for inter-server communications. I have been through _everything_ possible to be sure that I don't have any zaptel timing/irq problems (framebuffer, apic, acpi, smp irq affinity, irq latency, etc. etc) and have good zttest results with no frame slips, pops or clicks. After my PSTN gateway servers have been running for a few hours, I notice that some missing audio creeps into the start of each call (makes no difference if the call is ZAP-ZAP native bridge or ZAP-IAX). At best, you miss the first syllable of the first word. At worst, you can miss the first 3 or 4 seconds of audio. Further investigation shows that asterisk is lagging after the second leg of the call is answered (i.e. the time taken to bridge the channels gets longer). If the resultant call is a Zaptel native bridge, then the remaining audio is fine. If the resultant call is not zaptel natively bridged (eg. call is routed via another server, or asterisk remains in the media stream for another reason) then significant delay exists from one end of the call to another (simply put, asterisk seems to slow down). If I restart asterisk (even without removing and reloading zaptel drivers), calls are OK again for a period (typically around 12 hours). A workaround is to simply to install a cron job that periodically restarts asterisk when it's idle, but this is a less than ideal solution from my perspective. Something is definitely changing over time. A memory leak? Runaway process? I really need help in trying to troubleshoot this, as I've run completely out of both patience and ideas. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?
Sounds to me like that you want to log the phones into a queue, then simply logout the phones that you don't want to receive calls. If you were tricky, you could write a macro to log them in/out as they divert/undivert to/from voicemail. Eg. Dial an extension number to divert to VM (and log them out) then when they return, dial another number to do the reverse. Then simply route the calls to the queue using a ringall method. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == John Lange wrote: The first time I asked this to the list I didn't do a great job of it so I'm posting again with more details. Problem: when ringing multiple extensions, if one user has their phone forwarded directly to voicemail, it stops the whole group from ringing because the voicemail picks up immediately. Also, after hours incoming calls are to ring all extensions so anyone can pickup. But if one person in the office has their phone forwarded the same problem occurs. What we need is for asterisk, when ringing multiple extensions, to completely ignore the forward requests and just ring the remaining phones. Reading the source code I see there are two parameters for channels, allowredir_in allowredir_out. These offer me some hope that Asterisk has the ability but I couldn't figure out what these do or how to make use of them (I'm not a C programmer so maybe its just a red herring?). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-biz] New asterisk web gui for small/medium sized businesses
You asked for feedback, so here goes. Let me start by saying that I applaud your effort at getting involved in the project. I wish i could write a scrap of code, as there's literally dozens of things I'd like to contribute. Now... onto AWG. Personally, I'm not sure where this tool fits. It's too newbie for techos, and too techo for newbies. There are a multitude of other gui config tools floating around for SME customers, and some nifty user tools (like FOP, for example). snacktime wrote: I posted last week that I would get out a release asap, so here it is. Before I start in on putting up an actual website for it I thought I would put out a beta release to get things going. At this point there isn't a name for this project yet, as it's primarily an internal piece of software that we have been developing. For now I'll call it Asterisk web gui (AWG). AWG has a particular focus, which is to provide an easy to use interface for managing and monitoring asterisk, as well as a nice web interface for voicemail users as well. We are trying to make it as easy to install as possible. It plays nice with existing asterisk installations, and it won't overwrite any of your existing asterisk configuration. If you already have ruby installed on your system the setup time should be around 15 minutes. Once you have done one or two installations and know what the steps are, installation on a new system should average 5-10 minutes at the most. We have tested it on Freebsd and Debian. It should work on windows also. AWG is not intended to help you install asterisk and do your basic configuration. There are other software packages that do everything from start to finish such as AMP. AWG is dictatorial software. We will not include features by consensus unless they also fit our vision of what a tool like this should include. That said we want all the feedback we can get, particularly from businesses who are looking at it as a solution they might deploy for their clients. Just realize that it has a particular focus, and that's not going to change. There are also a few features not currently present that are on our todo list to get done asap. A basic interface for viewing CDR records, zap channel configuration, and a page to monitor real time information such as channels, peers, queues, etc.. There is a basic online demo at http://69.25.136.214:3000. The administrative login is user admin, password 'changeme'. The user login is username demo, password 'changeme'. At the moment there are not a lot of script templates installed, but the ones that are there will give you an idea of what you can do with the provisioning features. At the moment the demo is running in development mode where the errors are verbose and the code is reloaded for every page, so it's not as fast as you would see in a production environment. The setup guide and distribution file are at http://catalog1.paymentonline.com/~chris You can send any feedback to my directly at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] or on the list. The whole thing is licensed under the BSD license. ___ Asterisk-Biz mailing list Asterisk-Biz@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Missing audio from Zaptel channels
I have cross-posted this all over the place, and sent a copy directly to digium support, in the hope of getting to the bottom of a problem that has me pulling my hair out. I currently have 2 production PSTN gateway servers, running asterisk 1.2beta and TE406P cards (upgraded 405 cards, with hardware echo cancelers that we recently purchased on recommendation). We went to the beta version after installing the cancelers, as 1.0.9 kept segfaulting with the cancelers installed. Our PRIs terminate on a DMS100, at the same premises where our servers are co-located. Also in my farm, I have a dedicated IVR server, a VOIP gateway (SIP/IAX/H.323) and clustered MySQL servers running as FastAGI servers, to remove processor load from the PSTN servers. All servers are connected via gigabit Ethernet, and use IAX trunking for inter-server communications. I have been through _everything_ possible to be sure that I don't have any zaptel timing/irq problems (framebuffer, apic, acpi, smp irq affinity, irq latency, etc. etc) and have good zttest results with no frame slips, pops or clicks. After my PSTN gateway servers have been running for a few hours, I notice that some missing audio creeps into the start of each call (makes no difference if the call is ZAP-ZAP native bridge or ZAP-IAX). At best, you miss the first syllable of the first word. At worst, you can miss the first 3 or 4 seconds of audio. Further investigation shows that asterisk is lagging after the second leg of the call is answered (i.e. the time taken to bridge the channels gets longer). If the resultant call is a Zaptel native bridge, then the remaining audio is fine. If the resultant call is not zaptel natively bridged (eg. call is routed via another server, or asterisk remains in the media stream for another reason) then significant delay exists from one end of the call to another (simply put, asterisk seems to slow down). If I restart asterisk (even without removing and reloading zaptel drivers), calls are OK again for a period (typically around 12 hours). A workaround is to simply to install a cron job that periodically restarts asterisk when it's idle, but this is a less than ideal solution from my perspective. Something is definitely changing over time. A memory leak? Runaway process? I really need help in trying to troubleshoot this, as I've run completely out of both patience and ideas. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wanted to Swap! TDM400 FXO module(s) for FXS
Thanks for the suggestion, but in my experience fax machines on ATAs can yield unpredictable results, even at LAN speeds and uncompressed codecs. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Craig Guy wrote: Consider getting a PAP2-NA to connect your fax machine to - 2 x FXS ports for $99 - Original Message - From: Rod Bacon [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, October 26, 2005 8:46 AM Subject: [Asterisk-Users] Wanted to Swap! TDM400 FXO module(s) for FXS Does anyone out there have any TDM400 FXS module(s) that they want to swap for FXO (preferably in Australia). I have a quad-port FXO arrangement at the moment, but I need to plug a couple of fax machines into my * box... -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo canceller on TE406 Asterisk
I have similar problems with performance degradation over time. I'm about to post another message to the list (once I have some more information). Stay tuned. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Boris Bakchiev wrote: Hi, I have TE406P (2nd gen card with echo cancellation on-board). We still notice quite often echo on our PBX that is connected to one of the spans on TE406P (with calls routers to PRI provider on another span). I've tried to experiment with the echo cancellation on asterisk. I enabled echo cancellation in Zapata.conf to see if I can improve the situation and users started reporting warping bubble (description I got from one of the users) sound on calls from PABX-Asterisk-PRI (and other way). I was expecting that asterisk would disable its echo cancellation once it find on-board module. The strange thing I noticed that after system reboot things are now better. Although I cannot say for sure because the system was ever rebooted 2 times. Can anyone shed some light on this? Has anyone had similar problems? Or point me into right direction for troubleshooting? Regards Boris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wanted to Swap! TDM400 FXO module(s) for FXS
Does anyone out there have any TDM400 FXS module(s) that they want to swap for FXO (preferably in Australia). I have a quad-port FXO arrangement at the moment, but I need to plug a couple of fax machines into my * box... -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Samsung DCS 70
This may be a little off topic, but I'm hoping to find someone who knows something about integration with legacy phone systems, specifically a Samsung DCS 70. Our current service provider charges us a packet each time we want to make a small change, so I want to avoid using them to completely reprogram the entire system when I front-end it with an Asterisk box over the Christmas break. Does anyone out there have any experience with this model system? I (think) I have the correct software to re-configure it. I can get the password (and dial-in number) or access to the local RS-232 port. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Routing landline calls to asterisk.
Short answer: No, but... Long answer: Yes, and... Essentially, there are *certain* internal modems that will handle this function, but basically what you're talking about is an FXO card. You can pick up one for little outlay on eBay. Do a search on eBay for X100P. Then read the wiki for information on zaptel. Peter Ankerstål wrote: Is there possible to route ordinary landline-calls to the asterisk server and from there too our SIP-phones using a regular 56000 bps modem? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wrong caller id in CDR
I posted something on this a week ago, at which time I was told that this was an 'old' issue. Since then, I've spent hours looking, but can't find the answer. For some reason, some of my CDRs (both to CSV and MySQL) are being written with the wrong callerid. As best as I can determine, they are being written with the CLID of the _last_ caller to access the specific ZAP channel in question, not the current one. Has anyone ever seen this before? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI echo issues: solvable?
I'd be interested to know if this gets worse over time. Shutdown asterisk, remove card driver, load card driver, load asterisk then test. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == alan wrote: Hello, After solving the other low hanging fruit audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But occasionally, we get a call where the echo is delayed by a substantial amount (sometimes around 250ms), and sounds as loud as the remote party. One example: when one number (local to the same CO as our PRI) calls us, the echo on our end is unbearably bad. When we call them, No Problem. Am I right in guessing that we're unlikely to solve this in a system-wide manner on our end, and at best we'd have to convince the phone company they're misconfigured, for one remote phone number at a time? Some other specific questions: - Gain tuning: Is the ztmonitor quantitative target value 14500 or 14844? These two sources conflict on this point: http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html - Is the difference between 14500 and 14844 big enough to worry about? If my gain settings are incorrect, am I going to be seeing quantitative values a few hundred away, or ten thousand away (for example)? - Is gain tuning effective using CO or asterisk-local milliwatt sources useful on a PRI line? Presumably, the path to the local CO's milliwatt line is all digital, and the loopback path to call our own internal milliwatt source will almost definitely be all digital, so where would the loss come from? - Assuming gains are tuned correctly, or don't matter for the PRI, is there any other hope I have for solving these echo issues? Thanks, Alan Ferrency pairNIC pair Networks, Inc. [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enable mysql in asterisk
If you want my opinion, a single server (or even a small farm) is still easir to manage with conf files. A simple reload in the * CLI, and you're done. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Steve Daniels wrote: http://voip-info.org/wiki/ is your friend. More specifically: http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime HTH Steve P.S. Google should be your best friend, always ask him questions before the mailing list ;-) - Original Message - From: julien bos To: asterisk-users@lists.digium.com Sent: Monday, October 10, 2005 10:55 PM Subject: [Asterisk-Users] enable mysql in asterisk hi all expert, I am testing asterisk like small sip server, i installed asterisk in debian. It runs very well. I can use softphone to register, but each time i have modify the sip.conf, i find that it's not good way. So if i understand, avec asterisk version 1.2 i can use mysql to stock the information of the sip account. So can you show me how can i do that? Can you give me the link to document? Thank you so much. Julien ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wrong caller id in CDR
That I can live with. The main issue I have is that A calls B through a bridged zaptel call. The CLID on channels used by this call both show A's CLID - no problem. This call ends, and our system receives a test call from C on one of the channels used in the 1st call. Even though our application can see the correct CLID from C, asterisk still wites A's CLID in the CDR. This test call does nothing, except hangup as soon as it sees the call. Maybe by not answering the call, or allowing any time before hanging up, asterisk is not able to update the CLID on the channel? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == C F wrote: Yeah, meaning that when A calls B with callerID of 123 and B transfers A to C and while doing that the callerID is changed to 456 (callerID from B) then the CDR will show 456 is src. If you are trying to do billing based on this info then you are out of luck, as this is not accurate, rather look at account code as an option. On 10/11/05, Rod Bacon [EMAIL PROTECTED] wrote: I posted something on this a week ago, at which time I was told that this was an 'old' issue. Since then, I've spent hours looking, but can't find the answer. For some reason, some of my CDRs (both to CSV and MySQL) are being written with the wrong callerid. As best as I can determine, they are being written with the CLID of the _last_ caller to access the specific ZAP channel in question, not the current one. Has anyone ever seen this before? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wrong caller id in CDR
My preferred behaviour would be to completely erase the clid from the channel after the call. Obviously, this would only be done for zap channels configured with their clid set to asreceived. It would also only be desirable after any deadAgi scripts had run, or even by using a command as part of a deadAgi. (I already tried to set the callerid to in a deadAgi, it didn't work) == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == C F wrote: What you are saying is very interesting as well as important, I will investigate this. On 10/11/05, Rod Bacon [EMAIL PROTECTED] wrote: That I can live with. The main issue I have is that A calls B through a bridged zaptel call. The CLID on channels used by this call both show A's CLID - no problem. This call ends, and our system receives a test call from C on one of the channels used in the 1st call. Even though our application can see the correct CLID from C, asterisk still wites A's CLID in the CDR. This test call does nothing, except hangup as soon as it sees the call. Maybe by not answering the call, or allowing any time before hanging up, asterisk is not able to update the CLID on the channel? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == C F wrote: Yeah, meaning that when A calls B with callerID of 123 and B transfers A to C and while doing that the callerID is changed to 456 (callerID from B) then the CDR will show 456 is src. If you are trying to do billing based on this info then you are out of luck, as this is not accurate, rather look at account code as an option. On 10/11/05, Rod Bacon [EMAIL PROTECTED] wrote: I posted something on this a week ago, at which time I was told that this was an 'old' issue. Since then, I've spent hours looking, but can't find the answer. For some reason, some of my CDRs (both to CSV and MySQL) are being written with the wrong callerid. As best as I can determine, they are being written with the CLID of the _last_ caller to access the specific ZAP channel in question, not the current one. Has anyone ever seen this before? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with Wait SIP 486 DND
Or explained more clearly
The fallback rule is n + 101, so your voicemail busy priority needs to be 103
(2 + 101).
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
FWD: 512237 ICQ: 5662270
==
C F wrote:
Priority 103 should be dial and not hangup, that way it will do the
voicemail stuff with the wait as well.
On 10/11/05, Zack Odell [EMAIL PROTECTED] wrote:
Greetings,
I have implemented the following command to allow CNAM to be delivered to my
users.
exten = 9969,1,Wait(1)
This works great!
However it has spawned a new problem. When this is implemented into a full
dial plan. If a user is set to DND or sends a call to Voicemail by hitting
deny the caller gets a busy. Below is a result of the calls.
With the Wait(1) statement
-- Executing Wait(Zap/1-1, 1) in new stack
-- Accepting call from '3169321000' to '9969' on channel 0/1, span 1
-- Executing Dial(Zap/1-1, SIP/9969|20) in new stack
-- Called 9969
-- Got SIP response 486 Do Not Disturb back from 192.168.1.100
-- SIP/9969-d492 is busy
== Everyone is busy/congested at this time
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (default, 9969, 103) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Without the Wait(1) statement
-- Executing Dial(Zap/1-1, SIP/9969|20) in new stack
-- Called 9969
-- Accepting call from '3169321000' to '9969' on channel 0/1, span 1
-- Got SIP response 486 Do Not Disturb back from 192.168.1.100
-- SIP/9969-0c98 is busy
== Everyone is busy/congested at this time
-- Executing VoiceMail2(Zap/1-1, b9969) in new stack
-- Playing 'voicemail/default/9969/busy' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Channel 0/1, span 1 got hangup
This is the full dialplan for this extension -
exten = 9969,1,Wait(1)
exten = 9969,2,Dial(SIP/9969,20)
exten = 9969,3,Setvar(NewCaller=${CALLERIDNUM})
exten = 9969,4,SetCIDNum(${CALLERIDNUM})
exten = 9969,5,Dial(Zap/g1/13169321000,10,m)
exten = 9969,6,SetCIDNum(${NewCaller})
exten = 9969,7,Voicemail2(u9969)
exten = 9969,102,Voicemail2(b9969)
exten = 9969,103,Hangup
Now the CNAM is very important and I would love for this to work, so if
anyone has suggestions, please help.
Asterisk 1.0.5
Gentoo 2005.1
Quad-PRI card
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[Asterisk-Users] Zaptel Line Build Out
Can someone who is knowledgable in the traditional telco space please give me a layman's explanation (or point me to an appropriate url) of LBO as per the zaptel configuration file? # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel Line Build Out
Maybe I need to be a little more specific. I know what signal attenuation is. What I don't know, is how LBO (and specifically the implementation of it as used in the zaptel hardware/software) helps the situation. My servers are co-located with my carrier, and my PRI circuits are run through several patch panels, jumpers, etc. to another room, where they terminate on a DMS-100. I have asked the carrier for an estimated cable length, so i can correctly set the LBO. In the zaptel config, what is meant by DSX-1? What is CSU? Why would I use a -7.5db, -15db or -22.5db LBO? asterisk wrote: http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci211613,00.html - Original Message - From: Rod Bacon [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 09, 2005 6:42 PM Subject: [Asterisk-Users] Zaptel Line Build Out Can someone who is knowledgable in the traditional telco space please give me a layman's explanation (or point me to an appropriate url) of LBO as per the zaptel configuration file? # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.13/124 - Release Date: 10/7/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel Line Build Out
Yeah... sorta. So the CSU settings may be used when the E1 is pulled down to my premises, and I have a short cable connecting directly to the CSU device. I don't know why I'd need to change the LBO settings in that case, but I guess that doesn't really matter to me at the moment. In my case, I am approximately 40-50 metres (130-165 feet) from the switch (according to the telco's engineers), so an LBO of 1 in my span definition would theoretially be correct. asterisk wrote: http://www.adc.com/Library/Techpub/80348_1.pdf?refer=LibraryC=Copper_ConnectivityL=DS1_E1_Twisted_Pair_Products http://www.pcmag.com/encyclopedia_term/0,2542,t=DSUCSUi=42059,00.asp any help? Maybe I need to be a little more specific. I know what signal attenuation is. What I don't know, is how LBO (and specifically the implementation of it as used in the zaptel hardware/software) helps the situation. My servers are co-located with my carrier, and my PRI circuits are run through several patch panels, jumpers, etc. to another room, where they terminate on a DMS-100. I have asked the carrier for an estimated cable length, so i can correctly set the LBO. In the zaptel config, what is meant by DSX-1? What is CSU? Why would I use a -7.5db, -15db or -22.5db LBO? asterisk wrote: http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci211613,00.html - Original Message - From: Rod Bacon [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 09, 2005 6:42 PM Subject: [Asterisk-Users] Zaptel Line Build Out Can someone who is knowledgable in the traditional telco space please give me a layman's explanation (or point me to an appropriate url) of LBO as per the zaptel configuration file? # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.13/124 - Release Date: 10/7/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.13/124 - Release Date: 10/7/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delay before dialplan is launched?
For those who are interested... I set overlapdial=no, and things are all good again. D'oh! == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Rod Bacon wrote: I have several * servers, and have noticed something most annoying on recent builds. On an old server, running a CVS 1-0 checkout from 10th May, incoming zaptel calls (ISDN PRI) start processing the dialplan immediately, and look something like this... Accepting call from 'XX' to '' on channel 0/2, span 1 My other servers are running either the 1.0.9 stable tarballs, or 2.1 beta. They all include an additional line in the logs after the above line saying... -- Starting simple switch on 'Zap/X-X' On these servers, there seems to be a 2-3 second delay (approximately) between the time the call is received and the first dialplan command is executed. Is everyone else experiencing this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Latency on bridged PRI calls
Nobody has been able to answer this. Not even Digium at this stage, but I'm hoping someone here, smarter than I, will be able to. We are running some TE406P (upgraded 405Ps) cards performing mainly PRI bridged calls. After a server is brought up, calls sound absolutely perfect. Over time, delay (latency) creeps into the calls. What is really weird about that is apparently with the new Digium firmware, the native bridge is pushed down to the card, meaning the call never leaves the card (never hits the PCI bus). If this is the case, the latency must be being introduced in the card/driver. A restart of asterisk with removal/reload of the card driver fixes the problem (temporarily). This is in a production environment, and this is driving me insane. I'm about to try Sangoma cards, as I feel I'm really getting nowhere. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Latency on bridged PRI calls
Upon closer inspection, I don't think my system ever tries to establish a zaptel native bridge. Is there somewhere where this function is enabled/disabled? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Rod Bacon wrote: Nobody has been able to answer this. Not even Digium at this stage, but I'm hoping someone here, smarter than I, will be able to. We are running some TE406P (upgraded 405Ps) cards performing mainly PRI bridged calls. After a server is brought up, calls sound absolutely perfect. Over time, delay (latency) creeps into the calls. What is really weird about that is apparently with the new Digium firmware, the native bridge is pushed down to the card, meaning the call never leaves the card (never hits the PCI bus). If this is the case, the latency must be being introduced in the card/driver. A restart of asterisk with removal/reload of the card driver fixes the problem (temporarily). This is in a production environment, and this is driving me insane. I'm about to try Sangoma cards, as I feel I'm really getting nowhere. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clearing Caller-ID from Zaptel Channels
I have recently noticed that if you do a zap show channel XX on an on-hook channel (eg. no current call), then the channel shows the last known CLID. Under normal circumstances, this is overwritten when the next call comes in, providing a caller-id is received. If there is no caller-id, the call, at least from a cdr perspective, seems to inherit the identity of the previous caller. Is ths a bug, or by-design? Can this behaviour be modified? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clearing Caller-ID from Zaptel Channels
No. I'm running 1.2beta from Digium tarballs. Can you point me in the right direction? (the old thread?) == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Alexander Lopez wrote: This was addressed a while ago, are you running the lastest CVS??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Wednesday, October 05, 2005 9:38 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Clearing Caller-ID from Zaptel Channels I have recently noticed that if you do a zap show channel XX on an on-hook channel (eg. no current call), then the channel shows the last known CLID. Under normal circumstances, this is overwritten when the next call comes in, providing a caller-id is received. If there is no caller-id, the call, at least from a cdr perspective, seems to inherit the identity of the previous caller. Is ths a bug, or by-design? Can this behaviour be modified? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delay before dialplan is launched?
I have several * servers, and have noticed something most annoying on recent builds. On an old server, running a CVS 1-0 checkout from 10th May, incoming zaptel calls (ISDN PRI) start processing the dialplan immediately, and look something like this... Accepting call from 'XX' to '' on channel 0/2, span 1 My other servers are running either the 1.0.9 stable tarballs, or 2.1 beta. They all include an additional line in the logs after the above line saying... -- Starting simple switch on 'Zap/X-X' On these servers, there seems to be a 2-3 second delay (approximately) between the time the call is received and the first dialplan command is executed. Is everyone else experiencing this? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium hardware echo canceller, zapata.conf settings?
Do the echo cancellation settings in zapata.conf have any effect when hardware echo cancellation is installed on a 406p/411p? How can I tell if the echo is being cancelled by hardware or software? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM tts engine integration
Not bad.. but still not as good as Scansoft's... == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Tom Lynn wrote: On Sun, 02 Oct 2005 00:53:03 -0700, you wrote: I wrote a very very simple shell script and an even simplier macro to use the IBM TTS engine within asterisk for prompts. While its free you are limited on the number of requests you can do within a day. If anyone is interested its available at http://www.0xdecafbad.com/Asterisk-Text-to-Speech.html Nice solution, but what will you do if/when IBM pulls their demonstration page? Hopefully, by then you will have cached all of the necessary recordings. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] success story: TE406P (quadspan with hardware echocan)
Which version of asterisk and zaptel are you using? Will they work with 1.0.9 ? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Andrew Kohlsmith wrote: I just wanted to post here and let everyone know that the TE406P (quadspan T1/E1 with hardware echo can) kicks some serious ass. We've been running a PRI now for over a year with Asterisk (every single call in and out is through two Asterisk boxes, including faxes) and while the software based echo cancellation is more than adequate, we'd get the occassional edgy echo and very infrequently get full-out holy shit echo. So far the TE406 has eliminated that entirely. Anyway as I said I just wanted to post here and tell the world that at least as far as I have been able to determine, the extra cost of the hardware echo can is *well* worth the money. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo after running for several days?
There was a thread a while back about echo on calls increasing over time on FXO lines. I am finding this with TE405P cards as well. I had hoped that V2 of the firmware would fix the problem, but it would appear not). I know that there is a workaround script to restart asterisk and reload zaptel drivers, but i was hoping for a more 'solid' solution. Does anyone know of one? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clicks in audio with TE100P PRI
Which file does the jitterbuffer setting go in, zaptel or zapata.conf? I can't find it documented anywhere. What version of zaptel drivers include a jitterbuffer? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Alejandro G wrote: I tested all again. No matter if span=1,1,0 or span=1,0,0 if I configure jitterbufer=4 I have glitches that I'm almost sure that are holes in audio. If I raise jitterbufer=16 the problem disappear (or becames impercetible). Anyway I am interested in understand what is happening. Your issue is very likely the size of the zaptel jitterbuffers setting. If the zaptel driver is not immediately available to accept a frame of data it places it in an internal queue of pending writes. If that queue is full then the write is refused by the zaptel layer and then silently discarded by chan_zap causing a gap in the audio once it is played out of the zaptel card. If you crank up the debug level you will probably see 'Write returned -1...' (aka. EAGAIN) debugs that mostly correlate to the pops and clicks. Note that the zaptel driver legitimatly (if perhaps not appropriately) also refuses data when the channel is muted, such as during DTMF generation and at other times, so not _all_ EAGAIN debugs are a sign of problems. This makes perfect sense but again some issues of the problem do not match. I set debug at level 9 and there is no message of errors. Another thing I do not understand is why the same configuration: PAP2 - LAN - Asterisk - TE100P works perfect, and instead of LAN using internet generates the problem. Shouldn't it be the same for both configs? The only difference I see is that the rtp packets came from another Ethernet card, but if I call to terminate calls with another carrier using that eth works fine. What is clear is that jitterbuffer=16 corrects the problem. One more thing: no matter what codec I use, G729 or G711 the sound clicks are almost the same. Is anyway I could debug at RTP level in asterisk to see what is happening and check if there is packet loose? Thanks Alejandro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 changes?
Did you find a solution to this? Kib Eki wrote: yes, fedora 3 but without any changes at the sources Master Abi wrote: Are you using Redhat/Fedora? If I remember those init scripts is for Redhat/Fedora. I am using gentoo. Did you make any modifications to wct4xxp.c. or pass any parameters to zaptel. I see there is a #define SUPPORT_GEN1 in to wct4xxp.c which I commented out, but it made no difference. ztcfg seems to where the channels become unassigned. Thanks again. Kib Eki wrote: Hi, we also got one V2 TE405P card. It works fine now. At the moment we use for bridging the Pri to our old PBX. You must use zaptel, libri and asterisk v 1.0.8 or higher. We use 1.0.9 at the moment. zaptel: after make; make install i also executed make config. This copies the correct startup script to /etc/init.d/zaptel. Without this it also didn't worked for me. Master Abi wrote: Hi I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after putting in the upgraded board but did not change any conf, but the spans become active but will not come up. I guess I am missing something or are the any changes to the zaptel/libpri software that is required. I cannot find any info about this or does this new firmware only work with latest CVS. I am using 1.0.9 with 2.6.12 kernel Zapata Telephony Interface Registered on major 196 Found TE4XXP at base address fdfff000, remapped to f8928000 TE4XXP version c01a0164, burst ON, slip debug: OFF TE4XXP running with work queues. FALC version: 0005, Board ID: 00 Reg 0: 0x364e9400 Reg 1: 0x364e9000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE405P (2nd Gen) eth0: link up, 10Mbps, half-duplex, lpa 0x About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Unassigning channel 0/1! Unassigning channel 0/2! Unassigning channel 0/3! Unassigning channel 0/4! Unassigning channel 0/5! Unassigning channel 0/6! Unassigning channel 0/7! Unassigning channel 0/8! Unassigning channel 0/9! Unassigning channel 0/10! Unassigning channel 0/11! Unassigning channel 0/12! Unassigning channel 0/13! Unassigning channel 0/14! Unassigning channel 0/15! Unassigning channel 0/16! Unassigning channel 0/17! Unassigning channel 0/18! etc... This was working for 10 months before the upgrade. Master ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE405P V2 - Fantastic!
I anyone has any hesitations in upgrading their 405P (or 410P) to V2 of the firmware, read below; I installed one today (turnaround time around 2 weeks to Australia, inc. economy freight in both directions... impressive!) and have noticed immediate, significant improvements. Audio levels are better (have set tx and rx gains back to 0.0) and missed frames have gone (popping, clicking, etc.). Echo on bridged calls has also gone (I have now been able to disable echo cancellation on bridged calls, too!). I'm now rushing to get my other 2 upgraded. BTW: Make sure you have 1.0.9.2 zaptel drivers. The card didn't work with 1.0.9.1 drivers! I do NOT have the echo canceller module installed, as 90% of my calls are zaptel bridged calls. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Online TTS engines?
Can I get a copy of that PERL script? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Jeffrey Bird wrote: I have been using the web demo of ScanSoft's SpeechWorks - RealSpeak at http://www.scansoft.com/speechworks/realspeak/demo/default.asp. It has very nice output. I even managed to get a nifty perl script going that can do TTS from the command line for me. Jeffrey Bird Colin Anderson wrote: The one I like: http://www.rhetorical.com/cgi-bin/demo.cgi is toast. I think they went broke or got aquired by someone. Also, is there a Festival voice that sounds as good as Rhetorical or the AT T stuff? The default one is barely legible. Since Festival is a little brutal to configure, I'd like to get someone's recommendation then go through the pain of reconfiguring it only once. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zyxel Prestige 2000W Firmware - GOOD!
An update on this... I was wrong. The wireless problem was an altogether different issue. the wj0011 firmware finally made my phone useable, after 6 months of problems. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Rod Bacon wrote: If you see a wj0011 version of firmware for Zyxel Prestige 2000W floating around (I found it in a German forum), KEEP AWAY. It completely trashed the wireless networking in my phone. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] monitor peak channel use
I use Nagios to monitor everything on my network, including * channel usage. Google for the nagios check_asterisk plugin as a starting point. It's a simple perl script that could be run without nagios if desired (eg, from a cron job). It can connect to a server via the management interface, check channel usage and perform an action based on usage figures. We also have another script that does a similar thing, but connects and stays connected. It polls the server(s) every X seconds, and writes the usage into a MySQL database, so we have a live list of channel usage across all our servers. It also writes RRDs... == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Damon Estep wrote: Is there a way to trigger an action when a certain number of zap channels are in use, or is there a variable that stores max used channels that can be read? I use PRI for inbound calls, but outbound goes out via SIP, so the simple solution does not work. I need to know when the potential exists for inbound calls via PRI/Wildcard to be blocked because there are no more channels. Obviously asterisk would never know, since the call is blocked at the Telco, or is there still d-chan activity? Anyone know of a way to do this? Thx. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What have I misconfigured?
Why is each phone registering twice (2 different ports)? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Jonathan k. Creasy wrote: I'll change thatI was thinking minutes, don't know why, it's always secondsthat still doesn't explain why these phones are registering every 14-20 seconds -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tarpo, Louie Sent: Monday, September 12, 2005 7:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] What have I misconfigured? Your voIpProt.server.1.expires= value in the Polycom sip.cfg is set to reregister with Asterisk every 60 seconds. That's a bit much. Most people use 3600. Also check your maxexpirey and defaultexpirey values in your asterisk sip.conf. Louie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jonathan k. Creasy Sent: Monday, September 12, 2005 2:59 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] What have I misconfigured? I'm getting these messages every 7-10 seconds. -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP '532' at x.x.x.x port 56988 expires 60 -- Registered SIP '529' at x.x.x.x port 51444 expires 60 -- Registered SIP '529' at x.x.x.x port 64044 expires 60 -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP '532' at x.x.x.x port 56988 expires 60 -- Registered SIP '529' at x.x.x.x port 51444 expires 60 -- Registered SIP '529' at x.x.x.x port 64044 expires 60 -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP '532' at x.x.x.x port 56988 expires 60 532 started doing this last Thursday and 529 started doing it today. There are about 40 phones behind x.x.x.x. The two phones in question are Polycom IP301's with 1 line setup on them. There are 22 other Polycom IP301's there. -Jonathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 410P upgrade to 411P?
Does anyone know if the echo cancellation module can be retro-fitted to a 410P to turn it into a 411P? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Trunking Weirdness
I am having trouble getting trunking to work between a couple of my servers. All servers are running 1.0.9 stable version, and are working perfectly. All have a Zaptel card of some description, so timing is not a problem. Each server has a definition for each other server, using RSA auth, and qualify=yes (just so I can see ping times). trunk=yes is on globally, AND in each peer/friend definition. IAX2 SHOW PEERS shows a (T) next to each server, indicating that a trunk will be established. Servers on the same LAN segment seem to trunk OK. (IAX2 TRUNK DEBUG shows meaningful info). Conenctions where 1 server is behind a firewall works fine, but trunking does not seem to work. The only other difference is the codec. On the remote servers (the ones behind the firewalls), I'm running G729 (the Intel one). The LAN-based servers all talk to each other using G.711. Anyone have any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Trunking Weirdness
For those who are interested, the problem is being caused by IAX using the wrong outside IP address as it's source address. (Multi-homed firewall with numerous virtual interfaces). The workaround was to add additional definitiions to the other servers, so the connection will be trunked from the originating (incorrect) address. Apparently there is a patch against CVS HEAD which allows the definition of the source address on a per-peer basis. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Rod Bacon wrote: I am having trouble getting trunking to work between a couple of my servers. All servers are running 1.0.9 stable version, and are working perfectly. All have a Zaptel card of some description, so timing is not a problem. Each server has a definition for each other server, using RSA auth, and qualify=yes (just so I can see ping times). trunk=yes is on globally, AND in each peer/friend definition. IAX2 SHOW PEERS shows a (T) next to each server, indicating that a trunk will be established. Servers on the same LAN segment seem to trunk OK. (IAX2 TRUNK DEBUG shows meaningful info). Conenctions where 1 server is behind a firewall works fine, but trunking does not seem to work. The only other difference is the codec. On the remote servers (the ones behind the firewalls), I'm running G729 (the Intel one). The LAN-based servers all talk to each other using G.711. Anyone have any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 410P upgrade to 411P?
Did it make a lot of difference? Is the canceller effective? How much CPU will I save by doing it in HW? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Matt Florell wrote: Yes, it can. I just had one of my old TE405Pv1 cards upgraded to a TE406P(same process as TE410P to TE411P upgrade). The cost is quoted at $895US. You do need to send it to Digium though, not sure if they have a partner in AUS that is able to do upgrades or not. Just contact digium and request a RMA for a firmware upgrade and an echo-can daughter-board install. MATT--- On 9/8/05, *Rod Bacon* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Does anyone know if the echo cancellation module can be retro-fitted to a 410P to turn it into a 411P? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI in and out
I am wanting to front-end a legacy PBX with an asterisk box. I have done plenty of asterisk work over the last 6 months to PRI circuits, but not with a PBX being involved. I know I can use asterisk and digium cards in this manner, but do I need separate cards for the PRI - Asterisk side to the Asterisk - PBX side, or will a 4-port PRI card do the job? (I already have a spare one of these). In other words, can I use SPAN 1 as a timing source, then provide timing to the PBX connected to SPAN 2 of the same card? Any advice, or sample configs, would be greatly appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI in and out
It DOES help, thanks. Except for this the only difference between the first set of channels (1-23) and the second set of channels (25-47) is: signalling=pri_net group=1 context = fromprovider channel = 1-23 signalling = pri_cpe group=2 context=fromavaya channel=25-47 I thought the signalling setting was from the perspective of the * server, not the other side. For example, my PRIs to my provider are configured as pri_cpe, as I am the CPE. Your example seems to suggest the other way around. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zyxel Prestige 2000W Firmware - EVIL
If you see a wj0011 version of firmware for Zyxel Prestige 2000W floating around (I found it in a German forum), KEEP AWAY. It completely trashed the wireless networking in my phone. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I use MySQL in the dialplan?
You'll have a much more flexible solution if you keep your MySQL access out of the * dialplan, and put it in AGI. Matthew Boehm wrote: What the hell? NO! show application MySql app_addon_mysql is the name of the module. load app_addon_mysql.so -Matthew Quoting Ronald Wiplinger [EMAIL PROTECTED]: Matthew Boehm wrote: Ronald_Wiplinger wrote: I would like to put / get some data from an MySQL database. I want to use this MySQL database also via a web page. bye Ronald app_addon_mysql or use RealTime. *CLI show application app_addon_mysql Your application(s) is (are) not registered I want to use it for putting stored speed dial numbers into the per phone stored register, ... I guess I cannot get that with realtime done!!! bye Ronald Wiplinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940 - Disappearing Clock - SOLVED
P.S. I _had_ read the other posts that suggest changing to unicast sntp mode. This didn't help. I eventually setup a new ntp server on my LAN, and used it as a sync source. Everything seems OK now. Obviously a problem with my other ntp server. Cheers. Original Message Subject: Cisco 7940 - Disappearing Clock Date: Thu, 28 Jul 2005 11:50:35 +1000 From: Rod Bacon [EMAIL PROTECTED] To: asterisk-users@lists.digium.com This question is not actually * related, but please don't flame me! Is anyone out there using the 7.4 or 7.5 SIP firmware on their Cisco 79xx phones? I have a weird problem where my clock disappears after a period of time, and the only thing that will get it back is a reboot. Has anyone experienced this? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940 - Disappearing Clock
This question is not actually * related, but please don't flame me! Is anyone out there using the 7.4 or 7.5 SIP firmware on their Cisco 79xx phones? I have a weird problem where my clock disappears after a period of time, and the only thing that will get it back is a reboot. Has anyone experienced this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with compiling asterisk-oh323
I suggest you read the installation documentation again.
The error is telling you what the problem is.
You don't have pwlib and oh323 source compiled (using make opt only) and
sitting in the root/src directory. If it's somewhere else, edit the
asterisk-oh323 makefile to reflect the correct location.
wassim darwish wrote:
i ve downloaded
asterisk-oh323-0.6.6.tar.gz
I am getting this and anybody know howto fix this?
#tar zxvf asterisk-oh323-0.6.6.tar.gz
oh323]# cd asterisk-oh323-0.6.6
asterisk-oh323-0.6.6]# ls
asterisk-driver CONFIGURATION Makefile rpm
TESTS
BUGS COPYINGREADMErules.mak
wrapper
asterisk-oh323-0.6.6]# make
for x in wrapper asterisk-driver; do make -C $x build
|| exit 1 ; done
make[1]: Entering directory
`/home/wassim/asterisk-oh323-0.6.6/wrapper'
./check_ver /root/src/oh323/pwlib pwlib
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
./check_ver /root/src/oh323/openh323 openh323
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
cat: /root/src/oh323/openh323/version.h: No such file
or directory
cat: /root/src/oh323/openh323/version.h: No such file
or directory
cat: /root/src/oh323/openh323/version.h: No such file
or directory
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
g++ -Wall -x c++ -Os -DUSE_OLD_CAPABILITIES_API=1
-DWRAPTRACING -DWRAPTRACING_LEVEL=5
-DPWLIBVERSION=\..\ -DOPENH323VERSION=\..\
-I/root/src/oh323/pwlib/include
-I/root/src/oh323/openh323/include
-I/root/src/oh323/openh323/include/openh323
-I../asterisk-driver -c wrapper_misc.cxx -o
wrapper_misc.o
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'. Stop.
In file included from wrapper_misc.cxx:34:
wrapper_misc.hxx:35:19: ptlib.h: No such file or
directory
In file included from wrapper_misc.cxx:34:
wrapper_misc.hxx:61: error: expected class-name before
'{' token
wrapper_misc.hxx:63: error: `PMutex' has not been
declared
wrapper_misc.hxx:63: error: ISO C++ forbids
declaration of `PCLASSINFO' with no type
wrapper_misc.hxx:63: error: ISO C++ forbids
declaration of `parameter' with no type
wrapper_misc.hxx:68: error: `BOOL' does not name a
type
wrapper_misc.hxx:73: error: `PString' does not name a
type
wrapper_misc.cxx: In constructor
`WrapMutex::WrapMutex(char*)':
wrapper_misc.cxx:48: error: class `WrapMutex' does not
have any field named `PMutex'
wrapper_misc.cxx:50: error: `name' undeclared (first
use this function)
wrapper_misc.cxx:50: error: (Each undeclared
identifier is reported only once for each function it
appears in.)
wrapper_misc.cxx:50: error: `PString' undeclared
(first use this function)
wrapper_misc.cxx:51: error: `cout' undeclared (first
use this function)
wrapper_misc.cxx:51: error: 'class WrapMutex' has no
member named 'Class'
wrapper_misc.cxx:51: error: `endl' undeclared (first
use this function)
wrapper_misc.cxx: At global scope:
wrapper_misc.cxx:54: error: `BOOL' does not name a
type
wrapper_misc.cxx: In member function `void
WrapMutex::Signal(const char*, int, const char*)':
wrapper_misc.cxx:78: error: `PMutex' has not been
declared
wrapper_misc.cxx:78: error: no matching function for
call to `WrapMutex::Signal()'
wrapper_misc.cxx:77: note: candidates are: void
WrapMutex::Signal(const char*, int, const char*)
wrapper_misc.cxx:79: error: `cout' undeclared (first
use this function)
wrapper_misc.cxx:79: error: 'class WrapMutex' has no
member named 'Class'
wrapper_misc.cxx:79: error: `name' undeclared (first
use this function)
wrapper_misc.cxx:79: error: `endl' undeclared (first
use this function)
make[1]: *** [wrapper_misc.o] Error 1
make[1]: Leaving directory
`/home/wassim/asterisk-oh323-0.6.6/wrapper'
make: *** [subdirs_build] Error 1
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Re: [Asterisk-Users] PRI got event: HDLC Abort (6) on Primary, D-channel of span 1
2 - Check your span line in your zaptel.conf. You should be receiving timing, not giving it, when using a PRI (generally). Change the second number from 1 to 0. Save and restart asterisk. (span=1,0,0,esf,b8zs) I think you've got this cocked-up. A 0 in the second position tells zaptel to internally-clock the circuit, and ingnore the clocking information from the provider. Why would you want to do that? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DELL 2800 : PCI Parity error
I too had this problem, on a 2850, as well as the occasional missed IRQ. I went through all the usual zaptel tuning stuff Disabled fb, disabled ht, disabled acpi (left io-apic enabled), then moved irq affinity of zaptel card to second CPU so all interrupts from zaptel are on their own. My systems now run close to 100% in zttest, never miss an irq and don't seem to generate PCI parity errors any more. I don't know if I've fixed it, but you should really go through the whole process anyway. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == list wrote: Still not resolved On Wed, 2005-06-08 at 01:16, David John Walsh wrote: Frank Did you ever resolve this? If so what was the issue? On 03/05/05, list [EMAIL PROTECTED] wrote: Hi, I am struggling to get rid of a conflict on DELL 2800 : PCI Parity error (EB113 on the display) I am learning linux and asterisk as I go along, there might be obvious things I should know, but bear with me. From demsg below my 2 digium cards installed are listed (no config or connections done to digium cards yet), the conflict is with the TDM400P card, without that card, in any slot, no alarm. Zapata Telephony Interface Registered on major 196 Registered Tormenta2 PCI Controller version: 24 FALC version: TE110P: Setting up global serial parameters for E1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 Freshmaker version: 71 Freshmaker passed register test Uhhuh. NMI received. Dazed and confused, but trying to continue You probably have a hardware problem with your RAM chips Module 0: Installed -- AUTO FXS/DPO Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) wcte1xxp: Setting yellow alarm usb.c: registered new driver wcusb Wildcard USB FXS Interface driver registered TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) ramchip problem is false, without the card all ok, ramtests on machine as well. lsmod shows wcusb driver on zaptel, I dont need that, can I remove it? is that a problem or not? # lsmod Module Size Used byNot tainted usbserial 23964 0 (autoclean) (unused) lp 9156 0 (autoclean) parport38848 0 (autoclean) [lp] autofs416984 0 (autoclean) (unused) wcusb 19552 0 (unused) wctdm 41088 0 (unused) wcte11xp 22048 0 (unused) zaptel182080 4 [wcusb wctdm wcte11xp] e1000 77884 1 (autoclean) floppy 57552 0 (autoclean) sg 37388 0 (autoclean) microcode 6912 0 (autoclean) ide-cd 34016 0 (autoclean) cdrom 32896 0 (autoclean) [ide-cd] keybdev 2976 0 (unused) mousedev5688 1 hid22308 0 (unused) input 6176 0 [keybdev mousedev hid] ehci-hcd 20776 0 (unused) usb-uhci 26860 0 (unused) usbcore81152 1 [usbserial wcusb hid ehci-hcd usb-uhci] ext3 89960 6 jbd55060 6 [ext3] megaraid2 38344 7 diskdumplib 5228 0 [megaraid2] sd_mod 13904 14 scsi_mod 115112 2 [sg megaraid2 sd_mod] finally my interrupts, bit confusing to me, looks like I have dual processor, can see the NMI but what else can be found here? # cat /proc/interrupts CPU0 CPU1 0:32983953303167IO-APIC-edge timer 1: 3300 2876IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 8: 0 1IO-APIC-edge rtc 12: 236637 237965IO-APIC-edge PS/2 Mouse 14: 261779 262965IO-APIC-edge ide0 16: 0 0 IO-APIC-level usb-uhci 18: 0 0
Re: [Asterisk-Users] sound files
If you do a make install samples in the asterisk src dir, it will put them into /var/lib/asterisk/sounds Chadwick E. Labno wrote: where should the sound (.gsm) files be located? Currently the are in /usr/src/asterisk/sounds. I feel they should be located else ware, like in /etc/asterisk/sounds, I've copied a file into this directory but still no luck. What am I missing? Thanks Chad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not MRTG, what about ARGUS?
I use Nagios to monitor lines. I use the check_asterisk script that you'll find floating around the place. I connect via the mgmt interface. Added to nagios is nagiosgraph. This keeps historical RRD graphs of my line usage. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Carlos Alperin wrote: Ok, I got it. None use MRTG to track status history on Asterisk. Someone uses ARGUS? Any other tool? Someone track their lines? HEL Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HDLC bad FCS
Thankyou for an excellent post. Mike M wrote: Comments throughout. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HDLC bad FCS
I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server was experiencing crackly audio on one circuit, accompanied by HDLC bad FCS messages. The telco recabled and moved me to another port on the DMS-100. The audio is better, but there are still bad FCS problems on the span. I have moved the PRI in question to the other server, and the problem does indeed move with the circuit. There are no zaptel timing/interrupt problems present on either server. The fact that 3 PRIs are error free and that the problem moves with the circuit tells me that there is still a problem on the circuit. The telco believes that there is nothing more that they can do (provision a complete new circuit?). I don't get HDLC aborts, so the problem may not be _that_ serious. Does anyone have any comments? Would a newer (unstable) version of Zaptel drivers help? Would line-build-out parameter in zaptel.conf make any difference? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HDLC bad FCS
I think I need some help here. I didn't understand much of what you just said there. I though HDLC was a Layer 2 protocol? How can it have a location? FCS is a frame check sequence.. right? So I'm getting data out of sequence (or frames are missing from the sequence?) My gear is in a data centre, with the DMS-100 switch in the next room. Does this help? What is BER? Carlos Alperin wrote: I believe that you need to analyze the packets at your provider site. They should be able to do that. Is your HDLC located on your location or on your provider. This test should be done where Asterisk is running, because is where the problem is reported. Start to look for Line Analyzers for HDLC, in order to check BER. If BER is high enough, then the problem is internal on your server. Regards, Carlos Alperin Senior System Engineer Seneca Communications, LLC [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, July 04, 2005 7:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] HDLC bad FCS I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server was experiencing crackly audio on one circuit, accompanied by HDLC bad FCS messages. The telco recabled and moved me to another port on the DMS-100. The audio is better, but there are still bad FCS problems on the span. I have moved the PRI in question to the other server, and the problem does indeed move with the circuit. There are no zaptel timing/interrupt problems present on either server. The fact that 3 PRIs are error free and that the problem moves with the circuit tells me that there is still a problem on the circuit. The telco believes that there is nothing more that they can do (provision a complete new circuit?). I don't get HDLC aborts, so the problem may not be _that_ serious. Does anyone have any comments? Would a newer (unstable) version of Zaptel drivers help? Would line-build-out parameter in zaptel.conf make any difference? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No BUSY on PRI
I'm using a TE405P and stable version of Zaptel. When I call a BUSY number on my E1 PRI, I don't get a busy status. The caller hears a busy tone, but the CDR logs a NO ANSWER when the caller hangs up. Is this normal for this version of Zaptel? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel card AND Ztdummy together?
In case anyone is interested, loading Ztdummy AND a card driver at the same time will result in unpredictable timing issues. We heard intermittent echo/feedback on PRI channels. Rod Bacon wrote: I had a weird (unforeseen) situation today. We have a remote office with an * server and ISDN 10 service. We connect to each other over an IAX trunk with G729. Today, some of Sydney experienced a power surge which knocked out their ISDN services. Without a clock source on their PRI card, my IAX calls to them resulted in one-way audio (they could hear me, but I not them). Is it possible to load *both* the relevant card driver *and* ztdummy to guard against this occurrance? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel card AND Ztdummy together?
It's a Digium single-port job. No other timing sources aviailable (the * box IS the pbx). qrss wrote: What kind of card are they using? Is there only 1 telco circuit? If so, then I'm thinking their card should have detected the loss of service and switched to it's internal clock. Do they have a secondary clock source available across another circuit? Perhaps a tie line to a pbx that can be configured as a secondary? -Original Message- From: Rod Bacon Sent: Thu, June 23, 2005 12:03 am I had a weird (unforeseen) situation today. We have a remote office with an * server and ISDN 10 service. We connect to each other over an IAX trunk with G729. Today, some of Sydney experienced a power surge which knocked out their ISDN services. Without a clock source on their PRI card, my IAX calls to them resulted in one-way audio (they could hear me, but I not them). Is it possible to load *both* the relevant card driver *and* ztdummy to guard against this occurrance? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel card AND Ztdummy together?
I had a weird (unforeseen) situation today. We have a remote office with an * server and ISDN 10 service. We connect to each other over an IAX trunk with G729. Today, some of Sydney experienced a power surge which knocked out their ISDN services. Without a clock source on their PRI card, my IAX calls to them resulted in one-way audio (they could hear me, but I not them). Is it possible to load *both* the relevant card driver *and* ztdummy to guard against this occurrance? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Scratchy audio on Bridged PRI Calls
I have a number of servers with TE405P cards. The servers are DELL 1850's (which I _NOW_ see are listed on the digium not recommended page because of the ethernet interface). The problem I have is only during bridged calls. If I place a call into a service hosted on the box, or out to a VOIP phone, audio is crystal clear. If place a call through the box (a bridged PSTN call) the calling party hears some form of distortion when the other party speaks. Almost like a buzzing/crackling sound. I have been right through the interrupt sharing issue (disabled ACPI, APIC, Hyperthreading, Frame Buffer, etc. etc.) and am getting good results in zttest. I see NO IRQ misses or any other errors at the console. Does anyone have any other ideas? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Premptible Linux Kernel
Can anyone tell me if Asterisk would speficically benefit from the reduced latency of a preemptible Linux Kernel? I know it was recommended against in the early days, but I'm wondering if there are any updated opinions? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel HEAD with * Stable?
Will the CVS HEAD version of the Zaptel drivers work with the STABLE branch of *? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ee1000 Ethernet in Dell 1850
Digium's site now lists the Dell 1850 as a potential problem server, as it uses the intel ee1000 Ethernet chipset (as do a majority of servers in the market!). To my knowledge, ALL dell servers with Gigabit interfaces now use the same chipset. Does this mean the Digium cards can't be used in Dell servers unless you disable the onboard ethernet? I don't want to disable the onboard interface, as I use the IPMI management facility for lights-out management. I have a 2850 that doesn't have any audio problems (the reason that I contacted Digium in the first place), so I'm wondering if Digium are simply guessing at problems. Does anyone know anything specific about the supposed incompatibilities with the ee1000 kernel module? There seems to be an ever-growing list of situations where you can't use the Digium cards. This is a concern to me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream ATA Toasted
This is not an option for me, as the IVR menu is nuked as well... Luki wrote: A BETA firmware upgrade toasted my ATA286. It now has limited operations. Happened to me too... looked mostly dead, but not quite. Try a complete hardware reset. See section 8 on http://www.grandstream.com/user_manuals/HandyTone.pdf --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream ATA Toasted
A BETA firmware upgrade toasted my ATA286. It now has limited operations. It will get an IP address via DHCP and register to the last configured SIP server, but the web interface is gone as is the voice config menu. Apart from registration, there doesn't appear to be any other SIP functionality. An Ethereal dump does not show the device trying to grab a new firmware via tftp on bootup, so this is not an option either. Can it be fixed, or is it now rubbish? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztcfg server crash
Thanks for the info. Sergio Serrano wrote: Before change OS try to do next steps: first, stop asterisk. Second, you must do ztcfg -s to shutdown span. Unload modules, load modules if you need and do ztcfg -vv again. Start asterisk Regards Srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jason Walker Enviado el: martes, 14 de junio de 2005 6:07 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] ztcfg server crash I tried to get * stable on a 2.6xxx kernel for about 2 weeks. Then tried it out on a FC1 2.4.xxx kernel and found none of the issues. I am sure others have had success with 2.4.xxx, but I gave up;) BTW - I was using a TE110P and then a TE405P card for the zaptel install. Both were setup as T1s not E1s. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ztcfg server crash I am running Debian Sarge with a custom 2.6.11 kernel. I'll try building another kernel and recompiling the zaptel stuff. Jason Walker wrote: What OS/distro are you running? I experienced the same on Gentoo with the 2.6.xxx kernel. Switched to FC1 (2.4.xxx kernel) with the 1.0.7 CVS and have not had any issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztcfg server crash I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be partially screwed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 Packetization
Forgive this (possibly) silly question, but my upstream provider requires a packetization of 20ms. Using asterisk-oh323, I can set the number of frames per RTP packet. How does this equate to packetization in ms? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 Packetization
ok. I've worked out that G.711 is 1ms of audio per frame... what about G.729? Rod Bacon wrote: Forgive this (possibly) silly question, but my upstream provider requires a packetization of 20ms. Using asterisk-oh323, I can set the number of frames per RTP packet. How does this equate to packetization in ms? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 Packetization
I answered my own silly question. 10ms. If anyone needs a working OH323 config for Comindico (SPT) in Australia, please mail me (G.729 and G.711). Rod Bacon wrote: ok. I've worked out that G.711 is 1ms of audio per frame... what about G.729? Rod Bacon wrote: Forgive this (possibly) silly question, but my upstream provider requires a packetization of 20ms. Using asterisk-oh323, I can set the number of frames per RTP packet. How does this equate to packetization in ms? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztcfg server crash
I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be partially screwed. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztcfg server crash
I am running Debian Sarge with a custom 2.6.11 kernel. I'll try building another kernel and recompiling the zaptel stuff. Jason Walker wrote: What OS/distro are you running? I experienced the same on Gentoo with the 2.6.xxx kernel. Switched to FC1 (2.4.xxx kernel) with the 1.0.7 CVS and have not had any issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztcfg server crash I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be partially screwed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inject Audio into Existing Call
Other than using a conference, does anyone know of a way to inject audio into a live call between two parties? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server
On my * box at home (a dual PIII 1.2Ghz with 512Mb RAM), I'm running * (2 single-port FXO cards and SIP/IAX upstreams), MythTV (home theatre SW), file print services and other ancillary services. I have enough CPU grunt to decode video (watch DivX) and talk on the phone (inc transcoding). * on it's own is reasonably light on resources. Go for it! == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == Samy Antoun wrote: Hi, I'm planning to get my Asterisk box out of the LAN, get rid of my router and make the box acts as a Router, Firewall, DHCP Server (with Shorewall). I'll do that to be able to use some SIP clients remotely. Does anyone doing the same with the Asterisk box, is it a good idea, is there any other solution for the SIP emote Clients. Regards. __ Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CRM integration (was RE: CallerID)
This sounds remarkably like an IM problem We're in the process of building a CRM frontend that uses Jabber as the IM mechanism. The Asterisk server sends the URL via Jabber (PCs authenticated as extension number). The Jabber client (custom, written in Flash) receives the URL and automagically follows it. Michiel van Baak wrote: On 20:31, Sat 28 May 05, Gavin Hamill wrote: On Saturday 28 May 2005 20:21, Rusty Shackleford wrote: D'oh! I had misread the PP's statement and assumed he meant a bareback browser window. You are, of course, quite right. A Java app could handle this, but we are still left with the issue of having to install SOMETHING, even if it is a small Java app, on the client to make this work. What about this 'Ajax' stuff that's terribly trendy right now? It'd be a horrible polling implementation, but you could use a javascript Timer object to fire an XmlHTTPRequest every couple of seconds to check for new callerID at the IP address of the current browser? Cheers, Gavin. I dont do it with Ajax, but with my own written xmlhttprequest javascript. Did you check out my tar.gz file ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog Telephone Adapter
An IBM sales rep once told me... I can give you RELIABLE, FAST and CHEAP... any two of them at once. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] International Caller ID?
We have antiquated caller ID schemes here in Australia. We barely support numbers from other local carriers, let alone OS ones. Certainly no names either. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems trying to compile H323 from CVS-STABLE
Tony, I have managed to compile both versions (on separate servers, obviously), and have them working. My question is specifically related to which one do I choose?. To get the internal version working, I used the oldest versions of the libraries that I could find. Specifically, the 28th Aug 2003 builds from voxgratia.org (PWLib 1.5.3 and OpenH323 1.12.3) When I first loaded it, I DID get output from h.323 show codecs... now, strangely, it's empty. Also, h.323 show tokens reveals nothing. Call establishment, audio, call teardown all seem OK, but all calls seem to be in ULAW, no matter what I specify. The oh-323 channel seems OK, but doesn't like to play with certain codecs. I'm also concerned with the file handle situation you described. How can I debug this to see what is happening? (I'm not a programmer, so be gentle!) -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] more than one company hosting their PBX on the same machine?
Sigh... read the wiki. Search the lists. This has been answered at least fifteen times. You don't need multiple instances of *, just set up your dialplan properly. Hint: Contexts are the key. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which H.323 for Stable?
I'm new to H.323 and I have noticed that there are two separate channel drivers for * available - the inbuilt one, and oh-323. I had trouble compiling oh-323 with the current cvs stable, so I tried the inbiult one (with specifiec recommended versions of openh323 and pwlib). It compiled cleanly but I am told that it is not recommended (unstable?). Can someone with first-hand * H.323 experience offer any meaningful advice as to which way I _must_ proceed? This is for a live, busy, deployed environment. H.323 will be used to connect to an upstream provider (possibly CISCO gear?). Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Static on TDM Zaptel FXO
Make sure you have disabled framebuffer, apic and acpi. -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Clock Source
I apologise in advance if this is a silly question, as legacy telephone technologies are really not my forte. Is there an E1 card that can provide clock source? (E.g. Make my asterisk server look like a telco to my legacy PBX system?). What I am trying to achieve is the following: --ISDN---| Asterisk |---ISDN| Legacy PBX |-- -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Database of actve calls (as per astguiclient)
I have the need to maintain a pseudo-realtime database of active calls across a number of asterisk servers. The main purpose of this is in determining where to route calls (e.g. don't send calls to a server with no free lines) and also for monitoring/recirding calls. I know that astguiclient does this by telnetting into the * server management interface ever 333ms and updating a MYSQL database. Does this place much load on the * server, or the DB server? Will this sort of model scale to 30 servers, each with 120 Zap channels? -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P on Dell 2650
Are you running vga=normal in your lilo.conf? (disable frame buffer) and running kernel WITHOUT apic and acpi support? (append='noapic acpi=off). Making these changes, and disabling all other unused resources (to eliminiate IRQ sharing) got me to 100% consistently on a DELL 2850. -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_vpb Verbose Logging
Does anyone know if there is a way to turn DOWN the verbosity of the Voicetronix channel driver? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
