Re: [Asterisk-Users] ECHO Tutorial
On Mon, 2006-06-19 at 18:45 -0400, Gary Reuter wrote: On 6/19/06, Daniel Salama [EMAIL PROTECTED] wrote: Is there anyone that could explain to me the phenomenon of Echo or at least point me where I can learn more? This paper by Cisco is a great start: Echo Analysis for Voice over IP http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml (it's the first result I get when I google for echo in voip) There was also a good article in LJ late last year: http://www.linuxjournal.com/article/8424 -Seth ___ --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] FW: NuFone Update: DIDs
Anybody have any recommendations? IAX service preferred. -Seth On Tue, 2006-04-18 at 15:48 -0400, Wes Baehr wrote: Well this is disappointing. Time to find somebody else... -- Wes -Original Message- From: NuFone Operations [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 18, 2006 3:44 PM To: [EMAIL PROTECTED] Subject: NuFone Update: DIDs Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier supporting the Toll-Free and Michigan DID operations of NuFone, has threatened to terminate our services. We were only informed of such decisions on Thursday Afternoon, April 13th, 2006, which as anyone knows is abolutely not enough time to solve the complex matters at hand. Initially, we were hopeful that we could work out an acceptable deal, at the very least, to allow us enough time to move your telephone numbers to another carrier without much, if any service interruption. Sadly, this no longer seems to be the case. We are currently working with another carrier to host your Toll-Free numbers. However, you may want to consider submiting a Number Portability request to another carrier, to avoid any service outages caused by Telesthetic's threat to terminate our service. We can always port your number back to our service at later date, at no cost to you. If you would like to keep your Michigan telephone number you will need to contact Telesthetic directly at 248-724-0600 to determine if they will provide you service or not. We are going to do everything we can to survive yet another failed business partnership. We do not intend to give up. We can and will prevail through yet another time of great challenge. We will update you as soon as we have more information. Thank you for your continued support and words of encuragement. The NuFone Network http://www.nufone.net/ ___ --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] New Sangoma AA Series?
Hello All, I saw an add in my latest Linux Journal advertising Sangoma's new AA series of FXO/FXS analog cards with on-board echo cancellation, but I can't find any information at all on them. Even the link given in the advertisement is a dead end as far as I can tell. Anybody else seen/heard anything about this? -Seth ___ --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] How to 'read' ztmonitor and set gains
On Fri, 2005-07-15 at 20:44 -0700, Chris Coulthurst wrote: Being one the many people trying to track down echo 'ghosts' I ran across this page: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html suggesting ways to adjust the gain. I have a TDM400P 2x2 config with Kewlstart lines configured. I've located a local telco milliwatt test line, and when I call it, the gain numbers are no where near 14844. Now, this article refers to configuration with a channel bank, but suggests it would be similar on 'simpler equipment'. The numbers I get are around 4450 on the Rx. [EMAIL PROTECTED] zaptel]# ./ztmonitor 5 -vv Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) * Rx: 4452 ( 4452) Tx:46 ( 63) Is this good? Normal? Any suggestions, or a point in the right direction for the right documentation would be appreciated. P.S. This is CVS-HEAD Zaptel on a P3 550, Host bridge: Intel Corp. 440BX/ZX/DX - 82443BX/ZX/DX Host bridge (rev 3) Chris Coulthurst [EMAIL PROTECTED] I tried to follow the instructions in that message as well and my findings were the same as yours. I forget what they were exactly but by the time I boosted my gains to approach a reading of 14844 I was boosting by some insane db level and, as would be expected, my signal was severely distorted and way outside the range of what the mec2 echo canceler could handle. The only thing I noticed that might indicate where the 14844 number came from is that if you monitor a channel that is executing the Milliwatt() app it will report a tx level of. 14844. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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 to 'read' ztmonitor and set gains
On Fri, 2005-07-15 at 20:44 -0700, Chris Coulthurst wrote: Being one the many people trying to track down echo 'ghosts' I ran across this page: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html suggesting ways to adjust the gain. I have a TDM400P 2x2 config with Kewlstart lines configured. I've located a local telco milliwatt test line, and when I call it, the gain numbers are no where near 14844. Now, this article refers to configuration with a channel bank, but suggests it would be similar on 'simpler equipment'. The numbers I get are around 4450 on the Rx. [EMAIL PROTECTED] zaptel]# ./ztmonitor 5 -vv Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) * Rx: 4452 ( 4452) Tx:46 ( 63) Is this good? Normal? Any suggestions, or a point in the right direction for the right documentation would be appreciated. P.S. This is CVS-HEAD Zaptel on a P3 550, Host bridge: Intel Corp. 440BX/ZX/DX - 82443BX/ZX/DX Host bridge (rev 3) Chris Coulthurst [EMAIL PROTECTED] ___ 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] Festival questions
On Wed, 2005-07-13 at 23:34 -0700, Jason Walker wrote: Has anyone had any luck in changing the voices for Festival and Asterisk? I have Festival installed and working, but can not get the voice different from the default. Thanks, Jason Well, it's been a while since I had to do this so I'm going from memory, but I think you can change the default voice to a different one (assuming you have the voices installed correctly) from the voices.scm file. Look for default-voice-priority-list in that file. The first one in the list is the default. -Seth ___ 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] Flash and zap and # key
On Fri, 2005-07-01 at 08:01 +0200, Wilson Pickett wrote: I'm not sure if this is supposed to happen, but when I press the # key it seems to have the effect of flashing the hook, or at least letting me transfer. I am using Zap hardware. Do you hear a transfer voice prompt? Asterisk will intercept the # key if told to do so in the Dial application. In STABLE this is hard-coded I think. show application dial will give the details of how T and t can be used as options. If your dial commands are not using T and t as options, I'm not sure what is happening. If you are running CVS HEAD instead of stable you can set the transfer key(s) in features.conf. There is also a double ## patch floating around somewhere for STABLE I think. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Developing an Application in Asterisk
On Thu, 2005-06-30 at 18:35 +0530, Bharat M. Sarvan wrote: Hello Everybody, Can any one guide me in developing an application in Asterisk? Just as Dial ( ) is a registered application. So that I can develop modules of my own.If anybody has done it, kindly please let me know. Regards, Bharat M. Sarvan app_skel.c has all the boilerplate code you need. -Seth Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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: Good soft-phone on Linux
On Wed, 2005-06-29 at 10:40 +0200, Filippo Carone wrote: * Hamish Whittal ([EMAIL PROTECTED]) ha scritto: Hi Folks, I am wanting advise on a good soft-phone on Linux. I have looked at Gnophone but cannot seem to get it to compile under debian sarge. I am now looing at sipXphone seem to be picking up that it is not that stable, but perhaps someone here can advise on what softphone I can use on Linux. it may be more of what you need but using asterisk with the OSS/Alsa module turns it in a very efficient client (it can run also without X installed ;) X-Lite for Linux has been working fairly well for me. http://www.xten.com/index.php?menu=productssmenu=download -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] FOP related questions
On Wed, 2005-06-22 at 15:58 +0200, Daniel ANDRE wrote: Hello, I have downloaded and installed Flash Operator Panel. version 0.21. It works pretty well and I have some questions about it. 1. The text label of the buttons are partially hidden by their icons. Is there a way to adjust right margin for the buttons? Take a look at the op_style.cfg file. You'll probably find something to adjust in there that will help your issue. 2. I would like to have the fop brought in the front of screen whenever and extension rings. Sort of crm feature but with fop and not another url. Is there a way to do that? Not out of the box that I'm aware of. Would probably require a change to the flash code. 3. This question is notre directly related to fop but you may have the answer. I would like to have fop panel in tis own windows (no toolbar, menu, title, ...) either with FireFox and Internet Explorer. Any Idea? With firefox you can turn off the navigation and bookmarks tool bar and set it to run full screen. Make sure you have the Hide the tab bar when only one web site is open option selected in the preferences. Maybe not exactly what you were looking for. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] voicemail
On Thu, 2005-06-23 at 23:19 -0400, Michael Di Martino wrote: I am trying to setup voicemail for my iaxy device, however, i cannot get it to work voicemail never picks up. Below is my config. Am i doing anything wrong here From my Extensions.conf file exten = 7403,1,Dial(IAX2/7403/10) You did not specify a timeout in the dial command. Change it to: exten = 7403,1,Dial(IAX2/7403/10,xx) --- where xx is the number of seconds you want the Dial command to attempt to connect the call before it returns and proceeds to the next priority (i.e. voicemail). exten = 7403,2,Voicemail(u7403) exten = 7403,102,Voicemail(b7403) exten = 7403,103,Hangup From my voicemail.conf [telx.com] 7403 = 7403 Thanks Mike Hope that helps. -Seth ___ 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 HEAD with * Stable?
On Mon, 2005-06-20 at 12:28 -0400, Doug Lytle wrote: Florian Overkamp wrote: Hi, -Original Message- Will the CVS HEAD version of the Zaptel drivers work with the STABLE branch of *? Err, why specifically would you want that ? Florian, In our case, the CVS drivers (At the time that I did it) showed enhanced information coming across our Definity PBX, before we wern't getting CID info and we are now. Doug I'm about to do the same thing because I would like to try the fxotune utility. I would also like to use the numerical representation patch to ztmonitor which isn't in stable. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Line Noise HELP!
On Mon, 2005-04-11 at 17:06 -0400, Andre Normandin wrote: Hi, I'm having very similiar problems.. However, I'm running a development version, and it happens on both SIP phones, and on Analog phones connected via Sipura SPA-2000's (I have 2 different SPA2000's, and 4 analog lines.. Seems to happen on all of them as well).. The problem seems to be EXACTLY as described. THe call seems fine at first, then within minutes the call degrades to the point that neither end can hear each other.. First, the volume seems to lower, and then static, breaking up, etc.. I have both DIGIUM X100 cards for my pots lines (3 of them), and BROADVOICE for outgoing calls. It seems to happen no matter if I'm on an analog line (I.E. someone called me), or if it was me that initiated the call (BROADVOICE outbound). I do have a 'remote' SIPURA SPA2000 located at a friends house in a different state -- he is an extension on my pbx so he can call me, and he can call his friends locally (He just moved away) via my POTS or BROADVOICE line.. He experiences the same problems as I described above, unless he calls me directly at my 'internal' extension, or I call him at his 'internal' extension.. I.E. If it doesn't touch POTS or BROADVOICE, the problem doesn't seem to occur..?? The other interesting thing that has happened of recent development is that some people are complaining that they are hearing the 'electronic beep' sound as if the call is being recorded, but I am not recording the call. This has occured with my friend as well as incoming and outgoing POTS/BROADVOICE calls. If anyone has an idea, I'd love to hear it.. The problem is driving me (and others who talk to me) crazy!!! - Andre I'm not sure about your other problem... but I have heard others complain about beeping with a Sipura ATA. Had something to do with it incorrectly detecting DTMF when there was none. I believe it was fixed in the newer firmware. You might try updating the firmware and see if that fixes it. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Debugging Asterisk in GDB (DDD)
On Mon, 2005-03-28 at 15:04 -0800, Jay Ray wrote: Hi, I am running Asterisk on Fedora Core 3. I am trying to use DDD to debug Asterisk. However, when I attach the debugger to the Asterisk Process, the Asterisk CLI promt hangs. Does not give any output, and Asterisk stops processing calls... What could be wrong and what is the best way to debug Asterisk...? You might want to ask this question on the asterisk-dev list since those there are probably more familiar with a debugger + Asterisk. I have personally run Asterisk through gdb once or twice and never had a problem, but I started the process from within gdb. I've never tried to attach to an already running Asterisk process. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Remove a channel from receiving inbound calls
On Mon, 2005-03-28 at 12:02 -0500, Bob Sowers wrote: Ive got a small office setup with a TDM400 and 3 FXO cards. Id like to take away the ability of the 3rd FXO to receive calls (as this line runs through our old-fashioned fax machine) BUT still be able to use it for outbound calls. With our original, and very basic PBX we could modify the auto attendant on a particular PSTN line to pick up after 4 rings (which would allow the fax to pickup after 2) while the rest of the lines picked up after 1 ring. Anyone have a simple way for me to do this? No need to go to all that trouble. Zap can be set up to automatically detect fax tones. Then you can configure incoming faxes to be directed to you fax machine and do whatever you want with everything else. Gone are the days of dedicated fax lines. http://www.voip-info.org/tiki-index.php?page=Asterisk%20fax Look at the faxdetect stuff. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Start on system restart
On Mon, 2005-03-28 at 17:58 -0400, Chris Mason wrote: How should I get asterisk to start automatically on system restart? http://www.voip-info.org/wiki-Asterisk+Starting+and+Stopping Look at the section Starting Asterisk at Boot Time. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Square Key system
On Fri, 2005-03-25 at 08:00 -0800, Mark W Wood wrote: I have searched both the wiki and googled looking for a solution to a square key configuration. I need to have C.O. lines to appear on the buttons to facilitate a small office. All of the users can see each other and calls are put on hold and picked up by the other users instead of transferred. Has anyone done this? Can it be accomplished and how is it accomplished? Thanks in advance. The standard answer when this feature is needed is Asterisk is a PBX not a key system and the closest thing you will find is the FOP http://www.asternic.org/ if your users can stand running a separate app on a PC. You can use it in conjunction with call parking to sort of simulate what you are looking for. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Does X100P clone provide Timer?
On Wed, 2005-03-23 at 14:30 -0500, Dan Morin wrote: Does anyone know if the X100P clone cards provide the timer needed to run MOH and the Conferencing service? Yes, it will work fine. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Group Ring after Timeout
On Fri, 2005-03-18 at 16:59 +0100, Reuben Grech wrote: Dear All, I am listening to blips during conversations when I have an incoming call from an X100P card. This does not happen on all conversations. Any clues? :) Turn off call waiting in zapata.conf callwaiting=no -Seth Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] MozPhone
On Wed, 2005-03-02 at 14:38 -0500, skamp wrote: doing that gives me the same errors as trying to install from the web site Installation Error -214 firefox could not download the file at file:///home/username/mozphone.xpi Because : Cancelled whats wrong In Linux you need to be running firefox as root to install it. Make sure you close all open firefox windows before to try to run as root. If you have a firefox window open that was started from userland it will attach to that process which would not have root privileges. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Attended xfer
On Wed, 2005-02-16 at 14:11 +, Mark Benson wrote: As for the alternative to attended xfer, parking calls, I'm guessing this is just a case of blind xfering calls to a parking extension? That is correct... if 800 is your parking extension then you dial #800, you will hear what extension they were parked on (i.e. 801, 802, etc...) and they will hear MOH. Then call your party and tell them what extension the call is parked on. This is all much easier if you use phones with programmable buttons. Just set up Park, Park 1, Park 2, etc... buttons on all the handsets and nobody has to remember anything. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Festival patch
They are at the bottom of that post... or, make it easy on yourself and just get them from me: http://sremington.zapto.org/weblog/2004-07-04_14.52.21.html -Seth On Mon, 2005-02-07 at 07:48 -0600, Isaac McDonald wrote: Anyone know where I can get the patch described here: http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html I am getting compile errors when trying to compile speechtools... Any help would be greatly appreciated, Isaac -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Why use 'Answer'?
On Wed, 2004-12-22 at 08:41, John Hill wrote: Question: Do you need to answer to detect a fax? Yes. You need to answer the line so the calling fax will start sending the fax tones and * can detect them. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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 does t mean in a CDR entry?
On Mon, 2004-12-20 at 13:45, Me wrote: What does t mean in a CDR entry? The 't' probably means that the call ended up in the timeout extension. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Broadvoice Patch
On Tue, 2004-12-21 at 21:04, Nihal wrote: I'm trying to get the Broadvoice patch to work, and hoped that someone else might be able to shed some light for me. The patch appeared to go all fine, except when I turn on sip debug I see my asterisk server REGISTERing to broadvoice every 30seconds or so. Sounds like it's working correctly. Broadvoice set's their registration timeout pretty low AFAIK. My * boxes re-register every 20-30 seconds as well. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Broadvoice Patch Applied to CVS
Just in case anybody missed it, the Broadvoice patch has been applied to CVS HEAD: = Sat, 11 Dec 2004 23:33:48 -0600 (CST) Modified Files: chan_sip.c Log Message: Merge SIP authentication reuse patch (bug #2917) aka The Broadvoice Patch with modifications = Olle also has an updated patch for CVS stable (1.03) at http://edvina.net/broadvoice/patch.shtml -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Apply Patch for Broadvoice.
On Fri, 2004-12-10 at 20:02, Dealer Backup Admin wrote: Received errors as follows. snip Are you using version 1.0 or CVS HEAD? The patch will probably only apply cleanly on the 1.0 branch. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Apply Patch for Broadvoice.
On Fri, 2004-12-10 at 14:45, Dealer Backup Info wrote: Hello, I am using Broadvoice for my outgoing calls with my Asterisk box. Broadvoice is requiring my to apply a patch to my Asterisk. Instructions at the following link. http://www.broadvoice.com/support_install_asterisk.html Step 1 is what I need help with, not sure on how to apply patch. I have the rest of the instructions figured out. Copy the file into your /usr/src/asterisk directory. From /usr/src/asterisk run patch channels/chan_sip.c broadvoicesip.txt. As a side note: I have never been able to get outgoing calls to work with the host=proxy.XXX.broadvoice.com setting as they describe. I've always had to set it to host=sip.broadvoice.com. I'd be interested in how it works for you. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Another Unable to create channel of type 'Zap' (cause 0) error
On Mon, 2004-12-06 at 17:40, Alan Ingleby wrote: exten = 1000,2,Dial(Zap/1:555-1234,20,tr) Change this to exten = 1000,2,Dial(Zap/1/5551234,20,tr) Oh, and what extension do I use to reference an incoming call on my FXO port? exten = 1 ?? You want the s extension. http://www.voip-info.org/wiki-Asterisk+s+extension -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice - bad quality, dtfm mode
On Mon, 2004-12-06 at 12:34, Bartosz Wegrzyn - asterisk wrote: Now it is back, but the voice quality is terribe and the DTMF is not working.(Is the inbound mode the correct one?) I'm not sure about your quality problems since I have not had the same issue, but in regards to your DTMF question... it should be set to dtmfmode=inband for Broadvoice. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] IAXy ADSI ?
On Thu, 2004-12-02 at 20:38, Gary wrote: Now the question is can the IAXy be used to successfully drive and ADSI phone ? ADSI is currently only supported in Zap channels so the answer is, unfortunately, no. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] IAXy ADSI ?
On Fri, 2004-12-03 at 10:09, Andrew Kohlsmith wrote: On December 3, 2004 09:56 am, Seth Remington wrote: On Thu, 2004-12-02 at 20:38, Gary wrote: Now the question is can the IAXy be used to successfully drive and ADSI phone ? ADSI is currently only supported in Zap channels so the answer is, unfortunately, no. Are you sure? I thought I have seen posts saying ADSI over SIP and IAX were done (tones is tones, so long as the codec doesn't screw it up, it should get to the destination phone). Perhaps I should have prefaced my post with AFAIK ;) You may be right and I might have missed those feature enhancements (although I do subscribe to the CVS mailing and haven't seen anything ADSI come through in a while). It's always been theoretically possible but my understanding was only fully implemented in the Zap channel. I based this assumption on two things, personal experience and Mark's old mailing list posts. I tried to do ADSI over SIP about 3-4 months ago and did fairly extensive research on it at the time and the conclusion I came to was Zap only for now. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Patching asterisk for spandsp
On Mon, 2004-11-22 at 17:03, Peter Svensson wrote: Shouldn't the echo canceler cut out when it detects the 2.1kHz guard tone? The comments in ecdis.h and zaptel.c seens to indicate that. Yes it does. I see CLI output telling me so every time I receive a fax. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Spandsp and Asterisk
On Tue, 2004-11-23 at 09:00, Eric Hall wrote: Does anyone have an update patch file to get Spandsp installed? I'm running asterisk CVS-HEAD-11/19/04-21:53:37 on redhat 9.0 I installed spandsp-0.0.2 when runnig the patch I get patching file Makefile Hunk #1 FAILED at 41. Hunk #2 FAILED at 69. 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej Make sure you are trying to patch the Makefile in the apps directory, not the top-level Makefile. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice update
On Fri, 2004-11-19 at 18:01, Kevin wrote: I have applied the revised patch. After working through the steps to follow (I think there may be another mistake in the steps) I get a busy when calling out: -- Got SIP response 404 Not Found back from 147.135.0.128 -- SIP/sip.broadvoice.com-39f6 is circuit-busy When I change the host=proxy.dca.broadvoice.com (which is my closest) to sip.broadvoice.com it works. Is there a typo in the instructions? I am getting the same error on outbound calls although incoming calls work with the new patch and configuration. Making the change you describe fixed things for me. I can't imaging that it was a typo though because the whole section about choosing a proxy is specifically for setting the host= section in the config file. Any ideas about the above error? I'm very nervous about my current BV setup since it's not the official configuration. I'm just waiting for it to break. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Patching asterisk for spandsp
On Mon, 2004-11-22 at 14:38, Eric Rees wrote: When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED I haven't updated this in a while but you can try it and see if it works... http://sremington.zapto.org/downloads/asterisk/spandsp/Makefile.patch -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Can some bady help me ???
Do a ./configure Asterisk doesn't have a configure script. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] sending faxes with asterisk in between
On Tue, 2004-11-16 at 19:47, Joseph wrote: Is it possible to send a fax with asterisk in between? When I try to send a fax I've got some COMREC error I think it is some kind of communication error, it fail to negotiate protocol I think. I'm trying to send a fax over standard line (not over IP). The setup should be identical to any other analog device. Just plug it into a properly configured FXS port. http://www.voip-info.org/wiki-Asterisk+and+faxes http://www.voip-info.org/wiki-Asterisk+fax -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SendText
On Sun, 2004-11-14 at 13:02, Alessandro Gatti wrote: Hello, I was trying to use SendText to send a message to an extension, but it seems as if the message is being sent to the caller instead of the callee... e.g.: exten = 123, 1, SendText(hello world) Does anyone have any suggestion on how to override the behavior? Many thanks, Alex Well, like most applications it performs on the channel that called it. That means the caller in the terms you used. So when you dial extension 123 in your example the SendText() application will send hello world to you since you are the channel that executed it. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Voicemail shorter then (ex) 2sec - don't accept
On Sun, 2004-11-14 at 13:02, Joseph wrote: In which configuration file I can specify that I don't want to accept messages for example shorter then 2sec. ? I've looked in voicemail.conf but I couldn't find any setting that will support this option. In most cases message shorter then 2 or 3sec will not contain any message and I don't want system to record them and sending an email to me. You were looking in the right config file. The parameter is called maxmessage. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Voicemail shorter then (ex) 2sec - don't accept
You were looking in the right config file. The parameter is called maxmessage. I just checked and I think this is not the one. maxmessage is to limit the message to the amount of time you specify in seconds. What I was looking for was to discard all the messages that are 3sec. or shorter. You are correct. I had it straight in my head but wrote the email wrong :) The parameter I originally meant was minmessage which should set the minimum length of the voicemail message in seconds. A quick source code check confirms that any voicemail less than minmessage will get deleted automatically. Sorry about the confusion. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice number always busy
On Mon, 2004-11-15 at 15:01, Jerry Geis wrote: I am still getting a Busy message when I call in to my broadvoice number. Is anyone else still getting that or found a fix to it? I can call out all I want no problem. This seemed to start happening after the patch was applied. I've applied the patch on two separate * boxes (work and home) and both incoming and outgoing have been working fine. I'm using proxy.dca.broadvoice.com if that makes any difference to you. Does sip show registry show asterisk as registered with Broadvoice? -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ADSI questions for a 390 ADSI Phone
On Mon, 2004-11-15 at 15:40, Rob Emanuele wrote: I got a 390 Adsi phone (unlocked) hooked to my wct400. It seems too work pretty well. I'm having two little problems with it. 1) The idle title screen will not show up unless I manually press service and select Asterix PBX. What seems odd is that if I do not manually select it it follows the script correctly if the phone goes off hook. Summing up, my title screen is the Time and date (when it should say Asterisk PBX and have a soft key for voice mail) and going off hook shows Asterisk PBX and 3 soft keys programmed. The quick short answer is load your script into *both* the first and second slots. The first slot is triggered when the phone detects call activity. The second slot automatically gets loaded after being idle for 6-7 seconds. It sounds nice in theory but in practice it leads to mass confusion (in my mind anyway) and strange behavior when I have different programs in the first two slots. I sure some of that is my fault ;) I have something like this in my dial plan to load the ADSI programs: exten = 300,1,ADSIProg(saberlogic1.adsi) exten = 300,2,ADSIProg(saberlogic2.adsi) exten = 300,3,Hangup The only difference between the two is the FDN and SECURITY parameters at the top of the script. 2) Comedian mail always asks to download it self when it it already on the phone. Comedian Mail tries to do a FDM download (download a new script) to the phone first. If you don't change the FDN and security code to point to an unused slot you are either going to overwrite another script or get a download refused error. If you are running an older CVS version then you have to make a small source code change in app_voicemail.c and recompile. I wrote a small patch that lets you define this in the voicemail.conf file that went into CVS a little before the 1.0 release so if you are running the stable branch or CVS HEAD add the following to the [general] section in voicemail.conf: adsifdn=7BC6450C adsisec=9B609430 That would load Comedian Mail into the third slot on the phone. You could put it into the fourth slot if you wish by setting the appropriate FDN and security code. In other news, is there documentation for the .adsi scripts anywhere? None other than the asterisk.adsi sample and the source code that I am aware of. Browsing through res_adsi.c and app_adsiprog.c is very enlightening. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice number always busy
On Mon, 2004-11-15 at 16:14, Jerry Geis wrote: I does show registered. HostUsername Refresh State sip.broadvoice.com:5060 XXX1184 Registered This is probably not related but that refresh rate looks funny. Mine has been 15 since applying the patch. Do you see the debugging registration messages that were accidentally left in the patch on the CLI? (Assuming you didn't take them out). -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Voicemail shorter then (ex) 2sec - don't accept
On Mon, 2004-11-15 at 16:06, Joseph wrote: [snip] You are correct. I had it straight in my head but wrote the email wrong :) The parameter I originally meant was minmessage which should set the minimum length of the voicemail message in seconds. A quick source code check confirms that any voicemail less than minmessage will get deleted automatically. Sorry about the confusion. -Seth So I think this minmessage is available in ver. 1.0.1 and up. As I have ver. 0.9 and it is not there. According to the CVS log it was added on 03/18/2004. quote CVS log revision 1.21 date: 2004/03/18 22:49:06; author: malcolmd; state: Exp; lines: +2 -0 Bug # 1247: Add patch to specify minimum vmail length and discard that which does not meet the minimum /quote CVS log Grep the source you have for minmessage. If you find references to it you should be fine. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Unable to create channel of type Zap!
On Sun, 2004-11-07 at 13:37, Adnan Ahmed wrote:
hi,
I am using TDM400 with FXS and FXO modules,everytime i try to make call
of my analog phone it gives following errors:
Executing Dial( , ) in
new stack
app_dial.c:554 dial_exec:Unable to create channel of type Zap Everyone
is busy at this time.
Executing Congestion( , ) in new stack
Spawn Extension (outgoing,6943442, 2) exitd non-zero on Zap/1-1 ;in
which 6943442 is local number and '2' i don't know what is that.
Hungup Zap/1-1
I am new in this group and also asterisk too so don't bother with my
questions!
my configs are:
zaptel.conf
fxoks=1
fxsks=4
loadzone=us
defaultzone=us
zapata.conf
[channels]
signalling=fxo_ks
context=outgoing
channel = 1
signalling=fxs_ks
context=incoming
channel = 4
extensions.conf
[incoming]
exten = s,1,Dial,Zap/4
[outgoing]
exten = _NXX,1,Dial/Zap/1/${EXTEN:1}
This should be exten = _NXX,1,Dial(Zap/1/${EXTEN:1})
-Seth
kindly pointout my mistakes/errors and helping me out.
Thanks In Advance .
Adnan Ahmed.
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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Re: [Asterisk-Users] how to get Stable 1.X via CVS
On Tue, 2004-11-09 at 03:53, Wilson Pickett wrote: cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds So this means you would be getting 1.0? Or? It means you'd be getting the stable branch. (the most current stable version) That's good once you know it, but wouldn't this *appear* to get you 1.0? I can see how new users would be confused by this. It isn't that obvious to me, either. -r v1-0 is the best advice. The individual stable sub-releases are also getting CVS tags as well so if you want to get version 1.0.2 it's just -r v1-0-2 You can see all of the available tags by running cvs log filename on a file in the source tree. Towards the top of the output you will see a list of Symbolic Names which are the tags. They are in the form tagname: CVS version. The CVS version is not important for our purposes; they are mostly for CVS internal housekeeping. If all of tagnames get confusing (unfortunately the tag naming standard appears to have changed over time) just stick with v1-0. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 and ADSI Help
On Mon, 2004-11-08 at 17:47, Christopher Dobbs wrote: Does anyone know how to transfer ADSI information over IAX, I have looked at the code, and it apears that this is posible. I think ADSI currently only works with Zap channels. You are correct that it should be possible with any channel type but my understanding is that it's only currently implemented in the Zap channel driver. I don't have paperwork to back that up though :) What are you trying to do? Run an analog ADSI phone through an IAXy or something? -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] UK CID patch and version 1.0 CVS build
On Tue, 2004-11-09 at 09:02, James Botham wrote: Hi there, Not seen this error message before has anybody else, my perfectly working asterisk system (albeit no caller ID) has now stuffed up I was on the CVS head as of about a week ago but installed a UK caller ID patch so that I can get caller ID (no problems appeared to patch perfectly with no errors) after recompiling asterisk libpri and zaptel I now get the following error at the console and then it bombs out Any ideas ??? I wish Mark and company would just incorporate this in to Zaptel its a pain in the A** to us UK users. [pbx_loopback.so]Nov 9 13:54:13 WARNING[3593]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined symbol: pbx_substitute_variables_varshead Nov 9 13:54:13 WARNING[3593]: loader.c:429 load_modules: Loading module pbx_loopback.so failed! James Botham Client Support Consultant Computer Software Group plc www.computersoftware.com Looks like you have mixed CVS versions. Here's how somebody else fixed it... http://lists.digium.com/pipermail/asterisk-users/2004-October/070073.html -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] X100P CLONES again
On Tue, 2004-11-09 at 15:22, [EMAIL PROTECTED] wrote: What model of modems can be use as X100P? I can get a Motorola 62802-52, did anybody ever try it? I must buy some of the clones because in my country nobody sells anything of voip yet. Thank You __ Check the Generic heading: http://www.asterisk.org/index.php?menu=hardware -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Same Extensions in Multiple contexts
On Mon, 2004-11-08 at 15:43, Uma S. Pandey wrote: Hi For a customer, I am trying to setup 3 different companies on one asterisk box, and I need to assign extension 200 in three different companies. I was using different contexts, but was unable to get it to work. So, my basic question is - In Asterisk, Can we have same extension number in different contexts? For example: [Context_company_1] exten = 200,1,,, [context_company_2] Exten =200,1,.. [context_company_3] Exten =200,1,.. Thanks Uma Pandey This is certainly possible and in fact quite common. The actual extension that gets used just depends on which context you drop the incoming / outgoing call into. Maybe if you give us some more specifics / config examples we can help you out more. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Newbie X100P Clone question
On Fri, 2004-11-05 at 15:58, Steve Frank wrote:
I've got an X100P, I'm brand new to Asterisk. I've been able to set up
SIP extensions and have them working, now I've added the X100P in so I
can drop a line in and eventually be my outside world connection.
I've downloaded the zaptel code via CVS, and configured it up pretty
much exactly like
http://www.digium.com/index.php?menu=configuration#X100P by adding items
into my configurations. My current extensions.conf looks like this:
[general]
static=yes
writeprotect=yes
[bogon-calls]
exten = _.,1,Congestion
[default]
exten = _XX,1,Dial,Zap/1/${EXTEN} ; Press any 7 digit number and
try to dial that number through Zap channel 1
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,2,Playback(demo-congrats) ; Plays the demo-congrats file
after answering the line
You've numbered both lines as priority 2. Fix that and you'll be fine.
-Seth
exten = s,3,Hangup
[from-sip]
exten = 3073,1,Dial(SIP/3073,20)
exten = 3073,2,Voicemail(u3073)
exten = 3073,102,Voicemail(b3073)
exten = 3073,103,Hangup
exten = 3087,1,Dial(SIP/3087,20)
exten = 3087,2,Voicemail(u3073)
exten = 3087,102,Voicemail(b3073)
exten = 3087,103,Hangup
exten = 3089,1,Dial(SIP/3089,20)
exten = 3089,2,Voicemail(u3089)
exten = 3089,102,Voicemail(b3089)
exten = 3089,103,Hangup
exten = 3123,1,VoicemailMain(${CALLERIDNUM})
Here's zapata.conf:
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
busydetect=no
callprogress=no
callerid=asreceived
group=1
context=default
channel = 1
An inbound call to the extension doesn't play back the congrats demo
gsm recording. Running asterisk with -gc I get the following upon
dial in:
*CLI -- Starting simple switch on 'Zap/1-1'
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Nov 5 14:54:59 WARNING[1967]: chan_zap.c:5466 ss_thread: CallerID
returned with error on channel 'Zap/1-1'
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
What's up with the exited non-zero on the spawn extension?
Also, whenever starting Asterisk I always get this about 10 seconds
after init:
Nov 5 14:54:45 NOTICE[1958]: pbx_dundi.c:2841 destroy_trans: Peer
'00:50:8b:f3:75:bb' has become UNREACHABLE!
What does that mean?
Thanks very much in advance. This setup is very very interesting when
compared to our current production Interactive Intelligence CIC
system
Steve
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Re: [Asterisk-Users] Dynamic DNS causes problems
On Thu, 2004-11-04 at 02:58, Larry Hendrickson wrote:
Hi all,
This is my first post to this list, so I apologize if it is a newbie
question. I did quite a bit of reading and a number of Google searches
for answers and found people with dynamic DNS problems, but not the
same one.
I just recently set up Asterisk as a pbx system for my home using
Broadvoice. I would first like to say thank-you for an incredibly
effective program (it took me a while to get used to it, but now I am
quite impressed).
The only remaining problem comes from the fact that I have an ADSL
connection at my home and the pppoe changes my IP address every once in
a while. I have set my sip.conf 'host=' command up with a dyndns
hostname and everything works when I start *. The moment the IP address
changes, though, incoming calls continue to work but outgoing calls give
me a maximum retries exceeded error, and my Dial command exits after
about 3 seconds with NOANSWER. I have confirmed that this is caused
by the change in IP address by restarting *, successfully making a call,
rebooting my DSL router to have it get a new IP address, waiting for
dyndns to register the change (and confirming that this is correct), and
then immediately making another call that has this problem. When the IP
address doesn't change, I have confirmed that outgoing calls work after
several days. A 'service asterisk restart' always gets the outgoing
calls working again, but it seems odd that I would have to restart * so
regularly.
I am assuming that either * or the remote computer is either doing the
DNS lookup once and caching the IP address or there is a socket that is
opened and kept open through the IP address change and not reconnected
afterward (shouldn't the socket on both ends figure out that the
connection is no longer good and reconnect?).
As an ugly hack, I am tempted to have a cron job check for changes in
the ip address and restart *. The problem with this (or one of them) is
that I have to somehow make sure that it isn't in the middle of a phone
call when it does this. Is there a more elegant way of doing this?
I'm not sure about more elegant, but...
Have your cron job issue an asterisk -rx 'restart when convenient'
command instead of a hard restart. That will wait until there are no
active channels to restart. Also, issuing a 'sip reload' instead of
restarting * is probably sufficient to re-register with Broadvoice.
-Seth
Am I doing something wrong?
(And although getting a static IP would be the most elegant solution,
that really isn't an option now.)
Thank-you in advance,
Larry
-- sip.conf
[general]
externip=MY_HOSTNAME.dyndns.org
bindaddr = 0.0.0.0
port=5060
localnet=192.168.0.0/255.255.255.0
disallow=all
allow=gsm
allow=slinear
allow=ulaw
allow=alaw
context=incoming
dtmfmode=inband
register = MY_NAME:MY_PASSWD@sip.broadvoice.com
tos=0x18
srvlookup=yes
nat=no
[Broadvoice]
type=peer
username=MY_NAME
fromuser=MY_NAME
secret=MY_PASSWD
host=147.135.8.129
context=sip
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
nat=no
[broadvoice-incoming]
type=peer
dtmfmode=inband
host=147.135.8.128
context=incoming
qualify=yes
canreinvite=no
disallow=all
allow=gsm
allow=slinear
allow=ulaw
allow=alaw
insecure=yes
nat=no
[broadvoice-incoming2]
type=peer
dtmfmode=inband
host=147.135.0.128
context=incoming
qualify=yes
canreinvite=no
disallow=all
allow=gsm
allow=slinear
allow=ulaw
allow=alaw
insecure=yes
nat=no
--- the extension I am calling on
[trunkld]
;
; US long distance context accessed through trunk
;
;Pattern match US long distance calls
exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],20)
exten = _NX,2,Goto(error-${DIALSTATUS},1)
exten = _NX,3,Congestion
exten = _NX,102,Busy
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SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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Re: [Asterisk-Users] Hardware Support
On Thu, 2004-11-04 at 10:10, Mike Shultz wrote: Quick Question that I hope someone can answer. Will Asterisk work with basic PCI FaxModems instead of those expensive cards listed on the hardware page? No. The wcfxo driver only works with a very specific Intel/Ambient chipset. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ADSI phones and the Flash Key
On Fri, 2004-10-29 at 17:09, Martin Keding wrote: I have some Sayson / Aastra 480e ADSI phones. They work great except for one annoying feature. It's not really the phone, it's the ADSI programming feature in Asterisk. I can't figure out how to recreate the factor default flash screen key on the phone. When I create a ADSI script, it loads to the phone no problem. However it wipes out all of the factory default keys. I can't figure out what to put in the script to recreate the flash key. Help anyone? Martin I think you should be able to define a softkey like so... KEY flash IS Flash OR Flash FLASH ENDKEY ...and then use it in one of your subroutines like so... SHOWKEYS flash Hope that helps. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] record all calls
On Sun, 2004-10-31 at 23:53, Altus Syman wrote:
Good day all
I want to record all call on my zapvpbinternal channels.
I had a look on the net and and found astGUIclient,I want something easy
and simple that will save it in date/user files.
Please advice
Thanks Altus
If you want to record *all* calls as you say just add a call to the
Monitor() application in you dial plan before you call Dial(). I'm sure
you can get creative with the basename parameter to categorize the calls
the way you want.
exten = _8.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _8.,2,Monitor(wav,${CALLFILENAME},m)
exten = _8.,3,Dial(ZAP/g1/${EXTEN:1})
exten = _8.,4,Congestion
exten = _8.,104,Congestion
http://www.voip-info.org/wiki-Asterisk+cmd+monitor
-Seth
--
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SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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Re: [Asterisk-Users] Can bad person with SIPp attack Asterisk ?
On Fri, 2004-10-29 at 05:20, Robert Rozman wrote: Any more info how to configure Asterisk to limit the number of calls concurrently ? This is done with app_groupcount and the SetGroup, CheckGroup, and GetGroupCount applications. More info here -- http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Re: question about asterisk
On Fri, 2004-10-29 at 09:57, Olger Merlos Valverde wrote: Ok, only one question :), this card, (X100P) looks like one modem analog :) it's the same?? or have some diferents... Very thanks... :) It is an analog modem with a very specific Intel/Ambient chipset that the wcfxo driver supports. If you are so inclined you could Google your way to a lot more information about it. The general list consensus though is to go with the X100P for two very good reasons... 1. You are supporting the project 2. You get technical support from Digium with your purchase. -Seth at minimum you will need one card: Wildcard X100P ... [IPphone-Office1] | SIP/LAN | [Aterisk-Office1] | IAX/WAN | [Asterisk-Office2]---[Wildcard-X100P]---analog---[PBX-Office2]--PSTN | SIP/LAN | [IPphone-Office2] SIP IP phones are configured in /etc/asterisk/sip.conf Wildcard X100P card is configured in /etc/zaptel.conf and /etc/asterisk/zapata.conf IAX link between two Asterisk servers is configured in /etc/asterisk/iax.conf Also relevant: /etc/asterisk/extensions.conf for dialplan use all of the above (sip.conf, iax.conf, etc etc) as keywords for search at http://www.voip-info.org for details. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 channels
On Wed, 2004-10-27 at 12:10, Paulo Adriano wrote: What is the command to see the zap channels registered. Im getting an error when trying to access my outgoing line. No channel type registered for Zap Drivers are loaded but where do I register this so called zap channels ? Regards The command you are looking for is zap show channels. If * complains that no such command exists then you installed zaptel after you installed Asterisk. Recompile Asterisk and you should be fine. -Seth __ -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Nightmare on disconnecting Zap and SIP channel
On Thu, 2004-10-28 at 19:31, Jeremy Rusnak wrote: Hi, Shouldn't it be busydetect = yes ? I'm not sure if busydetect = 1 will work. Actually yes, true, y, t, 1, and on are all synonyms in * thanks to the ast_true() API function. Conversely... no, false, n, f, 0, and off all mean the opposite according to ast_false(). -Seth Also, callprogress seems to be quite buggy for lots of people, I would try turning that off. Jeremy On Thu, 28 Oct 2004 15:39:58 +0800, Asterisk Mania [EMAIL PROTECTED] wrote: my zapata.conf busydetect=1 busycount=7 callprogress=yes -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Transfer caller
On Wed, 2004-10-27 at 16:19, Me wrote: Is it possible to transfer a caller to another internal extension with a plain analog phone attached to an ATA? Sure. Just make sure you have the 't' and/or 'T' option enabled in your Dial command. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Error starting Asterisk.
On Mon, 2004-10-25 at 15:16, Oscar Bults wrote: The warning is gone, but asterisk still exits with the same error. Thanks for taking the effort though ;-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: Monday, October 25, 2004 8:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Error starting Asterisk. On Mon, 25 Oct 2004 20:32:51 +0200, Oscar Bults [EMAIL PROTECTED] wrote: When I issue the command asterisk -c I can see only a warning for file chan_oss.c: I don't work right with non-full duplex sounds cards try noload = chan_oss.so in /etc/asterisk/modules.conf and see if it works without OSS. rgds benjk Start up Asterisk with a 'g' option so it will do a core dump when it crashes. (i.e. asterisk -gc) Then you can load the core dump into gdb and get a backtrace to see exactly where it is failing. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] DUNDi in stable? (New subject)
On Wed, 2004-10-20 at 10:58, Deon Rodden wrote: That's good to know. But, not to sound dumb, I'm not a heavy CVS user, how do I get the latest stable? As of now. The way I'm used to doing it is: export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login cvs checkout zaptel libpri asterisk But that doesn't tell me if that's head or stable. The instructions say: cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds For stable. But my understanding is that will give me version 1.0; no bug fixes since the release of 1.0. I want the latest w/ bug fixes but no new features. My voicemail right now is not rigged for database support and such, just the standard voicemail.conf; So if I go to the latest, I don't want to be forced to retrofit my current voicemail setup. It will do exactly what you want it to. Because v1-0 is a *branch* tag CVS handles it a little differently from a regular tag. By checking out the branch you are now rooted on the stable 1.0 branch but you will still get the latest versions committed to that branch (read: you will get all the bug fixes). Runnning cvs update from your sandbox will contine to bring down patches applies to the 1.0 branch. BTW... if you want to convert your existing cvs HEAD over to the stable branch you can run update -r v1-0 instead of checkout. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Extensions.conf?
On Mon, 2004-10-11 at 13:57, Jeff Owen wrote: Im in need of a pointer. If I want to have based on a caller id, allow incoming calls to be prompted for a pin then allow the caller to call back out another available line to the number dialed after the pin was entered, how would I accomplish that in *? Basically I cant call long distance from work and want to call home, enter a pin and then call wherever I want. I currently have a PSTN interface and a BV connection as well as IConnectHere. You can use a combination of the anti ex-girlfriend feature and DISA like so... exten = s/XX,1,Answer exten = s/XX,2,DigitTimeout,5 exten = s/XX,3,ResponseTimeout,10 exten = s/XX,4,Authenticate(X) exten = s/XX,5,DISA,no-password|mycontext Replace the X's with the appropriate CID number and PIN and DISA will give you a dial tone from within the context that you pass to it. Just make sure you secure it enough that you feel comfortable against unwanted access. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] New Mailbox
On Mon, 2004-10-11 at 23:43, Joseph wrote: When I create a new mailbox number in /var/spool/asterisk/voicemail/default Do I have to copy any message files in there? I've noticed that * originally crated a sample mailbox number 1234 and there are two files: busy.gsm and unavail.gsm Are these files being plaid when with ex. Voicemail(u1234) command? Do I copy these two files to a new mailbox when I create one? You should never have to do anything other then set up your voicemail boxes in your config files correctly. Asterisk will automatically create the proper directory the first time a voicemail is left. Allison will walk you through recording busy and unavailable messages once you have logged into your mailbox. Just follow the prompts. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Setting time on ADSI phones from Asterisk
On Mon, 2004-09-20 at 15:14, Brian McSpadden wrote: There is some searching around to be done on ADSI documents, as others in this thead have pointed out... In answer to the time question, I am told that it is not possible to display the time on some of the ADSI phones, such as the Sayson 390's. I had a customer want it a few weeks ago, and when I called my distributor about it, they called Sayson and were told it was not possible...seems kind of strange that my $80 screen phone can't do something that a $10 phone from Wal Mart can. Brian I have a Sayson/Aastra 390 sitting right in front of me and it is telling me the date and time is currently 'Sep 20 3:29pm'. It worked out of the box based on the callerID time. It is on the topmost line of the OSD that isn't available to ADSI programs... must be in the firmware. -Seth On Sun, 19 Sep 2004 20:41:20 -0400, Dennis Cartier [EMAIL PROTECTED] wrote: Hi, Would anyone know of a way to set the time automatically on an ADSI capable phone from *? The phone in question is a Aastra 480e. While I am at it, does anyone have any helpful docs on the ADSI script programming? I have managed to do basic functions by modifying the asterisk.adsi file using stuff gleaned from the app_adsiprog.c file, but docs would be really helpful at this point. Tia, Dennis -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] festival
On Mon, 2004-09-13 at 15:31, Rich Allen wrote: iH have trouble getting the festival command to work. when i dial extension i have set up i get Sep 13 11:25:33 WARNING[344080]: app_festival.c:440 festival_exec: Festival returned ER festival does work correctly when i use it from the unix command line. my festival.conf file is set up per the wiki Did you apply the patch in /usr/src/asterisk/contrib? The /usr/src/asterisk/contrib/README.festival document might have some info to help you. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Spandsp - opencall.org offline
On Sun, 2004-08-22 at 07:35, Roland Zagler wrote: Please can someone send me the .tar.gz file of spandsp, the site is offline and i didn't find it anywhere! http://sremington.zapto.org/downloads/asterisk/spandsp/ until the opencall.org DNS servers are back up. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] spandsp
On Wed, 2004-08-18 at 23:20, Adam Goryachev wrote: On Thu, 2004-08-19 at 07:01, Seth Remington wrote: On Wed, 2004-08-18 at 14:45, David Filion wrote: Does anyone know of an alternate source for spandsp? opencall.org is down and all the links returned by Google just point to the dead site. Thanks David Filion I threw a copy up here for you: http://sremington.zapto.org/downloads/asterisk/spandsp/ It would seem the nameservers (both) for the domain are down. Is this the latest versions before the site went down?? As far as I know it is the latest. I grabbed it from opencall.org about a month ago. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] spandsp
On Wed, 2004-08-18 at 14:45, David Filion wrote: Does anyone know of an alternate source for spandsp? opencall.org is down and all the links returned by Google just point to the dead site. Thanks David Filion I threw a copy up here for you: http://sremington.zapto.org/downloads/asterisk/spandsp/ -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Dialplan problem - incoming calls get MOH, not ringing.
On Tue, 2004-08-17 at 15:44, Patrick Lidstone (Personal e-mail) wrote:
Chaps,
I recently added an incoming VOIP account to my asterisk box. When the
PSTN number associated with this account is dialled, the call rings once
and then asterisk starts playing music on hold, even though all the
extensions continue to ring. Variations of answer() and ringing() don't
seem to help. I'm sure I'm missing something spectacularly obvious, but
the wiki and googling the mailing list haven't shed any light. FWIW, my
ISDN-2e based incoming lines work just fine using similar dialplans.
Here is the excerpt from extensions.conf:
[my-sip-provider]
exten = 8441,1,answer
exten = 8441,2,Ringing
exten = 8441,3,SetCallerId( 30${CALLERIDNUM})
exten = 8441,4,SetCIDName(SIP 0${CALLERIDNUM})
exten = 8441,5,Dial(ZAP/2SIP/2010SIP/2009SIP/2011,120,trm)
exten = 8441,6,Voicemail,u1001
exten = 8441,106,Voicemail,b1001
You have the 'm' option enabled in your Dial() command. That will play
MOH to the calling party until the called channel answers.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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Re: [Asterisk-Users] Problem installing Software Fax SpanDSP support into Asterisk
On Thu, 2004-08-12 at 12:08, Jon Bebeau wrote: I'm trying to install the SPANDSP software into Asterisk to support incoming (mainly) Fax. I'm following the info in http://www.voip-info.org/wiki-Asterisk+Fax. I downloaded and installed the spandsp software from ftp://ftp.opencall.org/pub/spandsp/ and followed the directions in several documents listed on the on the Tiki page. I get down to patch Makefile.patch that fails with Hunk #1 FAILED at 35. and Hunk #2 FAILED at 68. The Makefile.rej is included below. Not being a Linux wiz-kid, I'm struggling to figure out what when wrong and how to fix it. The problem seems to be related to two missing programs app_todd and app_sql_odbc. Some searching around with Google, I find these things seem to have existed at one time (the todd anyway). It seems that the most recent documents: http://scottstuff.net/scott/archives/000152.html Dated March 28, 2004 and the earlier lists documents: http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html (cache), http://lists.digium.com/pipermail/asterisk-users/2004-March/041408.html (cache) and http://lists.digium.com/pipermail/asterisk-users/2004-March/041433.html (cache) (Thanks to Scott Laird) They don't say they have this problem. I'm thinking Asterisk has changed enough in recent times to require some other procedure. Any guidance would help. Environment: Redhat 9, Linux 2.4.20-31.9smp and Asterisk CVS-HEAD-08/02/04-15:03:06 on a Tyan single P4 3.06Ghz with Hyperthreading enabled. Is this Fax a viable solution? Any better/other suggestions for faxing? You are partly correct. The Makefile has changed since that patch was made and the patch program can't quite figure out how to apply it so it gives you the hunk FAILED errors. Try this patch instead: http://sremington.zapto.org/downloads/asterisk/spandsp/Makefile.patch -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Problem installing Software Fax SpanDSPsupport into Asterisk
On Thu, 2004-08-12 at 14:42, Jon Bebeau wrote: Thanks Seth. That fixed the Makefile problem and got me to the next problem: Compile errors in app_rxfax.c - below.. I'm thinking there are updated source programs (app_rxfax.c and app_txfax.c and friends) to go along with the updated Makefile? Jon app_rxfax.c compile log below-- gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c In file included from app_rxfax.c:14: ../include/asterisk/lock.h: In function `ast_mutex_init': ../include/asterisk/lock.h:300: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) ../include/asterisk/lock.h:300: (Each undeclared identifier is reported only once ../include/asterisk/lock.h:300: for each function it appears in.) make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Make sure to run make install from the top level Asterisk directory. My guess is that you are running it from the apps directory. The top level Makefile defines _GNU_SOURCE and the lack of it causes the PTHREAD_MUTEX_RECURSIVE error. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] called and callers buttons on bt100
On Mon, 2004-08-09 at 17:41, Jason Kawakami wrote: is there something that needs to be set up to make the 'called' and 'callers' buttons work on this phone? all i get is the backlight to switch on and off. Jason Kawakami Lift up the handset and try it. It doesn't work when it is on-hook. Just another quirk of the GS to add to the list. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 mirror
On Tue, 2004-08-10 at 09:19, [EMAIL PROTECTED] wrote: On Tue, 2004-08-10 at 07:35, [EMAIL PROTECTED] wrote: Hello! Is there a asterisk mirror? ftp://ftp.asterisk.org/pub/telephony/asterisk/ is slwww Use CVS, it is mirrored. thanks! But how stable is the cvs checkout? You can get the RC1 from cvs... just use the command cvs co -r v1_0_rc_1. Or you can use cvs co -r v1-0_stable to get the old stable version (not recommended). -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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]Google and the Asterisk list
On Tue, 2004-08-10 at 10:05, Michael Welter wrote: Since I joined the list in 2003, there has been an order of magnitude increase in functionality in Asterisk. Kudos to Mark and the developers. I'm finding that when I Google the list, posts that are more than a few months old are no longer relevant to me. Without using the Google API, is there a way to have Google sort responses in date order? Or, would there be a way to maintain a rolling archive--new posts would enter the archive and post older than, say, six months would drop out? Or maybe an archive containing n posts--as each new post arrived the oldest post would drop off? Thoughts? If you use their advanced search page (http://www.google.com/advanced_search) you can restrict your results to the past three, six, or twelve months. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] AbsoluteTimeout Inside A Macro
On Tue, 2004-08-10 at 13:11, Chris Shaw wrote: For one thing it's 't' not 'T', just like invalid is 'i' not 'I' -Chris Christopher L. Wade wrote: Hi all, Is it just me and not reading the docs right, or has anybody else had problems with the AbsoluteTimeout application and the 'T' extension when used inside a macro? Christopher is asking about AbsoluteTimeout so 'T' is correct. t (lowercase) is the timeout extension in a context (set with ResponseTimeout for instance). T (capital) is Timeout on AbsoluteTimeout. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ChangeMonitor syntax
On Mon, 2004-08-09 at 11:24, [EMAIL PROTECTED] wrote: I'm trying to use the ChangeMonitor command on the asterisk manager API, but I can't find the syntax anywhere. Asterisk only tells me: Action: ChangeMonitor But I don't know the parameters. Can anybody help me? It takes two parameters: Channel and File. Channel is the channel that you are monitoring and want to change the filename being recorded to and File is the new filename. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ChangeMonitor syntax
On Mon, 2004-08-09 at 12:57, Holger Schurig wrote: try help application changemonitor in the Asterisk CLI I'm sure you meant show application changemonitor. That will show the dialplan application ChangeMonitor but not the Manager API ChangeMonitor command. The show manager command ChangeMonitor gives the very unhelpful help that was mentioned in the original post. I had to grep the source in order to find the proper parameters. In fact none of the show manager command ... CLI commands give any of the required parameters. Some patches would probably be in order. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] voicemail attachment setup per user
On Thu, 2004-08-05 at 16:34, Gary Carr wrote: Is it possible to set the attach= setting on a per user or per context basis? We want to give our users the choice of no email notfiication, email notification with no attachment, or notification with attachment. Quoted from http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf: E-mail attachments: If you want some users, but not all, in VoiceMail2 to receive the VM message as an attached audio file in the e-mail notification, add an option to the user configuration in the correct context of voicemail.conf. Example: 1234 = 1234,Sample User,[EMAIL PROTECTED],,attach=yes -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] VoicemailMain Issues
On Fri, 2004-08-06 at 11:56, Robert Jackson wrote: I have a very bizarre issue for ya'll. Asterisk seems to crash after I hang up on VoicemailMain, but only if the user logs in. I am completely dumbfounded with this. We have been running our production system on asterisk HEAD 7/14/2004 for a few weeks now, and this error only happened when I updated to 8/4/2004. I am calling my voicemail extension via X-Lite, and the error message received on the console when asterisk crashes is simply Killed. Has anyone else seen this issue before? I am just trying to figure out if it is something in my config or if there my be a problem with CVS 8/4/2004. I don't have an answer to your particular problem but in general it sounds like a SEGFAULT or some other similar bug. Try this... 1. Start Asterisk with safe_asterisk 2. Cause asterisk to crash the way you describe. Asterisk will dump a core file into /tmp 3. Enter gdb asterisk /tmp/core. (you need to have gdb installed of course) 4. Enter bt while in gdb (or do a bt full) to see the back trace. You will probably see an Address out of range or similar error in the last function call on the stack. Take note of the function where the error occurred an the parameter that had the out-of-bounds memory address (if applicable). If your C skills are up to snuff to can try and debug it yourself. If not open up a bug on the bug tracker with all of the info you have collected. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] shared voicemail
On Thu, 2004-08-05 at 09:57, Altus Snyman wrote: Good day all I got my voicemail message working,thanks but now,keep in mind I'm using SIP We have,for example 4 people in our admin department.Each user has its own voicemail so that when their extension is dialed directly and not answered it gos to voicemail. But there is also a option to dial 3 for admin with will dial all 4 number in sequence.This I got working 100% but now I want a shared voice mailbox for the admin department.For example: If they call extension 4 it dials user 1 for 10 seconds,then user 2 for 10 seconds.if user 4 does not answer it answers with voicemail saying,this is the admin dep please leave.This message will then be inserted into each 4 of the admin users voice-mailboxes? Can this be done Thanks Altus In voicemail.conf set up your Admin mailbox: 101 = 101,Admin Mailbox,,,delete=1 This will allow you to record your Admin prompt but the delete=1 will auto delete the message from this mailbox. Then in extensions.conf: exten = 2,1,Voicemail(101102103104) Change the extension and priority to what you need but the gist of it is the mail will be left in all of the mailboxes that are concatenated with '' and the prompt from the first one in the list will be used. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] shared voicemail
On Thu, 2004-08-05 at 13:57, Wayne wrote: Seth Remington wrote: In voicemail.conf set up your Admin mailbox: 101 = 101,Admin Mailbox,,,delete=1 This will allow you to record your Admin prompt but the delete=1 will auto delete the message from this mailbox. Then in extensions.conf: exten = 2,1,Voicemail(101102103104) Change the extension and priority to what you need but the gist of it is the mail will be left in all of the mailboxes that are concatenated with '' and the prompt from the first one in the list will be used. -Seth Cool - was after the same kinda thing myself (just hadn't got round to looking yet :)) - anyways... you say that the mail is left in each of the extension mailboxes - is this 1 single copy that everyone can retrieve or 1 unique copy in each mailbox... It is a separate unique copy in each mailbox. The question being - if extension 102 went into voicemail and listened to and deleted a 'group message' does it still exists for extensions 103 and 104 to which they then have to go into their mailbox and delete the message? (hope that makes sense) Wayne With the above configuration it would still exist in the other mailboxes as well. If you want one person deleting it to affect everybody I would assume that you would need to keep the messages in a single shared mailbox (the 101 Admin mailbox in the above example). I'm not sure what your setup is but most channel types support checking multiple mailboxes to send a MWI. So you could alert the user if there was a message waiting in their personal mailbox or the group one. mailbox=101,102,103 in zapata.conf, sip.conf, etc... -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Get MWI from Telco's voicemail
On Wed, 2004-08-04 at 14:21, Scott Petersen wrote: Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco. Maybe I am misunderstanding you but why does this force you to use telco voice mail instead of * voice mail? You can also free that third line up for voice if you use faxdetect. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Auto-attendant with an IP trunk
On Wed, 2004-08-04 at 14:47, [EMAIL PROTECTED] wrote: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have extensions from 2000 - 2010. What happens is that Asterisk jumps out at the first digit '2' and says that nothing found with '2'. It doesn't even read the rest of the digits '000'. You might consider adding a DigitTimeout command at the beginning of your IVR. http://www.voip-info.org/wiki-Asterisk+cmd+DigitTimeout -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Playback doesn't work whith h323
On Wed, 2004-08-04 at 20:29, Jeremy McNamara wrote: M. Willigs wrote: Hi Jeremy My entry in the extensions.conf is like this: exten = 011001,1,Playback(tt-monkey) I didn't asociate the cmd Dial whit this entry, so, I can't answer the line You are not answering the line and that extension looks weird to me. Jeremy McNamara exten = 011001,1,Answer exten = 011001,2,Playback(tt-monkey) -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Emailing phone messages?
On Tue, 2004-08-03 at 11:22, Sean Garland wrote: Where do you set the outgoing mail server for use with asterisks mail system? It uses the command '/usr/sbin/sendmail -t' by default. You can use the mailcmd parameter in voicemail.conf to override that. From the wiki: Mailcmd allows the administrator to override the default mailer command with a defined command. Mailcmd takes a string value set to the desired command line to execute when a user needs to be notified of a voice mail message. http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] App.c
On Mon, 2004-08-02 at 12:34, Steve Hanselman wrote: Delete it and cvs update will retrieve it. cvs update -C app.c would also work. -C retrieves the clean copy from the repository and saves your local changes into another file. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] faxing
On Fri, 2004-07-30 at 03:40, Vladyslav wrote: BTW, compilation of rxfax with latest CVS-2004-07-29 fails. and Makefile.patch (which is on the site) should be modified as well. gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c In file included from app_rxfax.c:14: ../include/asterisk/lock.h: In function `ast_mutex_init': ../include/asterisk/lock.h:300: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) ../include/asterisk/lock.h:300: (Each undeclared identifier is reported only once ../include/asterisk/lock.h:300: for each function it appears in.) make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/install/cvs/2004-07-29/asterisk/apps' make: *** [subdirs] Error 1 You need to make sure that _GNU_SOURCE is defined or else you will get this error. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Aastra 480e phone ADSI config
On Thu, 2004-07-29 at 11:04, Martin Keding wrote: On a side note, I am tring to enhance the ADSI programing in the orignal script. Did your supplier give you any help with additional commands etc. I have not found any docs. So far. There aren't any other commands that Asterisk's implementation of ADSI supports other than the ones that are demonstrated in the asterisk.adsi script. That's not to say that ADSI doesn't have more commands, just that Asterisk does not yet support them. If you take a peek at the code in app_adsiprog.c you will see all the commands that are supported sprinkled through the code (specifically the adsi_process() function). -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 and festival
On Thu, 2004-07-29 at 11:32, Adam Lewis wrote: I setup and extension to test festival and when I dial it I get __SNIP__ -- Executing Answer(SIP/phone4-17ae, ) in new stack -- Executing Festival(SIP/phone4-17ae, mary had a little lamb) in new stack == Parsing '/etc/asterisk/festival.conf': == Parsing '/etc/asterisk/festival.conf': Found telco-pbx*CLI SIOD ERROR: unbound variable : tts_textasterisk Jul 29 10:59:08 WARNING[1015826]: app_festival.c:440 festival_exec: Festival returned ER == Spawn extension (sip, 555, 2) exited non-zero on 'SIP/phone4-17ae' __SNIP__ Did you also apply the /usr/src/asterisk/contrib/festival-1.4.3.diff patch? -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [SPAM] RE: [Asterisk-Users] Aastra 480e phone ADSI config
On Thu, 2004-07-29 at 13:55, Steve Woolley wrote: 1) My phone seemed to either be pre-programmed with the Comedian mail scripts or asterisk did not need to be patched for my phone to have functional Comedian vmail access screens. I have however noticed that a number of the Comedian vmail screens are incomplete (saying things such as Options Menu Not Done. I assume the Comedian vmail ADSI scripts are burned into the phone (by manufacturer) or something similar because I have not been able to find any Comedian scripts within Asterisk source. I wonder if there are more current Comedian vmail ADSI scripts available? The first time you access Comedian Mail it attempts to do a FDM download to the phone. The reason you can't find any Comedian Mail *scripts* is that there aren't any. It is using lower level C functions to do the ADSI data transmission instead of compiling a text file script like asterisk.adsi. Grep the app_voicemail.c file for adsi and you will see all of the ADSI code. There are two modes of ADSI. Feature download management (FDM) mode allows data (display text and softkey setups) to be downloaded and stored in the ADSI telephone. This data remains active on the phone even after the call that downloaded them has completed. Server display control (SDC) mode is used when there is a continuous connection between a generating device (i.e. asterisk, or a bank app that displays back account info on your ADSI phone). The ADSIProg application is an example of FDM mode. Comedian Mail is sort of a combination of the two. It downloads and stores some stuff (like softkeys) the first time you connect, but it is also SDC in the way it transmits the number of messages you have and the name and callerid of each one. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] 2 cards
On Wed, 2004-07-28 at 03:07, Altus Snyman wrote:
What about outgoing
How do I tell it all sales,sip 100+, to go out threw vpb card's channel
and all admin,sip 200+ to go threw zaptel?
Thanks for the help so far
You also do this through contexts. In your sip.conf you assign a context
to each phone like so:
[sip100]
type=friend
username=sip100
secret=XXX
callerid=Buckaroo Bonzai 100
host=dynamic
context=sales--- /* right here */
[EMAIL PROTECTED]
Then in your extentions.conf you would have:
[sales]
ignorepat = 9
exten = _91NXXNXX,1,Dial(Zap/1/${EXTEN:1}) /* use the zap */
exten = _91NXXNXX,2,Congestion
The above example would allow the sales people to dial 9 to dial a long
distance number on the Zaptel card. The context you use for admin would
explicitly use the vpb card instead.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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Re: [Asterisk-Users] Best Linux for Asterisk
On Wed, 2004-07-28 at 09:13, Eric Kirkland wrote: I'm having difficulty compiling the TTS stuff. You aren't very specific about the problems you are having compiling Festival but on the off chance that your problems were the same ones I had you might want to check out this: http://sremington.zapto.org/weblog/2004-07-04_14.52.21.html -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] BugetTone Bug Showstopper,
On Wed, 2004-07-28 at 21:00, James Gardiner wrote: How do I get Asterisk to recognise the # key from the granstream phone for doing transfers? I have tried a few different config with the phone and gone over the wiki on what to try, but no go.. Make sure the Grandstream is configured to send DTMF via SIP INFO instead of in-audio. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] 2 cards
On Tue, 2004-07-27 at 06:07, Altus Snyman wrote: My question is,can I configure each card for different departments,for example,all calls coming in on zaptel will say welcome to sales dep and calls going out from 100+ will go out on the vpb channels.And the same for the zaptel,calls coming in line on zap will say welcome to ADFAG admen dep and calls made by admin(sip 200+) will go out on zap channels? You can assign each Zap channel to a different context in zapata.conf. Once you have them separated into different contexts you have each line do completely different things in your dial plan on incoming calls. For example... In zapata.conf: ... context=sales channel=1 ... context=admin channel=2 ... Then in extensions.conf: ... [sales] exten = s,1,Answer exten = s,2,Background(welcome-sales) ... [admin] exten = s,1,Answer exten = s,2,Background(welcome-admin) ... -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 variables
On Tue, 2004-07-27 at 09:40, Bastian Schern wrote:
Hi *,
I have problems with the variables in the extensions.conf file.
--- snip ---
[global]
JOHN=SIP/17
[incoming]
exten = s,1,Answer()
exten = s,2,Playback(demo-enterkeywords)
exten = s,3,Background(demo-congrats)
exten = 1,1,Dial(SIP/17)
exten = 2,1,Dial(${JOHN})
--- snap ---
If I dial 1 it will work fine, but if I dial 2 I will get this Message
in the Asterisk CLI:
Jul 27 17:39:34 WARNING[425999]: app_dial.c:485 dial_exec: Dial argument
takes format (technology1/number1technology2/number2...|optional timeout)
[global] should be [globals]
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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Re: [Asterisk-Users] 2 cards
On Tue, 2004-07-27 at 09:48, Altus Snyman wrote: Ya but the one is zaptel nd one voicetronix so it uses vpb.conf for example sales The vpb.conf file allows you to define contexts for each of the channels just like zapata.conf so there shouldn't be a problem. Just use one context in zapata.conf and a different one in vpb.conf. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 for a large scale implementation
On Tue, 2004-07-27 at 09:55, Harry Schechter wrote: Nicholas Bachmann wrote: Is there a good place to find Asterisk consultants? There is a list here -- http://www.voip-info.org/wiki-Asterisk+consultants -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] VoiceMail Group Broadcasting
This bug report patch will do what you want with the cc and delete options to the mailbox. It was supposedly added to CVS but I have not seen it come through yet. Very confusing. I guess you could apply the patch manually. http://bugs.digium.com/bug_view_page.php?bug_id=0001361 -Seth On Mon, 2004-07-26 at 08:00, Jan Goericke wrote: Yes I did it. You need three things 1) An extension that records the vm in a predefined file. 2) An extension that plays the recorded message 3) An AGI that puts some call files into the outgoing directory. You can find a sample file at http://asterisk.gnuinter.net/files/digium/asterisk-ng/sample.call Hope that helps, J. Goericke On Sat, 24 Jul 2004, Frank wrote: Using the latest code from CVS. Has anyone figured out a way to setup any kind of Group or Broadcasting of Voicemail messages? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
