[Asterisk-Users] External channels getting connected
Greetings, I am having a funny problem where a user is on the phone with someone that dialed in. Another call will come in, the first is put on hold, the second answered. When the call is transferred the two external calls are connected and can speak with each other. As you might imagine that causes some surprise and problems. What is causing this and what can be done about it? Asterisk 1.0.9 T1-PRI Thanks ___ 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] AstTapi - Crashes w/ Windows 2000 - UrgentHelpneeded - May need to hire a developer
Title: Message I have it running on my Windows 2000 mahcine using STI products and don't have much of a problem. I would guess that it might be something on the workstations instead of the AstTAPI. Also, might be a little faster, easier, cheaper to just upgrade your existing workstations? Just a thought. Feel free to contact me off list if you would like. Thanks, Dustin -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul RodanSent: Friday, January 21, 2005 1:17 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] AstTapi - Crashes w/ Windows 2000 - UrgentHelpneeded - May need to hire a developer No word on this post. Please, can anybody help me? Is there a known issue with AstTAPI and Windows 2000? Or AstTAPI w/ Amicus Attorney? If were willing to hire a developer to help us fix AstTAPI, is there a developer out there willing to help us? Cost is of less importance than time right now. We need a developer ASAP. http://sourceforge.net/projects/asttapi/ and/or http://www.omniis.com/asttapi From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul RodanSent: Thursday, January 20, 2005 3:48 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] AstTapi - Crashes w/ Windows 2000 - Urgent Helpneeded - May need to hire a developer Were encountering a problem with AstTAPI crashing on Windows 2000 Workstations. The program were using is called Amicus Attorney, it uses a standard TAPI interface to be able to dial our clients, but on the 2 Windows 2000 workstations weve tried it on it has crashed, no errors or anything. When we select the Asterisk TAPI driver, the whole windows just closes/crashes w/ no apparent reason. Now Ive tested AstTAPI on my laptop, but its running Windows XP Pro w/ SP2 and I used Outlook. Is there a known issue with AstTAPI and Windows 2000? I know Amicus Attorney works with other TAPI drivers/interfaces, because the old system used this little external device hooked into the serial port (looked like an external modem) that allowed Amicus Attorney to be able to place calls out of our old phones/PBX. If there is a known issue, or a developer is willing to help us fix the code, were willing to pay. But time is of the essence. Please let me know! Thanks. ___ 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] Snom200 Firmware: I only see 2.04g
Try this one. Took me a while too. http://www.snom.com/download/share/snom200-2.05c-SIP.bin -Original Message- From: M3 Freak [mailto:[EMAIL PROTECTED] Sent: Monday, May 17, 2004 11:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Snom200 Firmware: I only see 2.04g Hello all, I've noticed several messages about the latest firmware on Snom's site, 2.05b, and today I see that another update is listed, 2.05c. However, when I go to the download page (http://www.snom.com/support_dl_en.php), the latest firmware version available for the Snom200 is 2.04g. Are the newest firmware releases not yet available, or am I doing something stupid? Thanks, Kanwar Systems Aligned Inc. www.systemsaligned.com ___ 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
[Asterisk-Users] Long pause between background and voicemail
Greetings, I have our system setup so that when I am not available my message gives you the option to either press 1 for voicemail or 2 to forward to my cell. The problem is the long pause after the choice has been made and before the vm-intro starts playing. The only thing that comes up in the debug that might apply is the following: Channel.c:981 ast_settimeout: Scheduling timer at 160 sample intervals Is there any way to shorten the length of the pause? Thanks in advance. Dustin Knutten ___ 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] Snom 200 MWI Button
-Original Message- From: Paul Oster [mailto:[EMAIL PROTECTED] Sent: Saturday, February 07, 2004 8:22 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Snom 200 MWI Button\ I'm trying to get the MWI button to work with my Asterisk configuration. The snom is accepting and responding to the Message indications from *, but when I press the MWI button, it is dialing my extension (the one with the voice mail on it). I'm wondering if there is a way to specify what extension to dial to check email in the configuration, either the phone, or * itself. Asterisk Version 1/30/2003 checked out and compiled this evening Snom Version 2.03o (most recent auto-update) Any help would be greatly appreciated. At one point Mark had talked about adding a voicemail= directive in sip.conf on the mailing list at one point, however grepping the code doesn't reveal a feature like that at this time. Anyone have success in getting the MWI button to work on Snoms? If so I would LOVE to hear from you. Paul M. Oster Paul, We have ours working pretty well. If you would like to contact me off site I will try to explain what we have done and maybe we can work together on the Snow 200 issue. Thanks, Dustin ___ 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 GUI Client - New verison 0.9
-Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Thursday, February 05, 2004 4:33 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Asterisk GUI Client - New verison 0.9 Hello, I have made many changes/improvements/bug fixes to the Asterisk GUI client I have written in Perl/TK and have released a third beta version on sourceforge: http://sourceforge.net/projects/astguiclient/ Here are the screen shots of the client application running on Linux and Windows: http://www.freedomphones.net/astguiclient_linux_0.9.gif http://www.freedomphones.net/astguiclient_windows_0.9.gif 0.9 - Third public release - 2004-02-05 The majority of the work in this release it to make it more stable and fix some pretty bad bugs. We created the Asterisk Central Queue System to address the problem with buffer-overflows in the manager interface of Asterisk causing total system deadlocks. We also completed and touched-up many other features that we didn't finish in previous releases. Here is the list of changes: - Several bug fixes - Inclusion of listing for active SIP/Local channels and ability to hang them up - Completely changed the method of conferencing to be more fluid - Added HELP popup screen - Added intrasystem calling funtionality - Updater changed to allow for SIP/Local channels - Recording for conferences is now able to record all audio in and out - Added ability to send DTMF tones within a conference - Changed alert window for updater being down timeout to 20 seconds - Added an option for using the new Asterisk Central Queue System(ACQS) that reduces the risk of deadlocks that occur with buffer-overflows on remote manager interface connections - Included new script to run at boot time and rotate the logs as well as a keepalive script for the new ACQS - Changed non-AGI server-side scripts to allow for a single config file - Detailed activity logging to text file option added - Activity logging added to all non-AGI server applications We have been using the same basic client for the last four months here at my company and it is running well on over 60 machines. Let me know what you think of it, especially the new Asterisk Central Queue System that is included with it. MATT--- ___ 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 Matt, Thanks for posting your utiliy. I would really like to use the utility you have written. Is there any installation help or instructions for win32? Pardon my ignorance. Thanks, Dustin ___ 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] Compiling problems with SuSE
-Original Message- From: Uwe Klein [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Compiling problems with SuSE From: Dustin Knuttgen on Sunday, January 18, 2004 11:47 PM We tried to use SuSE initially and had no luck compiling zaptel on either 8.2 or 9.0. We even had Digium take a look. After working on it for days we finally switched to Red Hat 9. Is there anyone who succeeded in compiling Asterisk with SuSE 8.2 or 9.0? HI Dustin, what kind of error did you get? something like this: pbx.c:581: warning: comparison between signed and unsigned pbx.c: In function `pbx_substitute_variables_temp': pbx.c:765: warning: comparison between signed and unsigned pbx.c:812: warning: comparison between signed and unsigned pbx.c: In function `pbx_builtin_hangup': pbx.c:4017: internal compiler error: Segmentation fault ?? I had problems with SuSE 8.2 and Asterisk from cvs dated ~12July2003 I got it fixed by adding 128MB of memory to the 32MB on this P200 machine. with 300MB of swap it should not have made a difference ( except taking forever ) but it did. G! UK -- Uwe Klein [mailto:[EMAIL PROTECTED] KLEIN MESSGERAETE Habertwedt 1 D-24376 Groedersby b. Kappeln, GERMANY phone: +49 4642 920 123 FAX: +49 4642 920 125 ___ 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 Uwe, I had a problem at the end when it does the depmod -a. We got an error with around ten modules. The only thing I could find related to the errors was something about PPP in the kernel or in the Makefile. Neither of which made any difference. Dustin ___ 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] compiling problems
Franz, We tried to use SuSE initially and had no luck compiling zaptel on either 8.2 or 9.0. We even had Digium take a look. After working on it for days we finally switched to Red Hat 9. Everything compiled without problem. Hope that helps a little. Dustin -Original Message- From: Franz Edler [mailto:[EMAIL PROTECTED] Sent: Sunday, January 18, 2004 3:48 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] compiling problems Hello, I still have problems with compiling Asterisk, and I am still on the first step at zaptel make clean; make install. I assume, that the troubles I have stem from a recent kernel-update I made. I upgraded from k_athlon_2.4.21_99_i586 to k_athlon_2.4.21_166_i586 via YaST Online Update.. Now I learned, that I have to provide also the kernel-sources for compiling zaptel. I have done that, but at the end of make install of zaptel I get the following errors on unresolved symbols: /sbin/depmod -a depmod: *** Unresolved symbols in /lib/modules/2.4.21-166-athlon/misc/wcusb.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-166-athlon/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-166-athlon/misc/ztd-eth.o My experience on development tools is unfortunately very small. What do I have to do to move forward? Any help very appreciated. Franz ___ 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
RE: [Asterisk-Users] Computing horsepower needed
-Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 10, 2003 8:45 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Computing horsepower needed In a test system I can take out half the RAM, slow the CPU clock or run the CPU without the cooling fan and just measure what happens. Yes, stupid do do those things in a system people are depending on. Agreed 100%. If you want to conduct experiments you do so by conducting them, not asking what the minimums are. I dunno, I am kind of in the same boat as Steven on this... if you're gonna experiment then experiment. Don't decide to get into * and the first thing out of your mouth is what's the bare minimum processor+ram you can get to make it work... buy something moderately new (P3, 128M, IDE disk) -- it ain't gonna break the bank, it's gonna be easier to find and likely be far more reliable than that P90 you have in the back room that's been gathering dust for the past 5 years. But... if you place yourself in the position of the newbie, where else could you ask given the documentation that truly doesn't exist (yet). Mark made the comment about a month ago that asterisk is this best kept secret in the world. The flip side of that is jumping on every newbie that comes along and pissing them off enough to leave the list (and the app). For those that have been around here for more then 30 days, you already know that's the nature of this list. If you don't like the questions, delete it and stop cluttering the list. I really have to agree with Rich. I am a newbie. I have been reading the lists and watching the IRC and doing what I can to learn. The general feel that I get is a feeling of intolerance for people trying to learn and understand. Is it intentional? Do the people in the know not want anyone else to be part of a great app? Just wondering. Please try to have some consideration for everyone and follow Rich's suggestion, If you don't like the questions, delete it and stop cluttering the list. I will continue to and use and learn because I think it is a great app. Just my observations. Thanks, Dustin Knuttgen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: FXO cards
Maybe even just 4 cards. I would like to know where it states that as well. I have been reading a lot and thought I was ok setting up my first system with 4 cards. Thanks, Dustin -Original Message- From: Michael Rowley [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 09, 2003 9:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: FXO cards Steven, I know how to RTFM. I have found no reference yet to how many x100P cards I can run in one box. Where did you find that reference? I have looked at the manual avaliable from *.org, downloaded the pdf of the x100p card tech info, it doesn't say it anywhere. 'course, I'm just sad as it would be a much cheaper way to go. I was afraid that there would be a problem with multiple cards in one box. I can get the four port Dialogic card for about 900$, or the 12 for 2K. But then were are the same price, or just a bit more than a channel bank and a t100p card. So, the docs say no more than 2 x100p cards sane, has anyone done it? put 5 or 6 in one box? Michael. On Tuesday, December 9, 2003, at 05:57 PM, [EMAIL PROTECTED] wrote: On Tue, 2003-12-09 at 15:18, Michael Rowley wrote: Hey guys, has anyone put 6 of the wildcat X100P cards in one machine? I am thinking about putting 6 in one machine, what is everyone elses experience Read the docs. 2 card maximum sane install. -- Steven Critchfield [EMAIL PROTECTED] Michael Rowley MD FP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
