Re: [Asterisk-Users] FXS or VOIP
Philip Edelbrock wrote: Jim Freeze wrote: [...] So for 5 phones, I would need 2 cards. And, the O'Reilly book says that I should not put 2 cards in the same box, so I would need another computer. [...] Whoa, I'm confused. Can't you use as many cards as you have slots? We've got just one 4-port card, but I've always assumed it was just a matter of purchasing and installing more to get 8 or 12 lines? It's a best practice to minimize interrupts and therefore it's usually not recommended to use more than 2 TDM400Ps in a system. In mid-density configuration you'd want to consider using a channel bank or a telephony card like Digium's TDM2400P (or a VoIP gateway). Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca www.gabcast.com ___ --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] 3RD REQUEST - Any Help Is Appreciated
[EMAIL PROTECTED] wrote: Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an I Don't Know. My third repost: Ok, I've been trying to figure out why my [EMAIL PROTECTED] won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. You should be posting to the [EMAIL PROTECTED] Help forum: http://sourceforge.net/forum/forum.php?forum_id=420324 or the AMP Help forum: http://sourceforge.net/forum/forum.php?forum_id=414452 or amportal-users mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca www.gabcast.com ___ --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] Most Stable Version of Asterisk
John Bittner wrote: Anyone know what version of Asterisk is the most stable running Real-time queues and agents ? Asterisk Business Edition? Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca www.gabcast.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Subscriptions
Douglas Garstang wrote: Maybe some from Digium will read this email and it will make a 0.001% contribution towards some of these things being fixed. Oh, and no... I can't switch to another solution. The decision was made above my head to go with Asterisk. It's my job to make it do all that 'enterprise-grade' stuff. My recommendation is for you to approach the decision makers of your organization and suggest that you submit Asterisk to the Open Source Maturity Model as described in Succeeding with Open Source by Bernard Golden. It sounds like you come from a pragmatist organization that demands a mature product and thus the book would provide much needed context for working with an open source project. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca www.gabcast.com ___ --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] Asterisk book feedback
Ross C wrote: Just curious what everyone (as in, the people that have read it or use it) thinks about the O'Reilly Asterisk book. I'd really like to delve into the nitty gritty of Asterisk, but I'm getting kinda tired of swimming through forums and Google results. I've been reading the wiki off and on for about a week now, but I'm wondering if a book would be the way to go to get a solid foundation. My IT career for the past 10 years has been based off of learn-as-I-go methods, but I'd really like to learn asterisk the right way. I have a couple Asterisk servers up and running and in use, but they're very small systems (~10 extensions, connected to 3 or 4 pots lines). I have some clients that want to use VOIP, but they're bigger businesses, and I'm not yet comfortable enough to roll out a bigger system. So if there are any other methods for learning Asterisk that I should consider, please do tell! Any opinions (on the book or otherwise) appreciated. Thanks! Another resource you might want to consider is Ted Wallingford's Switching to VoIP: http://www.oreilly.com/catalog/switchingvoip/ It uses Asterisk extensively in examples and provides good coverage of concepts like QoS, codecs, etc. that are important considerations in many Asterisk deployments. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca www.gabcast.com ___ --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] mpg123 on x86_64 (Opteron MP)
Patrick wrote: On Sun, 2005-12-11 at 09:56 +1100, Brad wrote: [snip] Anyone able to point me in the right direction to compile this app? It is running ubuntu.. Not really but iirc there is a Mandrake/Mandriva mpg123 SRPM floating around that had some x86_64 patches in it. Maybe you could try to track the SRPM down and use their patches to make it compile. We use MAD (http://www.underbit.com/products/mad/) on x86_64 systems. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] uip200 phone not work with 1.2
Jerry Geis wrote: Jason, I added: nat=never qualify=no and I still cant get a UIP200 to ring when calling it after using 1.2? Any other suggestions? Well, we're using UIP200's (BS 4.63 firmware) with 1.2. There is a bug (5780) re: rfc2833 g729 in 1.2 tarball but that doesn't seem to be the case here. Is the phone registering? sip show peers If yes, turn on sip debug and look for clues. Also look in the Asterisk log file (set in logger.conf). i.e. /var/log/asterisk/full Hope this helps. Regards, Jason Thanks, Jerry Jerry Geis wrote: / I have a handful of phones that work with 1.0.9. I was trying to upgrade // to 1.2 // and the UIP200 phones dont ring. // // below is my config for 1 phone. // // I tried it with and without the qualify=yes or qualify=no and did not // seem to make // a difference. still no ring. // // Any ideas on what might be the issue? // // THanks, // // Jerry // // // ; Jerry Phone // [528] // type=friend // dtmfmode=rfc2833; Choices are inband, rfc2833, or info // username=something // secret=something // disallow=all // allow=ulaw // allow=alaw // host=dynamic // context=smvoice-sip // callerid=Jerry 528 / Need to have: nat=never (or nat=route) Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] uip200 phone not work with 1.2
Jerry Geis wrote: I have a handful of phones that work with 1.0.9. I was trying to upgrade to 1.2 and the UIP200 phones dont ring. below is my config for 1 phone. I tried it with and without the qualify=yes or qualify=no and did not seem to make a difference. still no ring. Any ideas on what might be the issue? THanks, Jerry ; Jerry Phone [528] type=friend dtmfmode=rfc2833; Choices are inband, rfc2833, or info username=something secret=something disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid=Jerry 528 Need to have: nat=never (or nat=route) Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] Two Phones - Same extension?
Mike McMullen wrote: Hi All, I have an employee who works mostly in our office but maybe once or twice a week has to work from home to help care for her special needs child. As background we have AAH 2.0 running with 8 analog lines connected to two digium t400P cards. We have 10 sipura-841s as handsets in the office. I would like the employee to be able to make and take calls from her house when the she has to work from home. I'm leaning towards just installing s/w on her laptop with a headset for that setup. My question is how to handle setting her up so that she only has one extension shared between the office phone and her laptop. For this to work, do I need to unplug her phone from power/network in the office when she is at home or, hopefully, is there some other magic that can happen? AMP's Devices/Users configuration mode will give you what you want. 1 user, 2 devices (hardphone at the office, softphone at home). Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] IAX clients
Stefan Reuter wrote: Yes it would be really interesting if there are any IAX libraries for Java that are available under an open source license and that we might improve further. There is a growing demand for such a thing (for example see http://forums.digium.com/viewtopic.php?t=2431) Would be cool if we can create kind of a defacto standard, i.e. something that everybody uses. http://www.hem.za.org/jiaxclient/ (No affiliation.) Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] Asterisk 1.2 Aastra/Sayson 480i DTMF Problem
George Pajari wrote: We are experiencing problems with DTMF when using Asterisk 1.2 and the Aastra/Sayson 480i running 1.2.1.1002 firmware -- callers cannot navigate voicemail or other menus. Of course, we have the sip.conf set to RFC2283 (and nothing changed in our config files between 1.0.9 and 1.2 when things stopped working). Anyone else noticed this? We have a problem report into Sayson but are going to back out from 1.2 and revert to 1.0.9 in about 12 hours because of this and other problems with 1.2. Are you using g729? We noticed similar behavior with some phones (Uniden, Polycom) when they were using g729 rfc2833. Can't find the bug number... Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] AMP partially not working, Apache dying on segfaults?
Francesco Peeters wrote: Most of AMP seems to be working, but these do not: Trunks Outbound Routes PHPMyAdmin SysInfo These two applications are not packaged with AMP. When I click these, nothing happens, and the apache errorlog shows: [Sun Nov 20 00:22:56 2005] [notice] child pid 12771 exit signal Segmentation fault (11) [Sun Nov 20 00:23:00 2005] [notice] child pid 12783 exit signal Segmentation fault (11) It used to work, as I added an IAX2 trunk and Outbound routes before. I had this in an earlier install of [EMAIL PROTECTED], but as I needed to reinstall after I got EXT3 problems, I didn't make a fuss about it, but this time round I would really like to understand what is happening!... If you are running a BETA release of [EMAIL PROTECTED] you should post to the Help forum there. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] Newbie question. (Long)
Roger Hill wrote: I've been messing with Asterisk for a couple of weeks now. 1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu), downloaded the binary package. Now I'm trying to put the working installation on my production server along with HTTP etc. ( 700MHz, 256MB ram, uname -a gives Linux coach.hillconsult.com 2.6.14-1.1637_FC4 #1 Wed Nov 9 18:19:32 EST 2005 i686 i686 i386 GNU/Linux). That box, until yesterday, was running Fedora core 3. I tried the tarball download of 1.2.0.rc2, ran make OK, then make install, make samples. When I tried to run Asterisk, I got (immediately) Illegal Instruction. Tried on my FC4 laptop, worked just fine. Concluded I needed FC4, so upgraded the server yesterday. Six hours later... Reran make clean, make... Same problem. Then tried 1.2.0; same problem. Then tried 1.0.9; same problem. Finally removed everything to do with asterisk, pulled dowm 1.2.0 tar ball again, and re-installed. Same old problem, illegal instruction. I suspect bad RAM. I'd memtest it. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel Compile Error
Goran Donev wrote: The error message is: You do not appear to have the sources for the 2.6.9-22.0.1.EL kernel installed. make: *** [linux26] Error 1 I am installing it on a Cento 4.2 server. Can someone shed some light on this? yum install kernel-devel (or yum install kernel-smp-devel) Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] TDM04b on FreeBSD
Alejandro Mejia Evertsz wrote: Hi list! I successfully installed a Digium TDM04B card on FreeBSD 5.4 using zaptel drivers for FreeBSD (installed with ports). I'm using Asterisk CVS-Head and the card works fine, but when placing or recieving a call on any of the 4 fxo ports, users hear (both sides) a clicking noise. I also have a Wildcard X100P installed, and uses the same configuration (on zapata.conf) but that card doesn't make that strange noise during conversations. Please let me know if someone had this problem before me, and what you did to correct it. I don't know what else to try. Could the TDM400P be sharing an interrupt? systat? Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] TDM400 Card
Shaun Singh wrote: Is there some kind of limit to the number of TDM04B cards you can use in your Asterisk system (Red Hat 9, kernel 2.4, Asterisk CVS-v1-0-11-11/16/04-13:41:01))? I have 2 cards right now(rev B) with 8 analog lines connected to 8 FXO modules. I wanted to add 2 more analog lines but the third card (rev I) refuses to recognize the two new FXO modules. Digium have said their newer version TDM cards are backward-compatible. There is no problem with the PCI slot or IRQ. I'm using the motherboard (Asus P4P800-E) as recommended by Digium. Any ideas? Digium's TDM2400P is better suited to your configuration. Maybe ask Digium if they have some kind of trade in program? Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] Sangoma 102 installation problem
You need gcc-c++ FaberK wrote: Hi friends, during the installation, I receive that problem, but I've installed both Flex and, of course, C/C++ libraries. My OS is CentOS 3.6, completely updated. Any ideas??? Thanks - Compiling WANPIPE WanCfg Utility ... Failed! !!! WANPIPE WanCfg Compilation Failed !!! Possible solution: FLEX Package not installed Non-standard C/C++ library (eg: ulibc) -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] sangoma a104d install
Jason Kim wrote: Hi, While a104d install on asterisk 1.2 and CVS-HEAD patch for zaptel.c failed. Is it avaiable not yet? Their docs say they patch 1.0.9 and CVS-HEAD (not CVS Stable). So if you're running CVS_HEAD and it didn't work perhaps contact Sangoma. You can patch manually too. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] AMP and voicemail passwords
James Armstrong wrote: Anyone here using AMP and having problems with users chaning their voicemail passwords? They stick until I go into AMP and make changes then reload. The AMP settings contain the old password and are overwriting the new one saved by the user. What am I doing wrong or what is the correct way to do it? Please post to the amportal-users mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users and/or Help forum: http://sourceforge.net/forum/forum.php?forum_id=414452 Please include in your post info such as version of AMP, where the users are trying to change their passwords (i.e. phone (0 - 5) or ARI), etc. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems Installing MPG123 on a 64 Bit System
Vahan Yerkanian wrote: I'd recommend using native mp3 support that is available in CVS HEAD, as madplayer mp3 decoder gives a lower quality sound (audibly more cranky/noisy). I don't follow CVS commits but if that's the case the mpg123 target should be removed from the asterisk Makefile and the native mp3 support should be documented in ..doc/README.mp3 Jason Becker wrote: Steve Totaro wrote: Anyone know how to get around this? I am stumped. # make mpg123 [ -f mpg123-0.59r.tar.gz ] || fetch http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz [ -d mpg123-0.59r ] || tar xfz mpg123-0.59r.tar.gz make -C mpg123-0.59r linux cc1: error: CPU you selected does not support x86-64 instruction set Use madplayer instead. There are several reasons why Digium the Asterisk community should part ways with mpg123. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems Installing MPG123 on a 64 Bit System
Vahan Yerkanian wrote: I'd recommend using native mp3 support that is available in CVS HEAD, as madplayer mp3 decoder gives a lower quality sound (audibly more cranky/noisy). For archive purposes... Had to do some digging to find out what you were talking about - I guess you are referring to the section Using native Asterisk format_mp3 for Music on Hold* found here: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf Some of the comments suggest that this solution is far from robust. Would be interested in hearing others experience with this solution for MoH. Jason Becker wrote: Steve Totaro wrote: Anyone know how to get around this? I am stumped. # make mpg123 [ -f mpg123-0.59r.tar.gz ] || fetch http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz [ -d mpg123-0.59r ] || tar xfz mpg123-0.59r.tar.gz make -C mpg123-0.59r linux cc1: error: CPU you selected does not support x86-64 instruction set Use madplayer instead. There are several reasons why Digium the Asterisk community should part ways with mpg123. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] Asterisk management portal
Tomislav Parčina wrote: Does anybody have detailed instruction how to Install AMP? I have tried to install it using Installation Guide on their pages but I'm unable to satisfy AMP's PERL module dependencies. Please post to the amportal-users list: http://lists.sourceforge.net/lists/listinfo/amportal-users and/or Help forum: http://sourceforge.net/forum/forum.php?forum_id=414452 Please include in your post what PERL dependencies you are unable to satisfy and why. Please provide standard output in your post. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems Installing MPG123 on a 64 Bit System
Steve Totaro wrote: Anyone know how to get around this? I am stumped. # make mpg123 [ -f mpg123-0.59r.tar.gz ] || fetch http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz [ -d mpg123-0.59r ] || tar xfz mpg123-0.59r.tar.gz make -C mpg123-0.59r linux make[1]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r' make CC=gcc LDFLAGS= \ OBJECTS='decode_i386.o dct64_i386.o decode_i586.o \ audio_oss.o term.o' \ CFLAGS='-DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX \ -DREAD_MMAP -DOSS -DTERM_CONTROL\ -Wall -O2 -m486 \ -fomit-frame-pointer -funroll-all-loops \ -finline-functions -ffast-math' \ mpg123-make make[2]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r' make[3]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r' gcc -DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX -DREAD_MMAP -DOSS -DTERM_CONTROL-Wall -O2 -m486 -fomit-f rame-pointer -funroll-all-loops -finline-functions -ffast-ma th -c -o mpg123.o mpg123.c `-m486' is deprecated. Use `-march=i486' or `-mcpu=i486' instead. cc1: error: CPU you selected does not support x86-64 instruction set make[3]: *** [mpg123.o] Error 1 make[3]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r' make[2]: *** [mpg123-make] Error 2 make[2]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r' make[1]: *** [linux] Error 2 make[1]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r' make: *** [mpg123] Error 2 Use madplayer instead. There are several reasons why Digium the Asterisk community should part ways with mpg123. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] Uniden UIP200 Issues
Jeff Herring wrote: Phone won't register on LAN port registers but doesn't work on PC port. SIP to SIP works. Anyone have a Configuration that works out there? Phone has 4.63 Firmware Make sure you have nat=never (or nat=route). Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] My contribution to the issue of code- reversal
Andrew Kohlsmith wrote: On Monday 10 October 2005 08:12, Federico Alves wrote: reverse code and it surely is a legitimate operation. Open source is far more convenient, but how do we charge for the product? The business model is not there: the more popular the product is, the more remote the possibility of the creator making any money from it. Take Digium. The more experts on Asterisk pop-up, the less demand is for Digium services. In This is *precisely* why I believe that Digium's current model is terribly wrong. They are competing with the very people that make Digium money. In my humble opinion, I believe that Digium should not offer support to end users. [snip] How about the consultant side of things? How do you make your money when you are trying to cultivate and enitre business subculture around your product? Through licensing and support of the distribution and consultants. You provide tier-1 support and training materials to registered consultants around the globe. Perhaps a modest yearly license [snip] I can't really comment on the hardware side of Digium's business model other than it would seem Digium will have increased competition from Intel (Dialogic) and Sangoma (new Shark line). On the services side you seem to be describing the Compiere model: http://www.compiere.org/partner/index.html Excerpt: -begin- Our target customers are companies and people who offer Compiere solutions. We want to build a real partnership - not just distribution channels. You don't have a real partnership, if you are competing with your customers. Consequently, we want to restrict our services to helping our customers to offer / resell Compiere. We offer second level support, consulting, training and licenses to our Partners enabling them to offer comprehensive solutions to their customers. -end- Transitioning to a model like that of Compiere seems to make sense. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] Sprint Nextel sueing over VoIP patents
/asterisk-users/2004-February/036311.ht ml http://www.business2.com/b2/web/articles/0,17863,1059204,00.html FYI: Voiceglo and theglobe.com are the same company for all intents and purposes. Therefore, I am very interested to see if this is merely co-incidental or if there is a reason that Sprint picked out two providers that use Asterisk in their core. Despite hysteria or misinformation on this (and other) lists, there is no direct information that I've seen that this is Sprint making a blanket patent lawsuit against anyone using VoIP. Perhaps this is just some specific feature that they have a legitimate patent on which has been infringed. I doubt this is a codec patent issue, nor an equipment patent issue (as previously discussed on -biz list.) Is there anyone with better detail on the lawsuit specifics able to comment? JT ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.4/108 - Release Date: 9/21/2005 -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.4/108 - Release Date: 9/21/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] MAX PRI for single server
Andrew Kohlsmith wrote: On Thursday 08 September 2005 10:26, Simone Cittadini wrote: Is it true ? My boss is just asking me if it is possible to stuck 4* TE411P in a single server, for a total of 480 lines, someone can assure me it is possible/impossible (manageable/unmanageable) from real-life experience ? Don't do it. The most I've seen in any RELIABLE setup is two TE4xx cards in a decent server. Why would you want that many calls terminating on a single box anyway? Why not use two or three boxes to spread the load out and also reduce the chance of a problem on a box taking out your ENTIRE communications network? -A. Sage advice, but out of curiousity what happened to Digium's T3 card (the DS3000P)? Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] MAX PRI for single server
Matthew Boehm wrote: Jason Becker wrote: Sage advice, but out of curiousity what happened to Digium's T3 card (the DS3000P)? IIRC, Digium's T3 card isn't expected to be channelized. Also, IIRC it will have no on-board EC and no on-board encoding so I can't imagine the machine you would need to process that many calls. Hmm, looks like someone in the know needs to update the wiki: http://www.voip-info.org/tiki-index.php?page=Digium+DS3000P Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] MAX PRI for single server
Matthew Boehm wrote: Jason Becker wrote: Hmm, looks like someone in the know needs to update the wiki: http://www.voip-info.org/tiki-index.php?page=Digium+DS3000P Wow. Guess I'm not. Matthew, I in no way meant to imply that you are not in the know. I guess what I meant to say was perhaps someone from Digium could provide an authoritative response to the question of specifications for the card, its availability, etc. The wiki indicates that the card is channelized; you indicated that it may not be. Sincerely, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] FW: [EMAIL PROTECTED] - requesting help on oh323, ISDN BRI and iConnectHere DID
AbdelRahman Tarzi wrote: I know almost nothing linux, and don't really know that much about Asterisk (proper).. but I was 'pulled' by this subject and grabbed an [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] installation CD (version 1.3) and just went with it. Newbie doesn't quite describe it, I'm a banker.. this simply goes to show how easy Asterisk is becoming (I certainly hope this direction was meant to be inviting to people like me). I've been at this for a little over three weeks now. I've researched these topics but have not found satisfactory answers to the following: If there are places where an answer could be found, I'd appreciate some pointer(s). Please post to the [EMAIL PROTECTED] forum: http://sourceforge.net/forum/forum.php?forum_id=420324 and/or amportal mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users or Help forum: http://sourceforge.net/forum/forum.php?forum_id=414452 oh323 Once it (oh323) was installed, I was able to dialout from an h323 device with little problem. I have not, however, been able to setup the h323 device as an extension (or use it to communicate with other h323 devices, or through it to the pstn.. I was confused because in creating an extension (or a trunk) in [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] there was nothing to indicate that h323 was available. The dropdown in extension creation does not include oh323 (or some such) item. I need a clue as to how to setup an extension and how to setup a trunk (using an h323 device that is). The current version of AMP (1.10.008) supports the creation of Custom Trunks, which in your case could be an oh323 trunk. The next version of AMP (1.10.009, currently in second beta) will support Custom extensions, which in your case would mean that you could setup your oh323 endpoint and be able to leverage AMP's dialplan. ISDN Could someone please tell me whether the AVM Fritz card is a low pain solution to connecting to a BRI ? I've tried connecting an ASUS card (doesn't work with the available drivers/application) so I'm really asking in order to buy. I only have one BRI - this is at home) so would appreciate any help. Can't comment on the pain aspect but documentation on this configuration exists: http://www.voip-info.org/tiki-index.php?page=Asterisk+AVM+Fritz+CAPI+Driver+Install (Note: this would be configured in AMP as a Custom Trunk as well) iConnectHere From day one, I was able to create the trunk to dialout of iConnectHere, but despite finding several claimed correct settings for receiving its DID, I've not been able to. When I connect a Grandstream 101 to the line and power it up, it has absolutely no problem receiving calls from that DID but no matter what I try, I'm unable to receive into Asterisk (@home) .. Just so we're clear, I am receiving calls from fwd and through fwd from an ipkall number.. and I'm also recieving calls from the FXO on a sipura spa3k.. - iConnectHere don't help more than point to the wiki .. (and what I find there doesn't work). No comment. Never heard of this VoIP Service Provider. Just a note to describe my handicap with linux: I'm unable to capture a log (or perhaps it's captured and I'm just not aware where).. Asterisk: /var/log/asterisk/full (see /etc/asterisk/logger.conf) Linux: /var/log/messages [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] seems to dialout and receive calls using Macros.. I suspect it's a clever way of managing the setup, but I'm not sure where the various portions of SIP.conf, extensions.conf, extensions_additional.conf, extensions_custom.conf or indeed oh323.conf. - are relevant. Please search the [EMAIL PROTECTED] forum and/or amportal list forum for more info on these topics. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] [Asterisk-Dev] SIP Benchmarking / Stress Testing
Sherwood McGowan wrote: Anyone have a good tool(s) to use for simulating a bunch of calls? Benchmarking or stress testing? I only need SIP protocol, and do appreciate any replies...I realize I could google it, but I am looking for opinions as well. There is SIPp: http://sipp.sourceforge.net/ Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] OH323 with [EMAIL PROTECTED] - seems incomplete
Abdel Rahman Tarzi wrote: Following the install, I do not seem to have the option to create an oh323 extension or trunk. Something that I need to do. I realize it’s possible to edit the .conf but I needed to ask whether this was “normal” – doesn’t seem like it is to me. Naturally, I’m apprehensive that editing the .conf files manually may be overwritten by AMP, but even if not, it would seem like it should’ve been the norm to have “added” an entry like oh323 to SIP, IAX2 etcetera types of trunks. Also, extension types (which work from a drop-down list) should have been modified. I would appreciate if someone familiar with the application could inform whether this is “normal” before I attempt to add extensions and trunks manually. The next version of AMP (1.10.009) will remove any assumptions from the dialplan about the technology being used. In short, the dialplan begins to care about devices - and the devices can be any Asterisk supported technology. In the AMP interface you can create a Custom Device and provide an appropriate dial string. Feel free to check out 009 (currently in beta) to test your configuration on a Development machine. Please note that the implementation is subject to change. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 with [EMAIL PROTECTED] - seems incomplete
Abdel Rahman Tarzi wrote: I installed oh323 and everything seemed to go smoothly (compile everything upto calling through using oh323). I must admit, there is some behavior that’s doesn’t seem right but generally, I’m able to dial-out of any oh323 device whether to an extension or to a trunk. Audio is sometimes muted when dialing out until the extension or dialed number answers. Sound quality is good when it’s there. Following the install, I do not seem to have the option to create an oh323 extension or trunk. Something that I need to do. I realize it’s possible to edit the .conf but I needed to ask whether this was “normal” – doesn’t seem like it is to me. Naturally, I’m apprehensive that editing the .conf files manually may be overwritten by AMP, but even if not, it would seem like it should’ve been the norm to have “added” an entry like oh323 to SIP, IAX2 etcetera types of trunks. Also, extension types (which work from a drop-down list) should have been modified. I would appreciate if someone familiar with the application could inform whether this is “normal” before I attempt to add extensions and trunks manually. I apologize for spamming the list... I failed to mention that Custom Trunks support exists in the current version of AMP (1.10.008). Here is the text from the Custom Dial String tooltip: Define the custom Dial String. Include the token $OUTNUM$ wherever the number to dial should go. examples: CAPI/:b$OUTNUM$,30,r H323/[EMAIL PROTECTED] OH323/[EMAIL PROTECTED]: vpb/1-1/$OUTNUM$ -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Looking for Web based SIP endpoint
Bruce Ferrell wrote: Nico Giefing wrote: you need a sip-provider? - Original Message - From: Bruce Ferrell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 20, 2005 12:38 AM Subject: [Asterisk-Users] [OT] Looking for Web based SIP endpoint I think the title more or less says it all. Is there any such animal? TIA No, I need an endpoint I can put on a webpage https://sip-communicator.dev.java.net/ Don't know the current state of functionality with Asterisk. I couldn't get it to work many months ago - even with help from the developer. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Soft Phone
Jason Walker wrote: Any suggestions for IAX phones on Linux (without Wine preferred)? Kiax. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Mahler's Book - New Project
David Stude wrote: I'm currently gearing up for a possible PBX replacement project using Asterisk, and I'm just breaching the iceberg of information that's available. I typically like to have something thick with pages in front of me. Mahler's book was the first one to come up and it seems like a good place to start. However, the big name bookstores tell me it'll take up to three weeks, and this project simply can't endure that wait. Does anyone know where it's possible to get a paper copy *quickly*? Perhaps your local bookstore will have this O'Reilly offering: http://www.oreilly.com/catalog/switchingvoip/ It makes heavy use of Asterisk for instructional purposes. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Problem with CDR web page
Cavanna, Richard wrote: I am having a problem with my CDR web page (AMP). There is a banner on the page saying YOu MUST ACCESS THE CDR THROUGHT THE ASTERISK MANAGEMENT PORTAL! and it will not show any calls just No calls in your selection. I have checked the database and calls are being recorded in the database. I check the defines.php and it seems to be correct. Does anyone have any insight as to my problem?? ___ Please see: http://sourceforge.net/tracker/?group_id=121515atid=690572func=detailaid=1172758 Please post to the amportal-user mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users and/or Help forum: http://sourceforge.net/forum/forum.php?forum_id=414452 for issues specific to AMP (and its bundled applications). Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Visual ring notification
I have put a pbx into a resort with Polyycom phones, everythign works great, except the kitchen staff cannot hear the phone ring. I know many legacy systems employ a big red flashing light, any ideas on doing something similar? FYI, the Uniden UIP200 has a big red flashing light. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] OrderlyQ installations?
Jason Kawakami wrote: What experience can be shared about installing and running the OrderlyQ application? I have a bunch of queues set up and want to start adding some additional apps and this one looked promising but I have very little java experience and it doesn’t seem to be running properly. I don't mean to hijack this thread since the OP specifically mentioned OrderlyQ but - ICD (Intelligent Call Distributor) looks like it adds some sophistication to Asterisk ACD functionality and provides a flexible framework for customization: http://icd.sourceforge.net/tiki/tiki-index.php I'd be interested in hearing about ICD - specifically skills-based call routing, if anyone has done it. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] 12 FXO ports into Asterisk
Darren Wright wrote: I'm looking at $1500 for the bank plus $500 for the T1 for a 10-port FXO solution. 3 TDM cards are significantly less than that. Any other ideas? Voicetronix has an OpenSwitch12 card that can do 12 FXO ports. Telephonyware is selling them for $1450. Haven't used them, but no one has mentioned these yet, so I am. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] NVFaxdetect
Eric Rees wrote: I have googled this and come up empty. Has anyone had any problems compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am getting when I run make. app_nv_faxdetect.c: In function `nv_detectfax_exec': app_nv_faxdetect.c:210: error: structure has no member named `cid' app_nv_faxdetect.c:227: error: structure has no member named `cid' app_nv_faxdetect.c:265: error: structure has no member named `cid' make[1]: *** [app_nv_faxdetect.o] Error 1 Did you try changing the define in the file(s)? i.e.: // Use the second one for recent Asterisk releases #define CALLERID_FIELD cid.cid_num //#define CALLERID_FIELD callerid Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] FATAL: Error running install command for wctdm
Ronald Wiplinger wrote: FATAL: Error inserting wctdm (/lib/modules/2.6.8-24.11-default/extra/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm Freed a Wildcard module wctdm unsupported by SUSE/Novell, tainting kernel. wctdm: disagrees about version of symbol zt_receive Check out: http://lists.digium.com/pipermail/asterisk-users/2005-March/096532.html Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Re: @Home AMP call recording documentation
Dan Littlejohn wrote: If someone could point me to the incoming/outfoing call recording feature for AMP it would be greatly appreciated. Look in the Extensions Admin. Per Extension you can set: Record INCOMING Record OUTGOING The script /var/lib/asterisk/bin/archive_recordings will create some meaningful directory structure under /var/spool/asterisk/monitor. Until some management capability is added to the interface (i.e. playback) you'll have to get the files via (win)scp or make them available through a network service like samba. Please post to the amportal mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users if you need more detail. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Re: Re: Digium Website Update: Asterisk Business Edition
Brian Capouch wrote: As I have been reading this thread one missing angle that perhaps should be addressed by those who are bothered by the current licensing scheme is this: what alternative means exist out there for Digium to try to ensure their corporate existence? We can all see that in the backroom down in Huntsville there is a pretty fair-sized phalanx of people whose time is spent on Asterisk, not Digium's other business. Those people need to eat, and Digium needs to make a profit in order to insure that Asterisk isn't simply just maintained, but can grow and respond to what we all have to concede is a very rapidly-changing technological environment. [snip] One thing I see lacking in this thread is a discussion of alternatives that would meet the relatively small list of desiderata: keep Asterisk open and free, make enough money to pay for the ongoing cost of Asterisk development, and provide enough return that it would make sense for Digium to exist as a commercial enterprise. How else could it be done? John Koenig offered 7 open source business strategies: http://management.itmanagersjournal.com/management/04/05/10/2052216.shtml?tid=85 Optimization, dual-licensing, consulting, subscriptions, patronage, embedded, and hosted open source David Pool offered 3 more strategies: publishing, hardware, and training So in addition to dual-licensing, Digium also sells hardware (that point seems to have been forgotten) and offers consulting (implementation/integration/support). Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Softphone for Linux desktops
Eric Bishop wrote: We are successfuly running an Asterisk server with standard SIP hard phones and it is working well. We are looking to deploy some soft phones on our Linux desktops. There seems to be several floating about. Anyone out there with some good/bad experiences with particular Linux softphones. We only need g711 and prefer IAX but a SIP one will do Check out Kiax. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Tools for effectively manage Asterisk
[EMAIL PROTECTED] wrote: We tried AMP, very powerful but incomplete (CAPI is very important to us); The 1.10.008 version of AMP supports Custom Trunks. Text from the AMP tooltip: -begin- Define the custom Dial String. Include the token $OUTNUM$ wherever the number to dial should go. examples: CAPI/:b$OUTNUM$,30,r H323/[EMAIL PROTECTED] OH323/[EMAIL PROTECTED]: vpb/1-1/$OUTNUM$ -end- Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Tools for effectively manage Asterisk
[EMAIL PROTECTED] wrote: Is there any way to separate AMP stuff from asterisk, in other words to have AMP, apache and so on on a different pbx than asterisk? AMP does require file-system access for configuration of Asterisk. With some clever engineering the interface could be decoupled from the PBX. At present this decoupling is not a design goal of AMP. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] AMP and dialparties.agi
[EMAIL PROTECTED] wrote: Hi, i dont know if this is the right place to ask for AMP questions, im using it in production and have noticed high cpu usage and even hangs with the dialparties.agi scripts, is this scripts really necessary?, why not use DIAL command directly? thanks Please post to the amportal mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users or help forum: http://sourceforge.net/forum/?group_id=121515 Please include as much information as possible about your environment to help reproduce the problem (Linux distribution/version, show version from Asterisk CLI, calling behavior that might trigger the problem, etc.) Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Suggested Reading for VOIP
Tyler Younger wrote: Can anyone recommend any reading to help me get a better understand of VOIP technology and all the terms used in this discussion thread? I've read many documents on the wiki but I find it a bit lacking. Coming in June: http://www.oreilly.com/catalog/switchingvoip/index.html Excerpt from the Full Description: To help you better grasp the core principles at work, /Switching to VoIP/ uses a combination of strategy and how-to using Cisco internetworking devices, various makes of IP telephone equipment, and the Asterisk open source PBX software by Digium. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] AMP Handset Provisioning
Hi Stuart, Stuart Hirst wrote: I have in the past written a hack to AMP 1.10.004 to automatically generate Polycom configuration files such that when an extension is created you can select the type of handset as a Polycom and then enter the handsets MAC / Serial on the page and AMP would automatically create the correctly formatted config files. This was a dirty hack written in haste. I am about to start doing a patch for the latest version of AMP that will allow developers to added their own handset config plug-in or at least publish the method used to recreate correctly the previous Polycom hack such that anyone should be able to build their own handset configuration templates and have AMP auto provision whilst maintaining some consistency. Has anyone else done anything similar given that duplication is the root of all evil ? A Feature Request for this very capability was submitted earlier today: http://sourceforge.net/tracker/index.php?func=detailaid=1178008group_id=121515atid=690575 We'd welcome your contribution and can work with you to add this valuable feature to AMP. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems loading zapata module under suse 9.2 (cvs stable from 5 days ago) ?
Robert Rozman wrote: I've compiled Asterisk cvs stable (few days ago) unde Suse 9.2 without any problems. We're using te110p and wcte11xp module that is autoloaded by Suse 9.2. Card goes green after reboot, but this meesages appear in logs: Mar 22 11:28:51 linux kernel: Zapata Telephony Interface Registered on major 196 Mar 22 11:28:51 linux modprobe: FATAL: Error inserting torisa (/lib/modules/2.6.8-24.11-smp/extra/torisa.ko): Unknown symbol in module, or unknown parameter (see dmesg) Mar 22 11:28:51 linux modprobe: FATAL: Error running install command for torisa Mar 22 11:28:51 linux kernel: module torisa unsupported by SUSE/Novell, tainting kernel. Mar 22 11:28:51 linux kernel: torisa: disagrees about version of symbol [snip] Any idea what's wrong and where to start digging ? Have you tried the suggestion here: http://www.voip-info.org/wiki-Asterisk+Linux+SuSE e.g. for module in /lib/modules/`uname -r`/misc/*; do rm -i /lib/modules/`uname -r`/extra/$(basename $module); done followed up by a depmod. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Redhat 9 Music on hold
Daniel Burget wrote: I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines connected via TE405P. Everything works great, except MOH. I added an exten with MusicOnHold(30), and it plays just fine. Conferences have music when no one is in. I have SIP phones. When I place a call on hold, the CLI give no indication the call is on hold. I have set musiconhold(default) everywhere, removed it from everywhere, nothing seems to help. I am using 59r of MPG123, and do not have MPG321 installed. I did a 'make mpg123' from asterisk, make no difference. I believe it is a bug: http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/85000 although I don't know if a bug was ever filed. I had a cursory look at the time we were bitten by this but couldn't find one. Pulling a newer CVS Stable and rebuilding resolved the issue. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] cordless/wireless system with a ip base station?
Chuck wrote: does anyone know of a 2.4 or 5 ghz cordless phone system that has an ip base station? Uniden has the UIP1868: http://www.uniden.com/productsupport2.cfm?product=UIP1868 But there's no documentation to speak of. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems Starting Asterisk - FOP AM Portal
John Cianfarani wrote: Asterisk seems to start fine but the FOP op_server.pl doesnt seem to want to start. Ive tried running it by hand as the asterisk user but it doesnt spew any errors, and I cant find any log files that would help me troubleshoot this issue. Ive searched different archives and google but cant find much related to this problem. Please read the following: http://fedora.redhat.com/docs/selinux-faq-fc3/ and if you still have issues post to the amportal mailing list and/or Help forum. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] A bit of a survey: What do do if youneedmorethan 4 C.O. lines
Jim Van Meggelen wrote: Yep, that's a possibility, but it's rather more kludgy than I'd like. (heck, the channel bank and T1 is more kludgy than I'd like - an integrated card would be my preference). I haven't followed this thread closely but have you looked into the Voicetronix OpenSwitch cards? http://www.voicetronix.com.au/hda.htm Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Uniden UIP200, please help
Robert Burcham wrote: I have seen no responses to my earlier post: http://lists.digium.com/pipermail/asterisk-users/2005-February/089944.html and my problem persists. Would someone please share their configs and firmware versions? I sent you an email (off-list) the other day with configs attached. If you didn't receive it ping me off-list and I'll resend. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] /usr/bin/ld: cannot find -lidn
Matt Schulte wrote:
Bueller? Is this a lib of some kind? Google and lists bring up nada,
this is from ast cvs head latest on Fedora Core 3.
/usr/bin/ld: cannot find -lidn
collect2: ld returned 1 exit status
make[1]: *** [app_curl.so] Error 1
make[1]: Leaving directory `/usr/src/asterisk/apps'
make: *** [subdirs] Error 1
[EMAIL PROTECTED] asterisk]# uname -a
Linux zoot 2.6.9-1.667smp #1 SMP Tue Nov 2 14:59:52 EST 2004 i686 i686
i386 GNU/Linux
Looks like asterisk is using cURL:
CURLLIBS=$(shell curl-config --libs)
ifneq (${CURLLIBS},)
APPS+=app_curl.so
And cURL uses libidn:
http://curl.netmirror.org/libs.html
So you likely need:
http://mirrors.kernel.org/fedora/core/3/i386/os/Fedora/RPMS/libidn-0.5.6-1.i386.rpm
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
403.244.8089
www.coalescentsystems.ca
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Re: [Asterisk-Users] AMP with SUSE 9.2
Keith Burns wrote: *Hi,* *I have the newbie guide from AMPs website and (fair enough) it is all about whitebox linux.** Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ?* Please post to the amportal mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users or Help forum: http://sourceforge.net/forum/?group_id=121515 SUSE does some things differently - the main difference is the apache2 (httpd) configuration. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Install Method
[EMAIL PROTECTED] wrote: I would suggest you go with the easy road : - install CentOS : http://www.centos.org/ - then download Asterisk 1.0.4 (latest stable) : ftp://ftp.asterisk.org/pub/asterisk/ - install it by following this document : http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation [EMAIL PROTECTED] 0.3 uses CentOS 3.3 with a recent Asterisk: Asterisk CVS-v1-0-01/22/05-02:50:58 built by [EMAIL PROTECTED] on a i686 running Linux It also bundles AMP ;-) Project page: http://asteriskathome.sourceforge.net/ Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Any movement on IAX being submitted to a standards body?
Hi All, I'm aware of this statement by Mark circa July 2004: -begin- SIP is an IETF standard. While there is some fledgling documentation courtesy Frank Miller, IAX is not a published standard at this time. -end- And the thread How far is IAX to be a Standard circa November 2004. It is not my intention to start a protocol war; I was discussing IAX and SIP with an Editor at ZDNet and the question came up - I just wanted to provide him with the current status of the effort (if one is underway). Thanks. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] New verision of AMP - 1.10.004
Kanuri, Seshu (Company IT) wrote: Has this version improved the install process from what it was, to something where a guy with average intelligence (AKA dummy) can install without the need of a consultant. The Installation Guide provides step-by-step instructions: http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.1.pdf It might be better to characterize AMP as a blueprint or set of instructions for building your own *-based telephony server. You do need to invest some time and effort to build it - or you can pay someone else to do it for you. We do of course welcome efforts from the community to make the project better. An installation routine that handles all the popular Linux distributions is non-trivial. At present we are focused on providing the user base oft-requested features - like support for VoIP trunks which we delivered in this version. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] New verision of AMP - 1.10.004
Denis Galvão wrote: What about zap channels support in AMP!? Yes, zap channels are supported. FXS devices are not supported through our administration interface. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] AMP - Fax Detections
Sean Cook wrote: Does anyone know of any obscur reference for detecting an incoming fax. I currently have AMP running and everything else is working great. Installed the spandsp patches and software... using the default AMP extensions.conf, I start sending a fax, I hear it pick up and transfer to voicemail after 20s. Fax is set for system... Here is the detail from the extensions.conf [global] FAX_RX = system Looks like you have done some customization, please post to the amportal mailing list or Help forum. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] AMP - Fax Detections
Sean Cook wrote: Actually it is the default install, no changes yet... Maybe the dial group getting answered before fax detection... [EMAIL PROTECTED] root]# grep FAX_RX /etc/asterisk/extensions_additional.conf FAX_RX = system FAX_RX_EMAIL = [EMAIL PROTECTED] These parameters are in the *_additional.conf file, not extensions.conf. Also they are under [globals] not [global]. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Pitching Asterisk
Sean Cook wrote: The company I work for is looking at vendors for a PBX, one of the requirements is VoIP. I have been sitting there listening to people pitch very proprietary implementations of VoIP where you are locked in to their hardware, their interface... I know a little bit about asterisk (set up a couple offices with it... run it at home...) and would like to pitch it to this company. Does someone have a decent presentation that I could use as a starting point? Basically I am looking for a business oriented (not too technical) overview of asterisk, or asterisk for suits. http://graphics.cs.uni-sb.de/VCORE/Publications/mark_spencer/mark.smil A presentation by Mark Spencer that discusses the Business and Technical Details of Asterisk. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] udev or not?]
Lee wrote: I'm using White Box Linux, which is derived from RHEL 3. Kernel is 2.4.x AFAIK, udev is a feature of 2.6 kernels. I'm still not sure whether this system uses udev or not, but * works as long as I do a modprobe wcfxo prior to starting Asterisk. My goal right now is simply trying to get that module loaded at boot time on this system. Add the following to /etc/rc.d/rc.local: /sbin/modprobe wcfxo Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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: pc
TC wrote: I'd have to agree with the channel bank plan. You don't even need to buy a used channel bank. You can get a new Rhino for $1500: http://www.channelbanks.com/ dont forget these Rhino are *FXS ONLY* I noticed just this morning that a provider of *-based systems was advertising/selling FXO modules for the Rhino. I just called Rhino and they told me that FXO modules will be available in the New Year. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Best VM codec for Linux/OS X/Windows environment
Chris TenHarmsel wrote: Hi all, I've done some minimal searching on this topic, but haven't come up with anything conclusive. Right now we're using WAV to store voicemail messages, that then (for the most part) get sent to users in email when they have new voicemail. The reason for this is that we have a very mixed environment, with most people in linux, some in OS X and a couple in Windows, and I haven't been able to get wav49 or GSM working on all three platforms. Is anyone in a similar situation and have gotten either of those two compressed codecs to work? Is there something I have to install in linux to get either working (some sound library or something)? My experience has been that mplayer can handle the wav49 (GSM-encoded wave file) but that snackAmp can't. I (obliquely) suggested that Digium consider FLAC in another thread, I know OS X has an MacFLAC... To be honest, I'm not sure of the state/availability of a FLAC plugin for WMP. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Small PBX setup
[EMAIL PROTECTED] wrote: Hi all, I know that this has been passed around before, and I know that it happens about every 3 months or so, but evertime the answers change, so I thought I would pass it around again. A company I work for has 3 incomming lines and 4 phones. They require voicemail and MOH. Their phone systems VM hard drive died today, they were quoted a $2000 to replace it. I started to talk to them about asterisk and what it can do, and they were very happy. Have you considered AMP? http://amp.voxbox.ca Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Low Volume WAV Files in Email Attachments
Steven Critchfield wrote: Second, read the rants on licensing. Unless you find a BSD licensed mp3 encoding library and convince Mark of it's need, it is unlikely to make it to the core code base. When snackAmp blew up on GSM-encoded wav files I did some cursory research and found FLAC: http://flac.sourceforge.net/ The license for the libraries is a BSD-variant. I'm not an expert on audio formats so no flames please. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Phone Selection
Alex Brecher wrote: Anybody here have suggestions on these phones [Sayson 480i or the Cisco CP-7960] please ? IMHO you should also consider the Uniden UIP200. These phones offer good value, are professional looking and the firmware is much improved. We know that Uniden Engineering is running Asterisk in-house; they have been very responsive to issues we brought to their attention. Lots of people rave about the Polycom phones but we received an official response that they will not support their phones in an Asterisk environment. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Uniden UIP200 -- configured, but not working?
Ken D'Ambrosio wrote: Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? [22] type=friend host=dynamic context=local-access canreinvite=no qualify=300 callerid=Uniden SIP Phone 22 mailbox=22 secret=bar nat=no Try: dtmfmode=rfc2833 (The UIP200 only support rfc2833) and: nat=never (The UIP200 does not like rfc3581 (rport)) Check out: http://www.voip-info.org/wiki-UIP200 My colleague wrote the Issues with the UIP200 and Asterisk section. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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?
dean collins wrote: The Voxbox guys have done a great job getting the product up and running but are flat out servicing commercial customers so this was a nice way to help out. We have been very responsive to questions posted to the amportal mailing list and forums. Furthermore, the newbie installation guide was provided a few _days_ after your request for better documentation. Roger has indicated in this thread that he was able to install AMP by following the guide. Regards, Jason Check back on the wikki in about 1 weeks time for the final document. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oliver Stone Sent: Wednesday, November 24, 2004 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk gui? I had to give up after attempt to install AMP.. It's got very nice user interface, that is, AFTER you have sucessfully installed it.. I see it's got great potential, but current release is very difficult to install, even with the newbie guide. If you have fewer than 10 extensions to configure, it's probably not worth your effort to going thru all the trouble to install AMP., hoping you didn't screw up anything during the install process. . Just wondering how difficult it would be for AMP devs to develop a install wizard or a batch file that can automatically execute the install and download necessary dependencies... until then, I guess I'll be continuing to manually config my asteisk files On Mon, 22 Nov 2004 23:12:00 -0700, Brian [EMAIL PROTECTED] wrote: Perhaps rather than a GUI we should be wanting an IDE (as in Integrated Development Environment, not Intelligent Drive Electronics . . . bloody overlapping acronyms . . . but I digress . . . ). Even some basic syntax highlighting would improve the readability of extensions.conf immensely. Anyone know how to make THAT work in vim? I've hacked one together for UltraEdit that works reasonably well, but that's a Windows editor. Jim, Several months ago I was working on a VIM Asterisk syntax highlighting file, but stopped working on it due to lack of interest. I might try to add some stuff to it again if I have some spare time later this week. What I had done can be found at http://snurl.com/asterisk_syntax_vim . Contributions are welcome; just hit the edit button on the wiki :) -Brian ___ 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 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 -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Possible to display which extensions are in use on the phone's display?
Hi Richard, Richard Hansberry wrote: Hi all, We're considering using Asterisk in our small (8 user) office. There is one feature that we have on our current phone system that I haven't seen in the documentation that I've read that I'd like to be able to replicate with Asterisk. On our current phones (Iwatsu) we have a button on the phones for each extension that lights up when that extension is ringing or is in a call, so I can see at a glance if one of my coworkers is on the phone before I go barging into his office. Also, if I am in a coworker's office and my phone rings, I can hit my extension button on his phone and answer the call. Can this functionality be reproduced by Asterisk, and if so, which SIP handsets would you suggest using to do something like this? Right now, I'm looking at the Cisco 7960, but I'm open to other options. This question was dealt with in a recent thread. What it comes down to is a paradigm shift from a key system unit (which is what you describe) to a PBX. You can use an application like Flash Operator Panel (http://www.asternic.org/) in conjunction with Asterisk to see who is on their phone (extension). Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] $200 AMP documentation bounty
dean collins wrote: There is a $200 bounty for helping document a step by step guide to AMP, anyone on this list interested in making easy money feel free to contact me. I want to thank Dean for giving me this opportunity to address some of the concerns raised in regards to AMP. AMP began as a value-add to a _small business_ turn-key solution. We felt strongly that in order to approach small businesses with our integrated solution we needed a GUI administration interface that handled the mundane. When we looked at what was available circa Feb/March of this year we realized that we would be better off writing it ourselves. Most of the other (freely available) candidates still appear to me to be thin wrappers to the configuration files that require the end user to have a knowledge of Asterisk and programming logic. From the outset, AMP - or more specifically our piece that allows for setup of extensions, IVRs, etc. - was designed so that an Office Manager of a small business could use it for the day-to-day operational stuff. It's clear to me that our design goal has resulted in the attention of people whose expertise does not extend to the Linux/UNIX world. I personally am thrilled at this. I also understand the frustration these people must feel when met with the INSTALL instructions. We know that if a person has software installation / development experience on Linux that a fully functioning AMP can be installed in a couple hours. We cannot take responsibility for the Linux learning curve. As one poster to this thread commented, the best way to overcome it is to take a crash course in general Linux admin. I think this is good advice for the time being until we find a solution to the problem of providing the user base with well-written documentation. As to the specifics of providing the said documentation I am open to suggestions. Open source economics make it challenging for the software provider to also provide the other elements of a mature product. Elements like documentation and training. In fact, part of the reason we open sourced our piece of AMP was so that we could solicit help from the Asterisk community to service this need. asteriskdocs.org was a suggestion. We, Coalescent Systems, would also consider hiring a professional technical writer if we can ensure that the demand for documentation was significant and that purchase of the documentation (ala JBoss' model) would support the paid position. AMP is still targeted towards the small office and small business. I think we have made that very clear on the AMP homepage: http://amp.voxbox.ca In short, AMP, by design, limits the feature set of Asterisk and is intended for small office and small business use. It does NOT align well with the needs of larger organizations and is not suited for use in those environments. If Gregory Junker et al. want to design a standalone, cross-platform alternative I applaud their initiative. Finally, I want to thank everyone that has shown an interest in AMP and Coalescent Systems. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Clipping at start of call
[EMAIL PROTECTED] wrote: The UIP200 does exhibit the clipping when calling the internal Asterisk voicemail. (ie password is clipped to word.) I've added a wait in the dialing plan to fix that. We reported the clipping to Uniden as far back as July. Sorry I don't have any answers for you... even the beta of an upcoming firmware demonstrates this behavior. Please ensure that you report it to Uniden. Thanks. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Clipping at start of call
Michael Loftis wrote: --On Sunday, November 07, 2004 16:37 -0500 Jason Becker [EMAIL PROTECTED] wrote: We reported the clipping to Uniden as far back as July. Sorry I don't have any answers for you... even the beta of an upcoming firmware demonstrates this behavior. Please ensure that you report it to Uniden. I've also experience clipping though with cisco SIP phones as well as occasionally when dialing into our IVR from my Vonage (Cisco ATA) VoIP line at home. Only the Uniden UIP200 demonstrates the behavior for us. We have not heard it with the Grandstream phones or ATA, Sipura 2000 or Sayson 480i. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Linux and Windows
james wrote: The trouble is innertia. Most Windoze folks are so much into their Windoze routine, they won't even use a Linux or BSD box if you install it for them. I have got a friend in the UK who is always complaining about his Windoze box being down and having to rebuild it from scratch because of viruses, DLL hell, hardware quirks and god knows what else. For years I told him that he could get rid of all that once and for all very easily. When I started with Asterisk, there was finally something that got him interested because he's a telephone junkie. Well then, how did you expect your win-weenies to admin a hardware based phone system then? It sounds like they didn't admin them. Why should they admin a new * phone system? The fact that a phone system can exist in software and run on a computer doesn't mean that just anyone can admin it. Besides a myriad of OS related issues, there is a huge volume of telecom related issue involved too. Larger phone systems have always been based on a varient of Unix (and not just ATT/Avaya/Lucent). Admining Unix is part of being a phone system administrator. I too favour Linux over Windows but this kind of rhetoric doesn't help accelerate the adoption of *. I also believe in empowering customers. One of the benefits of AMP/voxbox is that the routine day-to-day administration of an *-based solution is abstracted from the configuration files. I fail to see any reason why knowledge of programming logic and a background in UNIX administration is necessary to setup an IVR. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Where to buy POLYCOM phones?
Jonathan Miller wrote: Hi all, I'm trying to put together a list of gear w/prices to implement an asterisk system. Does anyone know a good place to buy polycom phones? Their website isn't much help. Specifically looking for IP500 and IP600 phones. Thanks again! In Canada you can get them from CCP. West Canadian contact: Stacey Chamberland Canadian Communication Products Inc. 3657 Wayburne Drive Burnaby, BC V5G 3L1 [EMAIL PROTECTED] tel: 1-800-665-5726 fax: 1-604-263-9399 Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.voxbox.ca ___ 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] New Open Source Project: Asterisk Management Portal
Salutations, In hopes of accelerating the adoption of Asterisk and changing the landscape of the small business marketplace, we are contributing our administration interface to a new project that aims to bundle best-of-breed applications to produce a canned (but fully functional) turnkey small business phone system. Details of the project can be found here: http://amp.voxbox.ca Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.voxbox.ca ___ 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] GPL Violations (Was: Advice on OS Choice)
From: Brian West brian at bkw dot org: -begin- Personally if the copyright holder of a GPL software such as Asterisk or any GPL software allows someone to violate the GPL then allow them to skid by is setting a bad precedent for the GPL. I think if you find a GPL violation you must stand up and make an example out of said company. If a copyright holder doesn't standup and say something that only weakens the GPL's power. The GPL is still untested in the court system. -end- IANAL but you can make any changes you want provided you do not distribute/sell the changed software without making the changed code available. And AFAIK *Available* does not mean ship with. Available can mean download, CVS or other means. Distribution is what triggers the GPL. And use within a company does not, I believe, constitute distribution. I also think GPL violations are rare. But there was a highly publicized alleged violation by Cisco/Linksys: http://lkml.org/lkml/2003/6/7/164 To be honest I don't even know the end of that story (if it has ended)... probably some bureaucratic snafu. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.voxbox.ca ___ 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 SIP Communicator
JORA ROME wrote: I wan work * whith SIP Communicator, it is posible?, what is configurations? who can helpme? Thanks I couldn't get it to work either, circa early February. I had some correspondence with the developer and sent him logs, etc. but nothing ever came of it. He did say there were authentication problems. Maybe you'll have better luck. He can be reached at: [EMAIL PROTECTED] Cheers Resgards, Jose _ Charla con tus amigos en línea mediante MSN Messenger: http://messenger.latam.msn.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
Re: [Asterisk-Users] 3-4 port FXO card recommendations
Scott Laird wrote: On Mar 31, 2004, at 8:06 AM, Mark Buckaway wrote: In setting up Asterisk, I'm looking to dump my current phone system (Nortel Venture). I presently have three POTS lines. I would use a VOIP provider, but now are presently available in the Toronto, ON, CANADA area that support user owned hardware/software. I need a 416/647 area code number. In looking at FXO cards to atteach to the lines, I've seen the Digum single port FXO card and their multipoort FXS cards. So far, the cheapest route seems to be to buy three Digum FXO cards; however, the system I want to use them it only has some three/four PCI slots. Having one card with three FXO ports would be useful. I'm also open to using an external device. I bought a DLink dual port MCGP FXS device and that works well. The name of the game is CHEAP though...and Ebay has not been very useful. Rumors say that Digium is about to release FXO modules for the 4-port TDM400P. Assuming that they're priced similarly to their FXS ports, that would give you 4 FXO ports in 1 slot for around $300 US. That would be great cuz we were quoted $875 Cdn for a single Voicetronix OpenLine4 (4-port FXO) from a reseller in Ottawa (they do have discounts for bulk purchases mind you). Mark, contact me off-list if you want details but the reseller can be found off of Voicetronix's website. Cheers ___ 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] Plugging Asterisk Security Holes....
[EMAIL PROTECTED] wrote: Another topic of interest is securing the box itself. Does a firewall (hardware outside of the box or a linux based firewall) suffice the need? Depends what you are protecting against. If you want to assume some services are exploitable, you could try to break some of the exploits by firewalling off all ports not used, and prevent all outgoing connections from your box except for ports you use on that box. If you use netfilter, you can create rules that apply to user-ids as well, so you could allow asterisk more privileges. Nessus (http://www.nessus.org/) is a great vulnerability assessment tool one can use to determine if services are exploitable. Cheers ___ 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] question about CPU usage
Back in February I found * pinning the CPU (Slackware 9.1, Feb CVS). I
ran a strace and found that it was looping on this:
-begin-
write(1, \nUse STOP NOW to shutdown Asteri..., 35) = -1 EIO
(Input/output erro
r)
write(1, *CLI , 6) = -1 EIO (Input/output error)
read(0, , 1) = 0
ioctl(0, SNDCTL_TMR_STOP, {B38400 opost isig icanon echo ...}) = -1 EIO
(Input/o
utput error)
-end-
Mark kindly responded to me:
-begin-
In the mean time try running asterisk with no console. This is bug #864.
Preliminary analysis shows that after a restart now, one of the
ioctl()'s performed by editline fails with -1. Ignoring the ioctl made
the CLI non-functional. Happy to get any help I can in this regard.
-end-
Hope this helps.
Cheers
Martin Pycko wrote:
try to do ps -auxm to list all the threads of the asterisk.
Then connect with gdb to the thread that takes 99% of CPU and find out
what it's doing.
Martin
On Mon, 22 Mar 2004, Bill Hamlin wrote:
Nope same problem. I just started it and did a couple of ps aux's and got
this output:
[EMAIL PROTECTED] root]# ps aux|grep ast
root 20140 91.6 1.3 115880 6676 ? R15:43 1:10
asterisk -vgcd
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Re: [Asterisk-Users] windows alternitives to Asterisk?
hank smith wrote: hello I am just curious if there is any windows alternitives to Asterisk? can I also use them with free world dialup? thanks hank I've never used either of these and I'm certainly no authority on the subject, but here are a couple Windows based alternatives I've come across while studying the marketplace: http://www.nexpath.com/ http://www.artisoft.com/ You didn't say whether TDM was important to you, or whether free as in beer and/or freedom is desirable. I'd love to see a competitive analysis of * vs. alternatives (Windows or otherwise) on the voip-info wiki. Cheers ___ 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] Best OS for Asterisk
Steve Kennedy wrote: Probably a dumb question, but what's the best Linux variant to use to build/run an Asterisk server. Hardware is Compaq DL360 with a Widcard 410. Debian/Fedora Core ? Steve As others have said: Whatever you're most comfortable with. Having said that though, I'm partial to Slackware. It has a reputation for being stable and it has a simple package management system and adheres to a remote management philosophy (for lack of a better description). Cheers ___ 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] Asterisk pins CPU
Hello All.
Many times I've done a top and found that Asterisk is pinning the CPU,
even when Asterisk isn't being used (this is on a DEV box):
2044 root 15 0 5232 5228 2608 R98.6 8.5 626:58 0 asterisk
I'm running a recent build of Asterisk on Slackware (2.4.24 kernel):
www*CLI show version
Asterisk CVS-02/07/04-21:34:06 built by [EMAIL PROTECTED] on a i686 running Linux
A strace shows that it's looping on this:
-begin-
write(1, \nUse STOP NOW to shutdown Asteri..., 35) = -1 EIO
(Input/output erro
r)
write(1, *CLI , 6) = -1 EIO (Input/output error)
read(0, , 1) = 0
ioctl(0, SNDCTL_TMR_STOP, {B38400 opost isig icanon echo ...}) = -1 EIO
(Input/o
utput error)
-end-
The machine does not have a soundcard and my /etc/asterisk/modules.conf has:
noload = chan_alsa.so
noload = chan_oss.so
If I restart asterisk all is well, for awhile. I'm not sure what
triggers this behvaior. Anyone else getting this behavior?
I wish the lists were searchable... :(
Thanks.
Cheers
Jason
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Re: [Asterisk-Users] Asterisk pins CPU
Upon reading the bug I can confirm that I was doing:
asterisk -vvvc
through a ssh session/client (the Asterisk box is headless) and then for
whatever reason I would close that session/client and come back later
and start another ssh session/client and do a:
asterisk -r
and then probably a:
restart gracefully
triggered the behavior.
Thanks for the quick reply.
Cheers
Jason
Mark Spencer wrote:
In the mean time try running asterisk with no console. This is bug #864.
Preliminary analysis shows that after a restart now, one of the
ioctl()'s performed by editline fails with -1. Ignoring the ioctl made
the CLI non-functional. Happy to get any help I can in this regard.
Mark
On Sun, 8 Feb 2004, Jason Becker wrote:
Hello All.
Many times I've done a top and found that Asterisk is pinning the CPU,
even when Asterisk isn't being used (this is on a DEV box):
2044 root 15 0 5232 5228 2608 R98.6 8.5 626:58 0 asterisk
I'm running a recent build of Asterisk on Slackware (2.4.24 kernel):
www*CLI show version
Asterisk CVS-02/07/04-21:34:06 built by [EMAIL PROTECTED] on a i686 running Linux
A strace shows that it's looping on this:
-begin-
write(1, \nUse STOP NOW to shutdown Asteri..., 35) = -1 EIO
(Input/output erro
r)
write(1, *CLI , 6) = -1 EIO (Input/output error)
read(0, , 1) = 0
ioctl(0, SNDCTL_TMR_STOP, {B38400 opost isig icanon echo ...}) = -1 EIO
(Input/o
utput error)
-end-
The machine does not have a soundcard and my /etc/asterisk/modules.conf has:
noload = chan_alsa.so
noload = chan_oss.so
If I restart asterisk all is well, for awhile. I'm not sure what
triggers this behvaior. Anyone else getting this behavior?
I wish the lists were searchable... :(
Thanks.
Cheers
Jason
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