[asterisk-users] Dell, HP, Digium, homebrew - what do you use
With all the talk about servers, how about adding your server hardware only in a single line without quote to this thread? It will be easier to tally the results. Mine is a home made PC. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting Conference Request - Can thisbe done ?
Jon, I don't know the purpose of it either but that is what the client wants. - Original Message - From: Jon Pounder [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, September 18, 2007 5:59 PM Subject: Re: [asterisk-users] Interesting Conference Request - Can thisbe done ? Quoting Dovid B [EMAIL PROTECTED]: Hi List, I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to). what would they hear then ? if they can't hear anyone else, just an extension that goes nowhere they talk into would do what you need. I am guessing you didn't explain clearly enough though. Any ideas ? Thanks. Dovid Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell, HP, Digium, homebrew - what do you use
Compaq P3 1GHz server (about 6 or 7 yrs old) running 2gb RAM, 40(?)G HDD, single AX100P. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.14.3/1054 - Release Date: 06/10/2007 19:12 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which LDAP OID for iphones
Eric, I had the same feeling when I found it difficult to find sip hardphones oid stuff. Maybe th right thing is to split Users and Ressources data in different repositories. Users data might be stored in AD, and Ressources in an homemade database. I hope that people who disagree wouldn't hesitate to tell and argue. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk status in Debian
Thanks for the info, this is helpful :) On 10/4/07, Faidon Liambotis [EMAIL PROTECTED] wrote: Hello, This is a update on the current status of Asterisk in Debian. Apologies for the really long mail, it is targetted both to users and maintainers :) I'm Ccing asterisk-users as a one-time thing; users that are interested can subscribe to our list[1] for updates to prevent noise on a non-Debian list. Please Cc pkg-voip-maintainer on replies. 1: http://lists.alioth.debian.org/mailman/listinfo/pkg-voip-maintainers sarge/etch status - About a month ago, I fixed all the long-standing knownn vulnerabilities in both sarge/oldstable (1.0.7) and etch/stable (1.2.12). The updates are present security.debian.org a Advisory has been released (DSA-1358[1]), thanks to Debian's Security Team. These updates are fixing CVE-2007-1306, CVE-2007-1561, CVE-2007-2294, CVE-2007-2297, CVE-2007-2488, CVE-2007-3762, CVE-2007-3763 and CVE-2007-3764 (...). 1: http://www.debian.org/security/2007/dsa-1358 lenny status 1:1.4.11~dfsg-4 has been recently uploaded to unstable. The previously mentioned block by the openh323 dependency which currently fails to build in unstable (binutils bug: #440015) has been workaround-ed (by having less strict shlibs in openh323) From our POV, it's a good candidate for lenny/testing. However: - it depends on perl and net-snmp versions that are not present in testing and are not in a shape to be there; we'll need new versions from the respective teams. - asterisk needs to go together with yate because of a shared libpri dependency. However yate is being blocked[2] by gtk+2.0. - more importantly, asterisk produces an Internal Compiler Error of GCC 4.2 on hppa (#445336). Until it builds successfully there, it cannot migrate to testing. 1: http://bjorn.haxx.se/debian/testing.pl?package=asterisk 2: http://bjorn.haxx.se/debian/testing.pl?package=yate 1.4.12 -- Digium released 1.4.12 the day before yesterday. I have committed all the changes needed and we are now up to date. Fortunately, many of our fixes that I reported upstream have been merged. I have manually ported bristuff 0.4.0-test4 to 1.4.12; it needed many changes compared to the previous upstream updates. I will forward my changes to kapejod so that he can hopefully release a new version. supplementary packages -- * asterisk-addons (-mp3, -mysql, -ooh323c) are finally present in Debian and should be ready to migrate to lenny after Asterisk does. Digium released a new version along with 1.4.12 and I will update this ASAP. * asterisk-chan-capi, asterisk-spandsp-plugins, asterisk-oh323 had recents uploads and all are in a good shape. * I am going to drop rate-engine from the archive (#444712) since it has no users, it wasn't released with etch, has open bugs for a really long time and is unmaintained by upstream. * I tried compiling chan_misdn together with the mISDN maintainer (Simon Richter) and failed because of an mISDN API mismatch. Need to take another look. * asterisk-gui needs to be uploaded; Tzafrir? * are we going to upload ARI? If not, we should drop it from our SVN. * zaptel is in a good status and it's the only package from the suite that is migrating to testing. Things TODO that come to mind are: a) fixing a bug which results in /lib/modules/2.6.foo/modules.* files in amd64 and b) evaluate a switch to OSLEC as the default echo cancellator. Tzafrir is doing an excellent job on maintaining this package by himself :) * Right now, we are shipping asterisk-sounds-main which is the main asterisk sounds in English in GSM format -- exactly as shipped in the original tarball by Digium. Kilian, Tzafrir and me were pondering on the idea of shipping separately all sounds as shipped by Digium in all formats (besides WAV), each in a separate package. This should serve our users better but has an obvious problem of size. This is not decided yet. ABI issues -- Most -if not all- of these plugins build-depend on asterisk-dev i.e. use Asterisk's development headers. These headers are tied to the ABI and this can only be expressed in dependencies manually. asterisk-chan-capi was compiled with 1.2 asterisk-dev, had a = 1.2 dependency but segfaults on 1.4 (#441237). There are currently no similar problems that I know of. However, we should expect more of these when we transition to 1.6 which will most probably have a different ABI. I'm leaning towards a solution: * Add a Provides: asterisk-1.4 to asterisk. * Replace Depends: asterisk (= 1.4.0) (or similar) with Depends: asterisk-1.4 on all external modules. This should help in *breaking*, dpkg-wise, the modules when a new version is uploaded which in turn will prevent a new version from entering testing until all plugins are recompiled. pushing our work upstream - On the
[asterisk-users] Good Book to learn SIP
Hi List, I am trying to learn SIP in its entirety. I have so far found: http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403 Anyone know of any other books that are worth reading ? Thanks. Justin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for recommendations on Nufone and Gamachi
I have used nufone in the past and the quality is good. - Original Message - From: Alejandro Lengua To: Commercial and Business-Oriented Asterisk Discussion ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, September 22, 2007 7:48 PM Subject: [asterisk-users] Looking for recommendations on Nufone and Gamachi I am having several problems with voipjet. Due to this I am evaluating a second (and a third alternative). I have recommendations for Gafachi and Nufone. Can anybody share their experiences with any of them? Thanks in advance Alejandro -- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell, HP, Digium, homebrew - what do you use
EPIA 5000 wit 512MB RAM, TDM400P (1xFXO) and FritzCard for ISDN. Running Mandrake 9.2 as later distros I have tried (Fedora) wont play nicely! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell, HP, Digium, homebrew - what do you use
Home built Jetway VIA CN700 1.2Ghz, Dual GigE, 1G DDR2, Linux From Scratch 6.3, Asterisk 1.4.12, Samba, LAMP Untangle (shortly) -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell, HP, Digium, homebrew - what do you use
Alan Lord wrote: Home built Jetway VIA CN700 1.2Ghz, Dual GigE, 1G DDR2, Linux From Scratch 6.3, Asterisk 1.4.12, Samba, LAMP Untangle (shortly) + Single FXO (x100p) -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New Embedded Distro
Hi All, A couple of weeks ago I noticed Askozia PBX, which is a new embedded Asterisk OS distro at http://askozia.com/pbx. This caught my attention for two reasons; it uses v1.4 of Asterisk, and it uses the m0n0wall development framework to build on FreeBSD with a PHP based GUI. I've used m0n0wall for years, and FreeNAS also, which shares the same OS/GUI framework. I booted the latest build of Askozia PBX on a small system fors testing. The GUI looks nice. I' not certain if I want v1.4 in production as yet. If that proves to be the case then Askozia looks like a candidate to replace Astlinux, which is v1.2 and has essentially no GUI. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Oddball time problem in CID
Hi Al, That was it, Thank you!!! Al lists wrote: check tz option in your voicemail.conf On 10/5/07, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I have a really oddball time problem. When I check the server time using 'date' it is correct. When I review the time in Freepbx (under time conditions) it is correct. When I look at the time stamp in the CDR it is correct. When I review the time displayed for a voicemail in a web browser it is correct. When I hit *98 and then my extension the CID says a time that is some 6 hours off (early)??? I am really confused where could CID be getting this bogus info??? I am using Centos 4.5, Asterisk 1.2.7.1 http://1.2.7.1 and Freepbx version 2.3.0.3 http://2.3.0.3 Thanks Chuck Bunn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.14.4/1055 - Release Date: 10/7/2007 10:24 AM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk status in Debian
On Thursday 04 October 2007 23:26:16 Faidon Liambotis wrote: Backporting stuff from trunk may be error-prone and is not easy to draw a line of which stuff we should backport. I'm open to suggestions on other modules that may have sense in backporting. There are a number of packages for which backports have already been done from trunk. The three that I've done are: app_stack (Gosub, Return) which incorporates optional arguments and will serve as a replacement for Macro (except without the limit on the number of nested executions); func_odbc, which permits the retrieval of multiple rows from a single query via a special mode; and cdr_adaptive_odbc, which matches up CDR variables with the identically named database table column name, simplifying the action of adding new bits of data to your CDR, without cramming them all into userfield (and conversely, letting you drop columns that are not important to you). http://svncommunity.digium.com/view/app_stack/1.4/ http://svncommunity.digium.com/view/func_odbc/1.4/ http://svncommunity.digium.com/view/tilghman/branches/1.4/ -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Curiosity Max Calls
Hi is there a tool to know what was the maximum calls that asterisk managed? Thank you ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curiosity Max Calls
Il Neofita wrote: Hi is there a tool to know what was the maximum calls that asterisk managed? http://areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curiosity Max Calls
You can test it with sipp: http://sipp.sourceforge.net/ Alexandros Doug Lytle schrieb: Il Neofita wrote: Hi is there a tool to know what was the maximum calls that asterisk managed? http://areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 Doug -- BSc. Alexandros Manakos University of applied science - Hagenberg Sichere Informationssysteme Tel Germany: 0211 - 592170 E-Mail: [EMAIL PROTECTED] Web: www.manakos.de ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Asterisk doesn't answer to incoming call from pstn.
Note: forwarded message attached. - Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. ---BeginMessage--- Hi: I installed A102d sangoma's card successfully but Asterisk doesn't answer to incoming call from pstn and console doesn't show any message of incoming call in the other word when I dial the number of E1 I can't connect to asterisk and dial the number of extension. I'd apreciate any idea. My configuration files: zaptel.conf: Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:3 bus:1 span: 1] span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 #Sangoma A102 port 2 [slot:3 bus:1 span: 2] span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 zapata.conf: ;autogenerated by /usr/local/sbin/config-zaptel do not hand edit ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A102 port 1 [slot:3 bus:1 span: 1] switchtype=national context=from-pstn group=0 signalling=pri_cpe channel = 1-15,17-31 ;Sangoma A102 port 2 [slot:3 bus:1 span: 2] switchtype=euroisdn context=from-pstn group=0 signalling=pri_cpe channel = 32-46,48-62 extensions.conf: [from-pstn] exten = 611,1,Answer() exten = 611,2,Echo() - Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us.---End Message--- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Embedded Distro
Astlinux does seem to be growing cob webs a bit. Askozia doesn't support Zaptel cards in the GUI and not sure if it is possible to configure them manually. There is no Voicemail storage mechanism yet. It's still very basic but a nice start. -Original Message- From: Michael Graves [mailto:[EMAIL PROTECTED] Sent: Sunday, October 07, 2007 6:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] New Embedded Distro Hi All, A couple of weeks ago I noticed Askozia PBX, which is a new embedded Asterisk OS distro at http://askozia.com/pbx. This caught my attention for two reasons; it uses v1.4 of Asterisk, and it uses the m0n0wall development framework to build on FreeBSD with a PHP based GUI. I've used m0n0wall for years, and FreeNAS also, which shares the same OS/GUI framework. I booted the latest build of Askozia PBX on a small system fors testing. The GUI looks nice. I' not certain if I want v1.4 in production as yet. If that proves to be the case then Askozia looks like a candidate to replace Astlinux, which is v1.2 and has essentially no GUI. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting DTMF digits
I forgot to add that this is a T1 ISDN PRI line on which I am sending the DTMF digits. Regards Arpit On 10/5/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi, Is there any way to get the DTMF digit preferably in the extensions.conf . The dtmf digits would be entered by the user like1234567890P1234# . It doesnt matter whether to put 'P' or '#' as long as I am able to extract the digits 1234. I saw a couple of solutions but all of them required entering the DTMF digits once the call was established. Regards Thanks -- AM -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Embedded Distro
Hey everyone, On 10/7/07, shadowym [EMAIL PROTECTED] wrote: Astlinux does seem to be growing cob webs a bit. Askozia doesn't support Zaptel cards in the GUI and not sure if it is possible to configure them manually. There is no Voicemail storage mechanism yet. It's still very basic but a nice start. This is probably my fault for not updating the screenshots page in such a long time. The most common Zaptel cards should be working in the GUI now. There's still a lot of work to be done before a 1.0 but we're cruising along as quickly and as stably as we can. Regards, -Michael ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curiosity Max Calls
SIP is only one piece of the puzzle. This will show you the max calls using SIP but what about TDM? What about IAX? I have seen 95 simultaneous calls come in over a PRI with NFAS (Sangoma) eating 60%-70% of the CPU (HP DL320 3ghz single core with a gig of RAM) with no codec conversion. Ulaw all the way, just TDM to SIP handoff. I wonder about these 8 port cards. How beefy of a server would you need to light up all the channels in the same scenario as above. How about the once announced Digium DS3 card (that I never saw come to market), that board must have some powerful onboard circuits or require a very powerful server SGI Numalink setup. I guess with dual procs and quad core systems, maybe thats not an issue anymore. Thanks, Steve defjam01 wrote: You can test it with sipp: http://sipp.sourceforge.net/ Alexandros Doug Lytle schrieb: Il Neofita wrote: Hi is there a tool to know what was the maximum calls that asterisk managed? http://areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Book to learn SIP
Justin Case wrote: Hi List, I am trying to learn SIP in its entirety. I have so far found: http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403 http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403 Anyone know of any other books that are worth reading ? Thanks. Justin The RFCs are online as well as anything else you could want to know. Are you just a book person? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with 1.4.12.(1)
Hello all, I have successfully complied and installed 1.4.11 and recently noticed that 1.4.12 (and subsequently 1.4.12.1). Unfortunately, while 1.4.11 was running fine, 1.4.12.1 seems to build fine, but after a few minutes, I receive the following message a few times: ERROR[17670]: astobj2.c:114 INTERNAL_OBJ: bad magic number 0x11 for 0x6174c0 If I try to make a call, I get: Segmentation fault and then asterisk crashes. I'm on Mac OS X Server 10.4 (PPC). Primarily, I am wondering if anyone else has experienced this, and if so, if there is perhaps some build option that I should be using that I am not, in order to avoid this. Any help would be greatly appreciated! -- George Qualley IV D.J. G Mint Thunder Lightning Sound and Lighting Voice - (515) 255-3698 • Fax - (515) 309-1910 www.MidwestDJs.com • www.ThunderAndLightning.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curiosity Max Calls
The board never came to market 1. because the demand. 2. impossible to do with zaptel. /b On Oct 7, 2007, at 3:23 PM, Steve Totaro wrote: How about the once announced Digium DS3 card (that I never saw come to market), that board must have some powerful onboard circuits or require a very powerful server SGI Numalink setup. I guess with dual procs and quad core systems, maybe thats not an issue anymore. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Embedded Distro
shadowym wrote: Astlinux does seem to be growing cob webs a bit. Askozia doesn't support Zaptel cards in the GUI and not sure if it is possible to configure them manually. There is no Voicemail storage mechanism yet. It's still very basic but a nice start. -Original Message- From: Michael Graves [mailto:[EMAIL PROTECTED] A couple of weeks ago I noticed Askozia PBX, which is a new embedded Asterisk OS distro at http://askozia.com/pbx. This caught my attention for two reasons; it uses v1.4 of Asterisk, and it uses the m0n0wall development framework to build on FreeBSD with a PHP based GUI. I've used m0n0wall for years, and FreeNAS also, which shares the same OS/GUI framework. I booted the latest build of Askozia PBX on a small system fors testing. The GUI looks nice. I' not certain if I want v1.4 in production as yet. If that proves to be the case then Askozia looks like a candidate to replace Astlinux, which is v1.2 and has essentially no GUI. A few comments on this as one of the Astlinux developers. Asterisk 1.4 is and has been in a beta branch for some time. The developers feel that while 1.4 is the future, in many cases 1.2 is a much more stable platform. Also while there hasn't been a release with significant changes recently, there has been significant action behind the scene. Askozia looks like it has made a huge amount of improvements and changes since it was first introduced. There are significant differences between the two platforms. Each project has different goals, different licenses and different strengths. Astlinux is built on a very customizable framework that can allow it to be used for other applications besides Asterisk. We have several supported vpn's, full firewall support, and a published development environment. There's support for Digium, Rhino and Sangoma zaptel hardware. Askozia appears to be centered around the GUI. While that's not necessarily a bad thing, an experience Asterisk user may want to manipulate the raw config files. Networking support appears to be limited. In any case, I applaud the author of Askozia for his efforts. Taking on a project of this size takes a considerable effort. In many cases it's a volunteer effort. Regards, Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Embedded Distro
On Sun, 07 Oct 2007 18:25:36 -0500, Darrick Hartman wrote: A few comments on this as one of the Astlinux developers. Asterisk 1.4 is and has been in a beta branch for some time. The developers feel that while 1.4 is the future, in many cases 1.2 is a much more stable platform. Also while there hasn't been a release with significant changes recently, there has been significant action behind the scene. Askozia looks like it has made a huge amount of improvements and changes since it was first introduced. There are significant differences between the two platforms. Each project has different goals, different licenses and different strengths. Astlinux is built on a very customizable framework that can allow it to be used for other applications besides Asterisk. We have several supported vpn's, full firewall support, and a published development environment. There's support for Digium, Rhino and Sangoma zaptel hardware. Askozia appears to be centered around the GUI. While that's not necessarily a bad thing, an experience Asterisk user may want to manipulate the raw config files. Networking support appears to be limited. In any case, I applaud the author of Askozia for his efforts. Taking on a project of this size takes a considerable effort. In many cases it's a volunteer effort. Astlinux has been very good for me for over two years. In my application its principle role is Asterisk, the various VPN, firewall and other features not required. Normally I would not be attracted to a GUI at all. Witness the fact that I have no interest in Trixbox. However, the m0n0wall GUI has been very effective in m0n0wall and FreeNAS. If the Asterisk implementation of same is reasonably flexible then it would be a significant advantage over Astlinux. Time will tell. Each will no doubt have its place. Asterisk on small format hardware is an important frontier. Ease of use for typical users is also important. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk hangs on STRPTIME
hello,
running asterisk 1.4.11 on CentOS 4.5
I am getting no response on function STRPTIME() the system just hangs,
STRFTIME() is working fine as seen below. Same thing happens whether
I called in from a softphone or via teliax.
While executing the following code :
;
exten = s,n,Set(v_ts=)
exten = s,n,Set(v_ts=${STRFTIME(|America/New_York|%Y-%m-%d)})
exten = s,n,NoOp(${v_ts})
;
exten = s,n,Set(v_ts=)
exten = s,n,Set(v_ts=${STRPTIME(2007-10-06
05:04:03|America/New_York|%Y-%m-%d %H:%M:%S)})
exten = s,n,NoOp(${v_ts})
;
I get the output :
-- Executing [EMAIL PROTECTED]:6] Set(IAX2/4883-1, v_ts=) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(IAX2/4883-1, v_ts=2007-10-07) in
new stack
-- Executing [EMAIL PROTECTED]:8] NoOp(IAX2/4883-1, 2007-10-07) in new
stack
-- Executing [EMAIL PROTECTED]:9] Set(IAX2/4883-1, v_ts=) in new stack
If this is a reported bug that has been fixed in 1.4.12, I can upgrade to
it,
but I'd like to know.
tia.
-baji.
--
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 and G723 and how to install it
Hi List; From where I can buy the G.729 and G.723 licenses, and how I can install it on Asterisk so I can use it? Anyhelp? Regards Bilal Don't let your dream ride pass you by. Make it a reality with Yahoo! Autos. http://autos.yahoo.com/index.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk hangs on STRPTIME
could this be the reason for my problem ?
( I am using a 64 bit AMD processor )
2007-09-12 20:12 + [r82285] Tilghman Lesher [EMAIL PROTECTED]
* main/stdtime/private.h, main/stdtime/tzfile.h,
include/asterisk/localtime.h, main/stdtime/localtime.c: Working
on issue #10531 exposed a rather nasty 64-bit issue on
ast_mktime, so we updated the localtime.c file from source.
Next we'll have to write ast_strptime to match.
1.4.12 changelog
http://svn.digium.com/view/asterisk/tags/1.4.12/ChangeLog?view=markup
thnx,
-baji.
--
On 10/7/07, I wrote:
hello,
running asterisk 1.4.11 on CentOS 4.5
I am getting no response on function STRPTIME() the system just hangs,
STRFTIME() is working fine as seen below. Same thing happens whether
I called in from a softphone or via teliax.
While executing the following code :
;
exten = s,n,Set(v_ts=)
exten = s,n,Set(v_ts=${STRFTIME(|America/New_York|%Y-%m-%d)})
exten = s,n,NoOp(${v_ts})
;
exten = s,n,Set(v_ts=)
exten = s,n,Set(v_ts=${STRPTIME(2007-10-06
05:04:03|America/New_York|%Y-%m-%d %H:%M:%S)})
exten = s,n,NoOp(${v_ts})
;
I get the output :
-- Executing [ [EMAIL PROTECTED]:6] Set(IAX2/4883-1, v_ts=) in new
stack
-- Executing [EMAIL PROTECTED]:7] Set(IAX2/4883-1, v_ts=2007-10-07) in
new stack
-- Executing [EMAIL PROTECTED]:8] NoOp(IAX2/4883-1, 2007-10-07) in new
stack
-- Executing [EMAIL PROTECTED]:9] Set(IAX2/4883-1, v_ts=) in new
stack
If this is a reported bug that has been fixed in 1.4.12, I can upgrade
to it,
but I'd like to know.
tia.
-baji.
--
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Book to learn SIP
On 10/7/07, Justin Case wrote: Hi List, I am trying to learn SIP in its entirety. I have so far found: http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403 Anyone know of any other books that are worth reading ? http://www.geocities.com/intro_to_multimedia/books.html and grab the RFC 3261 http://www.faqs.org/rfcs/rfc3261.html as well as the source in asterisk (1.4.11 here) asterisk-1.4.11/channels/chan_sip.c I would guess that reading the RFC and understanding the implementation in Asterisk will take you to places few others have been to...good luck ! -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Book to learn SIP
Telling someone to read the RFC bah.. might as well give them a blanket and pillow because they will fall asleep. chan_sip is just ugly in every way. /b On Oct 7, 2007, at 9:26 PM, Baji Panchumarti wrote: http://www.faqs.org/rfcs/rfc3261.html as well as the source in asterisk (1.4.11 here) asterisk-1.4.11/channels/chan_sip.c ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Book to learn SIP
Now wait a minute, I favor books and bookmarks myself, but the young man seems determined to conquer the kingdom of SIP, so we told him where the princess is hiding :-) Maybe he has the fire of a dragon in him and just sipping SIP wont do. -baji. -- On 10/7/07, Brian West wrote: Telling someone to read the RFC bah.. might as well give them a blanket and pillow because they will fall asleep. chan_sip is just ugly in every way. /b On Oct 7, 2007, at 9:26 PM, Baji Panchumarti wrote: http://www.faqs.org/rfcs/rfc3261.html as well as the source in asterisk (1.4.11 here) asterisk-1.4.11/channels/chan_sip.c -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good Documentation...
Hello, I'm new to this list, but I've been hacking around on Asterisk for a few months now. I'm getting ready to transition 3 small businesses that I own to a fully VOIP system with a IAX trunk (no POTS lines) but I've hit some snags and I'm looking for some good documentation. I've read Asterisk TFOT and Voip-info.org and those resources have been invaluable for working through a lot of things, but I've also found that a lot of the documentation (even the actual docs that ship with Asterisk) seem to be quite old/outdated. That said, I was wondering if there is some more current resource that I'm missing. Any help would be appreciated! -- George Qualley IV D.J. G Mint Thunder Lightning Sound and Lighting Voice - (515) 255-3698 • Fax - (515) 309-1910 www.MidwestDJs.com • www.ThunderAndLightning.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
