[Asterisk-Users] problem to reach sip client which is connected to asterisk when t he call coming from another sip server.
Hi, I have set-up following set-up. The sip clients is connected to the asterisk and will also be registrar in the asterisk. The asterisk is register like a client to our sip server with same user names that the clients have. When I tried to call on of the sip-clients, the asterisk answer the call instead of forwarding the call. Is it somebody that knows how I should set-up the configuration so the asterisk forwarding the call? Should I registrar the asterisk like a peer to our sip server instead of a client? / Niklas ___ 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] [OT^2]: Getting at the fan on an IBM Thinkpad 600E Laptop?
I'm doing a tutorial at Astricon and the plan is to use my laptop as a demo server. Today it failed to boot and after a bit of sleuthing it turns out the fan is sticking from time to time on bootup. Apparently there is a sensor and if no spin, no go. Moving everything right now would be draconian, yet I can't take a chance on a no-boot while at the show. Does anyone know how bad a job it is to dig into one of these units and clean up/oil/whatever the fan? Thx, and sorry for the OT. B. ___ 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] English vs American voice files
I've spent the afternoon recording all the files for the English speaking VM etc. I've parked the file here http://www.g7ltt.com/VoIP/vmukmale.tgz I did it with Audacity at 44.1KHz x 16bit and thenused sox to raise the levels to -3db and then again to down sample them into 8KHz GSM files. The few that I've listened to sound fine. Hi Mark, If you're going to publish these for public use it would be great if you could make them available in two versions, both the Asterisk 'standard' .gsm format, but also either in 8KHz/8bit/alaw raw or wav and/or 32Kbit ADPCM format - these do give a noticable increase in quality for local/PSTN users of telephony applications over GSM format. Either that, or if you could make the original 44.1K 16bit masters available so others could create the alternatives. Unfortunatly *'s ability to play these cleanly seems a bit broken at the moment, but at least we'll have them for when its fixed! Linus ___ 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] RE: [Asterisk-Dev] Hardware details for the Digium TDM400P
[EMAIL PROTECTED] wrote: I have a DSP based system that is working on a four port FXS system using a 200MHz arm processor. Well.. since we are talking about this topic I owe you guys notes of my experience with SC1100 CPU used by various boards (www.soekris.com , www.pcengines.ch etc.). We made a Linux distro and compacted it into 32MB flash. Installed asterisk and PBXware (http://www.bicomsystems.com/products/C/SC/319/131/ ) onto it. All systems working great. What we have not done yet, is to test X100P/TDM400 and actual number of simultaneous channels using other user agents. This test is planned is to be preformed in next few days just after we come back from astricon. I will update you with the results. SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT^2]: Getting at the fan on an IBM Thinkpad 600E Laptop?
On Sun, Sep 19, 2004 at 02:53:52AM -0500, Brian Capouch wrote: Moving everything right now would be draconian, yet I can't take a chance on a no-boot while at the show. Does anyone know how bad a job it is to dig into one of these units and clean up/oil/whatever the fan? Not off hand but IBM has maintence manuals with complete dis-assembly instructions on their site. Normally on laptops getting at the fan isn't that hard but if it needs replacing rather than just a bit of oil you might find getting the correct part for a reasonable price a bit of a challange. -- Ray ___ 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] Is Dual CPU machine solution for using Asterisk with other general apps (like home automation, web server, ...) in home environment ?
Hi, I'm currently thinking of putting more functionalities to Linux server box. Major is Asterisk, but would also like to add video surveillance, home automation and limited (for only domestic up to 4 users) web, file and mailserver apps. I know there are problems running Asterisk with other such apps on same machine - but wonder if I can take advantage of dual processor machine in this situation (where one CPU would run Asterisk, and other for all other non-critical apps) ? Can P4 with hyperthreading help ? How good is Linux support for dual CPU or hyperthread technologies ? Any experience, advice, more info or pointers for further exploration for this situation ? Thanks in advance, Regards, Robert. ___ 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] Is Dual CPU machine solution for using Asterisk with other general apps (like home automation, web server, ...) in home environment ?
I'm currently thinking of putting more functionalities to Linux server box. Major is Asterisk, but would also like to add video surveillance, home automation and limited (for only domestic up to 4 users) web, file and mailserver apps. I know there are problems running Asterisk with other such apps on same machine - but wonder if I can take advantage of dual processor machine in this situation (where one CPU would run Asterisk, and other for all other non-critical apps) ? Can P4 with hyperthreading help ? How good is Linux support for dual CPU or hyperthread technologies ? Any experience, advice, more info or pointers for further exploration for this situation ? Running other apps on the machine is not a problem at all. It all depends 100% upon how the apps are actually used and not on the fact they are running. In other words, * doesn't consume any significant cycles when no calls are in progress; apache doesn't consume anything if no one is hitting pages; etc, etc. The only way to know whether its going to function for sure is either to know/understand how processor intensive your apps are (including *), or, try and evaluate it. If * is only used for call setup (eg, no transcoding, no digium cards), then cycles are basically only used during the short duration call setup process. If * is expected to handle multiple codecs (eg, trans- code) and you have one or more digium cards installed (that require interrupt servicing), obviously the processor is more heavily loaded and the issue becomes 'how many simultanous calls is it expected to support'. Likewise, if your home surveillance is configured to handle full motion streaming video with storage, that's an entirely different load then is storing a 32k jpg once per minute. Same with mailserver; if you subscribe to ten asterisk-type lists with 200+ postings per day each, that's very different then a home system receiving ten emails per day. Lots of folks have implemented * on 200 mhz systems (and smaller), but very few have actually load tested their systems to know where the cutoff is before echo cancellation (as one example only) is impacted. Only you can guess at the load given the mix of apps that you're expecting. ___ 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] X100p on VIA EPIA-V problems
Hi All, I hope I'm posting this to the appriopriate list, and that cross posting to two lists is OK. (If not, I'm sure I'll hear about it quickly :)) I'm running Asterisk on my (new) VIA EPIA-V motherboard. This seems to be the ideal platform for a home version of asterisk - its small, quiet, low power, and should have plenty of computing horsepower if only it would work! I'm running Redhat 9, Kernel 2.4.20-8, and Asterisk from CVS-HEAD-07/07/04-21:01:10 The phones in my house talk to asterisk via a Sipura SPA-2000. I have a X100p card (not from Digium, I regret, but one of the OEM cards sold by Diginetworks). Here' is a snipit from the boot log: Sep 14 06:32:18 anchor kernel: Zapata Telephony Interface Registered on major 196 Sep 14 06:32:18 anchor zaptel: Loading zaptel framework: succeeded Sep 14 06:32:19 anchor kernel: wcfxo: DAA mode is 'FCC' Sep 14 06:32:19 anchor kernel: Found a Wildcard FXO: Wildcard X101P Sep 14 06:32:20 anchor kernel: usb.c: registered new driver wcusb Sep 14 06:32:20 anchor kernel: Wildcard USB FXS Interface driver registered Sep 14 06:32:21 anchor kernel: Registered tone zone 0 (United States / North America) Sep 14 06:32:21 anchor zaptel: Running ztcfg: succeeded Everything works perfectly, except for the following problem: Sporadically -- about once in 6 hrs, Asterisk reports a Red Alarm from the X100p. Thereafter, the X100p no longer works -- no outgoing calls can be placed; no incoming calls answered. The problem can be cleared in one of two ways - the phone line can be unplugged and plugged back in again, or asterisk can be shut down, the zaptel module uloaded from the kernel, and then reloaded. Here is another snipit from the log: Sep 13 05:40:50 NOTICE[16384]: registered database handle 'mysql1' dsn-[MySQL-asterisk] Sep 13 05:40:50 NOTICE[16384]: registered database handle 'mysql2' dsn-[MySQL-asterisk] Sep 13 05:40:50 NOTICE[16384]: res_odbc loaded. Sep 13 05:40:50 NOTICE[16384]: Registered Config Engine odbc Sep 13 05:40:50 NOTICE[16384]: res_config_odbc loaded. Sep 13 05:40:53 WARNING[16384]: Unable to get our IP address, Skinny disabled Sep 13 05:40:53 WARNING[16384]: Unable to open /dev/dsp: No such device Sep 13 07:48:14 NOTICE[98311]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.93' Sep 13 08:30:29 NOTICE[98311]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.93' Sep 13 08:30:29 NOTICE[98311]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.93' Sep 13 08:33:11 NOTICE[262160]: Unable to create channel of type 'SIP' Sep 13 08:33:11 NOTICE[262160]: Unable to create channel of type 'SIP' Sep 13 08:36:39 WARNING[180236]: Detected alarm on channel 1: Red AlarmSep 13 05:40:50 NOTICE[16384]: registered database handle 'mysql1' dsn-[MySQL-asterisk] Also, from time to time something goes wrong (maybe one call in 5) while a call is in progress, and instead of hearing the person's voice, one hears garbage sounds -- odd tones and pops. The X100p worked reliably for several weeks on my old dual processor 400Mhz pentium box running the same Redhat kernel and Asterisk source. The Via box is otherwise completely stable. I've tried just about every conievable BIOS setting on the Via box, I've recompiled asterisk, zaptel and the kernel specifying i386 as the archetecture. I've run memtest86 in the hope of finding a memory problem. I've tried a 2.6 kernel (2.6.6 with special via patches) with exactly the same results. Has anyone else out there got this configuration to work? My choices now seem to be 1) Buy a new X100p (from Digium this time!) 2) Buy a new motherboard - (I'd like to keep the low-power, low noise and mini-itx form factor) 3) Give up on the X100p and get a Sipura SPA-3000 for an FXO port - I'd loose the timer, so music-on-hold might not work so well Anyone have any other suggestions? Other things I might try? Ways I could go about debugging this? Here is what lspci says about the installed pci devices: lspci -vvv 00:00.0 Host bridge: VIA Technologies, Inc. VT8601 [Apollo ProMedia] (rev 05) Subsystem: VIA Technologies, Inc.: Unknown device aa03 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbort- MAbort+ SERR- PERR+ Latency: 8 Region 0: Memory at e000 (32-bit, prefetchable) [size=64M] Capabilities: [a0] AGP version 2.0 Status: RQ=7 SBA+ 64bit- FW- Rate=x1,x2,x4 Command: RQ=0 SBA- AGP- 64bit- FW- Rate=none 00:01.0 PCI bridge: VIA Technologies, Inc. VT8601 [Apollo ProMedia AGP] (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz+ UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort+ SERR- PERR+ Latency: 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0
RE: [Asterisk-Users] uk caller id
Graham Turner [EMAIL PROTECTED] wrote: dear all, i am looking to enable CALLERID on an Asterisk system comprising a X101P FXO interface connecting to BT PSTN in the uk seems this is supported by the interface but there seems to be varying information on how to enable it in zapata.conf 1. usecallerid=uk 2. ukcallerid=yes being two of the configuration statements offered The current method is usecallerid = uk. Of course, you need to patch Zaptel and Asterisk first. The ukcallerid = yes was used in an earlier version of the patch. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Re: Zaptel compile error - unresolved symbols
Message: 12 Date: Fri, 17 Sep 2004 20:35:32 -0400 From: Rollo Tomnasi [EMAIL PROTECTED] Subject: [Asterisk-Users] Zaptel compile error - unresolved symbols To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Hello - any help is greatly appreciated. I am trying to compile zaptel on debian 2.4.26-1-386. I have a single X100P card installed. When I run '/usr/src/zaptel/make clean;make install' I get the following: depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/torisa.o depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/zaptel.o /sbin/depmod -a depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/torisa.o depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/zaptel.o [ -f /etc/zaptel.conf ] || install -m 644 zaptel.conf.sample /etc/zaptel.conf When i run depmod -ae: depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/torisa.o depmod: __write_lock_failed depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/zaptel.o depmod: __write_lock_failed depmod: __read_lock_failed Can anyone point me in the right direction? Thanks! make sure you have module versions enabled in your kernel and the correct .config darren ___ 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] English vs American voice files
Erm, didn't think of that. Stupidly I deleted the individual wav files. Not a problem though as I have the 3 master files that I recorded them all into. I'll just have to slice it up again. That'll be a few days as I've got family arriving today. Mark Linus Surguy said: I've spent the afternoon recording all the files for the English speaking VM etc. I've parked the file here http://www.g7ltt.com/VoIP/vmukmale.tgz I did it with Audacity at 44.1KHz x 16bit and thenused sox to raise the levels to -3db and then again to down sample them into 8KHz GSM files. The few that I've listened to sound fine. Hi Mark, If you're going to publish these for public use it would be great if you could make them available in two versions, both the Asterisk 'standard' .gsm format, but also either in 8KHz/8bit/alaw raw or wav and/or 32Kbit ADPCM format - these do give a noticable increase in quality for local/PSTN users of telephony applications over GSM format. Either that, or if you could make the original 44.1K 16bit masters available so others could create the alternatives. Unfortunatly *'s ability to play these cleanly seems a bit broken at the moment, but at least we'll have them for when its fixed! Linus ___ 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 -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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
Re: [Asterisk-Users] Agents and Queues
On Fri, Sep 17, 2004 at 11:09:49AM -0500, Paul Traue, Jr. wrote: I've just installed asterisk as a new phone system for our office but am having difficulty with the queues. Specifically I need a way to redirect our sales queue to voicemail when no one is logged in to the queue. I see I can use the joinonempty=no setting, however this setting doesn't work if you use the agent functionality (at least not with AgentCallbackLogin). I could, of course use the AddQueueMember/RemoveQueueMember, however my experience with our version (as well as several previous versions) is once an extension is ringing, it will continue ringing that same extension forever (tried for 5-10 minutes). Can anyone think of a way to accomplish what I want without using the Queue timeout parameter (when someone's logged in and taking phone calls, calls need to stay in the queue)? CLEANUP Paul Paul, I have run into the same problem. The easiest way to fix it would be to not use Agents anymore, as when they are members of a queue there is no way for app_queue to see if they are logged in or not (leading to the problem we're having). However, this bug: http://bugs.digium.com/bug_view_page.php?bug_id=0001693 may provide the tools we need to work around this problem, namely, not putting Agents statically into queues.conf, but adding them dynamically using AddQueueMember when they log on. Austin ___ 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] openh323 compile for Asterisk
HI, I have the latest RC2 of Asterisk on a RH 9 non-modified-load box. I have an Avaya IP phone that uses h323, so I am trying to compile h323 into Asterisk. Now, I downloaded pwlib and openh323 tar files and I have compiled this according to the instructions: pwlib: ./configure make opt openh323: ./configure make opt cd asterisk/channels/h323 make cd asterisk make clean make install I am getting an error when I start asterisk with the -cccg that it can't find the libpt_linux_x86_r.so.1.5.2 when it tries to load the h323 channel. I have verified that the file does exists in the he pwlib/lib directory and with a size. I have the path to this directory in my profiles PATH and I included the path in my ld.so.conf file. I am missing something but what? TIA, Trevor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100p on VIA EPIA-V problems
Why is wsusb loading? The X101P uses the wcfxo module. Lyle - Original Message - From: Andy [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 7:29 AM Subject: [Asterisk-Users] X100p on VIA EPIA-V problems Hi All, I hope I'm posting this to the appriopriate list, and that cross posting to two lists is OK. (If not, I'm sure I'll hear about it quickly :)) I'm running Asterisk on my (new) VIA EPIA-V motherboard. This seems to be the ideal platform for a home version of asterisk - its small, quiet, low power, and should have plenty of computing horsepower if only it would work! I'm running Redhat 9, Kernel 2.4.20-8, and Asterisk from CVS-HEAD-07/07/04-21:01:10 The phones in my house talk to asterisk via a Sipura SPA-2000. I have a X100p card (not from Digium, I regret, but one of the OEM cards sold by Diginetworks). Here' is a snipit from the boot log: Sep 14 06:32:18 anchor kernel: Zapata Telephony Interface Registered on major 196 Sep 14 06:32:18 anchor zaptel: Loading zaptel framework: succeeded Sep 14 06:32:19 anchor kernel: wcfxo: DAA mode is 'FCC' Sep 14 06:32:19 anchor kernel: Found a Wildcard FXO: Wildcard X101P Sep 14 06:32:20 anchor kernel: usb.c: registered new driver wcusb Sep 14 06:32:20 anchor kernel: Wildcard USB FXS Interface driver registered Sep 14 06:32:21 anchor kernel: Registered tone zone 0 (United States / North America) Sep 14 06:32:21 anchor zaptel: Running ztcfg: succeeded Everything works perfectly, except for the following problem: Sporadically -- about once in 6 hrs, Asterisk reports a Red Alarm from the X100p. Thereafter, the X100p no longer works -- no outgoing calls can be placed; no incoming calls answered. The problem can be cleared in one of two ways - the phone line can be unplugged and plugged back in again, or asterisk can be shut down, the zaptel module uloaded from the kernel, and then reloaded. Here is another snipit from the log: Sep 13 05:40:50 NOTICE[16384]: registered database handle 'mysql1' dsn-[MySQL-asterisk] Sep 13 05:40:50 NOTICE[16384]: registered database handle 'mysql2' dsn-[MySQL-asterisk] Sep 13 05:40:50 NOTICE[16384]: res_odbc loaded. Sep 13 05:40:50 NOTICE[16384]: Registered Config Engine odbc Sep 13 05:40:50 NOTICE[16384]: res_config_odbc loaded. Sep 13 05:40:53 WARNING[16384]: Unable to get our IP address, Skinny disabled Sep 13 05:40:53 WARNING[16384]: Unable to open /dev/dsp: No such device Sep 13 07:48:14 NOTICE[98311]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.93' Sep 13 08:30:29 NOTICE[98311]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.93' Sep 13 08:30:29 NOTICE[98311]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.93' Sep 13 08:33:11 NOTICE[262160]: Unable to create channel of type 'SIP' Sep 13 08:33:11 NOTICE[262160]: Unable to create channel of type 'SIP' Sep 13 08:36:39 WARNING[180236]: Detected alarm on channel 1: Red AlarmSep 13 05:40:50 NOTICE[16384]: registered database handle 'mysql1' dsn-[MySQL-asterisk] Also, from time to time something goes wrong (maybe one call in 5) while a call is in progress, and instead of hearing the person's voice, one hears garbage sounds -- odd tones and pops. The X100p worked reliably for several weeks on my old dual processor 400Mhz pentium box running the same Redhat kernel and Asterisk source. The Via box is otherwise completely stable. I've tried just about every conievable BIOS setting on the Via box, I've recompiled asterisk, zaptel and the kernel specifying i386 as the archetecture. I've run memtest86 in the hope of finding a memory problem. I've tried a 2.6 kernel (2.6.6 with special via patches) with exactly the same results. Has anyone else out there got this configuration to work? My choices now seem to be 1) Buy a new X100p (from Digium this time!) 2) Buy a new motherboard - (I'd like to keep the low-power, low noise and mini-itx form factor) 3) Give up on the X100p and get a Sipura SPA-3000 for an FXO port - I'd loose the timer, so music-on-hold might not work so well Anyone have any other suggestions? Other things I might try? Ways I could go about debugging this? Here is what lspci says about the installed pci devices: lspci -vvv 00:00.0 Host bridge: VIA Technologies, Inc. VT8601 [Apollo ProMedia] (rev 05) Subsystem: VIA Technologies, Inc.: Unknown device aa03 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbort- MAbort+ SERR- PERR+ Latency: 8 Region 0: Memory at e000 (32-bit, prefetchable) [size=64M] Capabilities: [a0] AGP version 2.0 Status: RQ=7 SBA+ 64bit- FW- Rate=x1,x2,x4 Command: RQ=0 SBA- AGP- 64bit- FW- Rate=none 00:01.0 PCI bridge: VIA Technologies, Inc. VT8601 [Apollo ProMedia AGP] (prog-if 00 [Normal decode])
Re: [Asterisk-Users] uk caller id
Kevin, thanks for post reply . i have installed asterisk / zaptel from cvs distribution as of 17/09/04 so i assume this does it have configured zapata.conf as per instruction but i would have expected to have seen the callerid on the asterisk console as it receives the call but then may be not ?? the relevant my extensions.conf is ; exten = s,1,answer exten = s,2,Dial(SIP/1001|10) it is quite possible that callerid is being seen by * but i would have expected it to have been echoed to the console or at least written to the CDR entries ??? would you have any suggestions as to how to confirm this going a bit further on, the whole point of this exercise is to allow this CALLERID to be displayed on the console of a SIP peer (7940 ip phone) that is defined as an asterisk extension thanks 4 yr help GT - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 1:48 PM Subject: RE: [Asterisk-Users] uk caller id Graham Turner [EMAIL PROTECTED] wrote: dear all, i am looking to enable CALLERID on an Asterisk system comprising a X101P FXO interface connecting to BT PSTN in the uk seems this is supported by the interface but there seems to be varying information on how to enable it in zapata.conf 1. usecallerid=uk 2. ukcallerid=yes being two of the configuration statements offered The current method is usecallerid = uk. Of course, you need to patch Zaptel and Asterisk first. The ukcallerid = yes was used in an earlier version of the patch. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Re: X100p on VIA EPIA-V problems
I paused myself when I saw this. The generic /etc/init.d/zaptel (that you get if you do make config) tries to load wct4xxp, wct1xxp, wcfxo, wcfxs, and wcusb Paring down the list to just wcfxo generates exactly the same problems. Cheers, Andy. Message: 12 Date: Sun, 19 Sep 2004 09:32:37 -0500 From: "Lyle Giese" [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100p on VIA EPIA-V problems To: "Asterisk Users Mailing List - Non-Commercial Discussion" [EMAIL PROTECTED] Why is wsusb loading? The X101P uses the wcfxo module. Lyle ___ 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 ATA box for sale by Linksys
Fry's Electronics has a new Linksys 2 line ATA box for sale for $59.99 retail. They have a version with a router for $89.99. We picked the non-router version up and it seems to be a rebadged Sipura SPA-2000. The box has a Vonage service package inside as well, but it does work with other services. The box also has a User Guide meant for end-users that is very well written [no Engrish] and explains the calling features and install well. I imagine that the wholesale price of these ATAs will be very attractive if they are selling for $60 retail! ___ 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] ASTCC Help
Hello, I am hitting a brick wall with ASTCC. If someone can help me get a working Calling Card system up and running and give me some explainations on how they did it I will trade them a Grandstream 486 or a BT102 phone in white, your choice (brand new) I want someone who has done it before not just someone who thinks that they can do it. ___ 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] Dial 0 to outbound
? - Original Message - From: Carlos Arnt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 5:14 AM Subject: [Asterisk-Users] Dial 0 to outbound Hi Folks. I see that can put 0 to call out using a x101p (zaptel) or even a pstn service. Thats great, but when press the 0 i just dial then the numbers to call out. There is any way after hit 0 (ear) the line sound ?? I know it's just a style way put some users, really like it !! So after hit 0 to call for example a pstn the user will ear the line sound to dial out. I read lot's of doc's but can't find nothing explaining this method. Thanks alot ! Carlos. ___ 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] Asterisk as an outbound call machine?
I wouldn't trust it to do any real detection. I use the press # mod in 6 sec mod to be able to fwd to other phone #s without risking hitting the answering machine or wrong person. I don't believe there is any real way to detect what you are after as far as if the call is picked up. You would get status for busy and such though. -- William ___ 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] Dial 0 to outbound
If your phone is on a Zap, MGCP, or SCCP interface then look at the ignorepat option in extensions.conf. If your phone is SIP or H323 then this is handled by the phone. Most SIP and H323 phones do not allow you to continue dialtone after dialing a digit. On Sun, 2004-09-19 at 11:28, Steve Totaro wrote: ? - Original Message - From: Carlos Arnt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 5:14 AM Subject: [Asterisk-Users] Dial 0 to outbound Hi Folks. I see that can put 0 to call out using a x101p (zaptel) or even a pstn service. Thats great, but when press the 0 i just dial then the numbers to call out. There is any way after hit 0 (ear) the line sound ?? I know it's just a style way put some users, really like it !! So after hit 0 to call for example a pstn the user will ear the line sound to dial out. I read lot's of doc's but can't find nothing explaining this method. Thanks alot ! -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] Dial 0 to outbound
ignorepat = 9 ; Continue dialtone after dialing 9 if i am reading your question correctly - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 12:28 PM Subject: Re: [Asterisk-Users] Dial 0 to outbound ? - Original Message - From: Carlos Arnt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 5:14 AM Subject: [Asterisk-Users] Dial 0 to outbound Hi Folks. I see that can put 0 to call out using a x101p (zaptel) or even a pstn service. Thats great, but when press the 0 i just dial then the numbers to call out. There is any way after hit 0 (ear) the line sound ?? I know it's just a style way put some users, really like it !! So after hit 0 to call for example a pstn the user will ear the line sound to dial out. I read lot's of doc's but can't find nothing explaining this method. Thanks alot ! Carlos. ___ 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
[Asterisk-Users] New App: WaitForSilence
I just posted this app, which I've been testing for a while and should be ready for inclusion in CVS. http://bugs.digium.com/bug_view_page.php?bug_id=0002467 This app was put together so as to be able to deal with answering machines when making outbound calls. The idea is that you probably don't want to start playing your soundfile until the call has been fully answered by a human, and there has been a given amount of silence. Here is a simple implementation; note that it can be called repeatedly to deal with cadences of different situations: Exten = 7001,1,WaitForSilence(200,7) Exten = 7001,2,WaitForSilence(700,1) Exten = 7001,3,Playback,outboundmsg Exten = 7001,4,Hangup This in (1) waits for silence 200ms, 7 times (1.4 seconds). Then in (2), it waits for silence 700ms once. This deals with the possibility of multiple short periods of silence occuring within or before an answering machine message (200 x 7 instances), and then waits for 700ms of silence to be SURE that the remote side really is ready for you to talk. This essentially emulates human behavior when calling an answering machine, and in my tests, this configuration works very well. This general idea could also be coupled with CDR information (ResetCDR, Answer, etc) to log statistics about termination of calls to humans or answering machines, depending on the amount of post-answer delay experienced while waiting for silence. A longer amount of wait would indicate an answering machine (10 sec?) while a short amount of wait (1-2 sec?) might indicate a human simply saying hello?. You could make an accurate distinction based on that. I leave that as an exercise for the reader. :) This code is based on *'s internal DSP functions. It should be ready for inclusion in the CVS head. Just edit apps/Makefile to include app_waitforsilence.c in app list. This app can be used for good or for evil; please use only for good. Dave Troy ___ 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] First time asterisk installation problem
OL Hi all, OL I am trying to install asterisk on my system, the compiplation and OL installation process all seem to work fine (make ; make install ; make OL samples). OL But astersik fails to start. Is the sample configs not supposed to OL work out of the box? OL Even more confusing, it seems to fail at different points every time I OL start it, but this is probobly because of threads starting differently OL or something? OL I can't really figure out exactly what it is that makes it fail, if OL anyone can give me a clue I would appreciate it. OL Startup log follows below. OL /Ola [See orignal post for log] An update on my progress... I have started to look at the source to see why asterisk fails to start, and so far my conclusion is that the asterisk process gets a SIGINT signal sent to it from some process. Is this normal behaviour if something fails in the startup process? /Ola ___ 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] No sound from IVR scripts, yet calls placed without any problem.
Matt Riddell: This is really unlikely but is it possible he has an internal firewall or something and the Asterisk box is in the DMZ? Sadly not. That's the only thing I could reasonably expect to be causing the problems, but they're on the same LAN in the same network and there's no firewall anywhere between them. :-( 1) Power is different (Extremely unlikely and PC wouldn't work) Indeed. 2) LAN is different (maybe left over setting from ghost i.e bindaddr etc) Another good idea, but everything's on a block of real addresses which I had moved over to his connection, so nothing should change there either. Maybe a NAT problem? I.E. rtp being blocked? I'd agree that it looks like RTP is being blocked, but there's no NAT involved and it's starting to get really really annoying now! What protocol are the calls? It seems to happen irrespective of the codec (does the same with uLAW, aLAW and GSM). 3) VOIP accounts are different Well, this is the interesting thing - the phone (SNOM 200) and asterisk box haven't had anything changed since they were working here (well, apart from the extra HFC card, that is). Thanks for your comments and suggestions though - keep them coming, there's sure to be something I missed. I think I may just recompile * and try it all from scratch again. Hmmm. Nick Barnes Senior IT Consultant. ___ 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] Re: [Asterisk-Dev] X100p on VIA EPIA-V problems
I'm running Asterisk on my (new) VIA EPIA-V motherboard. This seems to be the ideal platform for a home version of asterisk - its small, quiet, low power, and should have plenty of computing horsepower if only it would work! I'm running Redhat 9, Kernel 2.4.20-8, and Asterisk from CVS-HEAD-07/07/04-21:01:10 The phones in my house talk to asterisk via a Sipura SPA-2000. I have a X100p card (not from Digium, I regret, but one of the OEM cards sold by Diginetworks). Here' is a snipit from the boot log: I've been told very recently by a self-proclaimed linux expert (who happens to be involved with selling systems and motherboards, including the VIA) the VIA boards have a terrible PCI bus implementation that has caused lots of problems. The 'expert' has been involved with linux for years, is involved rather heavily in various audio apps, but has zero experience with asterisk. I don't have any experience at all with the VIA, so have no factual knowledge or experience. Simply passing on what I was told when I talked to him about a replacement motherboard. ___ 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] Timing source on SMP system - Disable RTC for zaprtc
Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Thanks, Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Bielicki Sent: Saturday, September 18, 2004 9:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Timing source on SMP system try zaprtc from www.junghanns.net. Works fine in my SMP systems - Original Message - From: Chad Brown [EMAIL PROTECTED] Date: Sat, 18 Sep 2004 20:23:54 -0700 Subject: [Asterisk-Users] Timing source on SMP system To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. I'm sure the DL360 G2 has a OHBI rather than UHBI controller. That said, I got this message during modprobe ztdummy: [EMAIL PROTECTED] zaptel]# modprobe ztdummy /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed I then moved on to zaprtc. However, I was told that this solution will not work with SMP systems. My DL360G2 is a dual proc machine. I'm running out of options here...please advise. Thanks, Chad M. Brown Infrastructure Architect identity mine, inc. - http://www.identitymine.com [EMAIL PROTECTED] 253.927.7737 - Office 866.4ID.MINE (866.443.6463) - Toll free 253.405.6726 - Cellular 253.444.5170 - Fax ___ 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 -- Michael Bielicki ___ 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] Re: [Asterisk-Dev] Hardware details for the Digium TDM400P
On 19/09/2004 16:12 Senad Jordanovic said the following: Well.. since we are talking about this topic I owe you guys notes of my experience with SC1100 CPU used by various boards (www.soekris.com , www.pcengines.ch etc.). i'd be eagerly awaiting these results. i've tested a 16MB image of asterisk/picobsd on a soekris net4511 (100Mhz AMD Elan SC520, 64MB RAM) and it handles up to 8 simultaneous SIP calls very well as long as it does not have to do any transcoding (ulaw on SIP and the IAX transfer). with transcoding (ulaw - gsm), it fscks up on the second call onwards with both calls sounding very, very robotic. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new ATA box for sale by Linksys
Please explain how you got the PAP2 to work with another carrier? I spent over an hour on the phone with 3 levels of Linksys support staff and 2 levels of Vonage staff telling me that the PAP2 CAN NOT be used on any other service except vonage because they burn the vonage information into the firmware. Please explain... Matthew Matthew: When the PAP2 was first available, it was only sold as a Vonage locked version (the same one that I and it sounds like you got...got nowhere with it). Since then Linksys has released the PAP2-NA which is not locked to any particular service provider. The part number is the key... Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new ATA box for sale by Linksys
I had 2 senior level management people at linksys corp confirm that this would not be possible until December. They both told me that they are currently in development of a 'non-locked' version but that it would not be in stores until December. Did you find these PAP2-NA at Fry's as well? Online somewhere? Thanks, Matthew - Original Message - From: Marty Mastera [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 1:27 PM Subject: RE: [Asterisk-Users] new ATA box for sale by Linksys Please explain how you got the PAP2 to work with another carrier? I spent over an hour on the phone with 3 levels of Linksys support staff and 2 levels of Vonage staff telling me that the PAP2 CAN NOT be used on any other service except vonage because they burn the vonage information into the firmware. Please explain... Matthew Matthew: When the PAP2 was first available, it was only sold as a Vonage locked version (the same one that I and it sounds like you got...got nowhere with it). Since then Linksys has released the PAP2-NA which is not locked to any particular service provider. The part number is the key... Marty ___ 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] Timing source on SMP system - Disable RTC forzaprtc
It was my understanding that you don't 'disable' rtc, but recompile it as a kernel module. Again, just my understanding as I can't try it until monday. Matthew - Original Message - From: Chad Brown [EMAIL PROTECTED] To: Michael Bielicki [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 1:13 PM Subject: RE: [Asterisk-Users] Timing source on SMP system - Disable RTC forzaprtc Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Thanks, Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Bielicki Sent: Saturday, September 18, 2004 9:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Timing source on SMP system try zaprtc from www.junghanns.net. Works fine in my SMP systems - Original Message - From: Chad Brown [EMAIL PROTECTED] Date: Sat, 18 Sep 2004 20:23:54 -0700 Subject: [Asterisk-Users] Timing source on SMP system To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. I'm sure the DL360 G2 has a OHBI rather than UHBI controller. That said, I got this message during modprobe ztdummy: [EMAIL PROTECTED] zaptel]# modprobe ztdummy /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed I then moved on to zaprtc. However, I was told that this solution will not work with SMP systems. My DL360G2 is a dual proc machine. I'm running out of options here...please advise. Thanks, Chad M. Brown Infrastructure Architect identity mine, inc. - http://www.identitymine.com [EMAIL PROTECTED] 253.927.7737 - Office 866.4ID.MINE (866.443.6463) - Toll free 253.405.6726 - Cellular 253.444.5170 - Fax ___ 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 -- Michael Bielicki ___ 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
Re: [Asterisk-Users] new ATA box for sale by Linksys
Matthew Boehm wrote: I had 2 senior level management people at linksys corp confirm that this would not be possible until December. They both told me that they are currently in development of a 'non-locked' version but that it would not be in stores until December. Did you find these PAP2-NA at Fry's as well? Online somewhere? I just googled and found several sites that offer the NA version. On one I have up before me right now (costcentral.com) the description says, 2PT PHONE ADAPTER VOIP GENERIC VERSION I would think that would indicate that it's not the Vonage-locked version. I was just about to buy a couple of them, but now you've got me scared :-) B. ___ 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] Timing source on SMP system - Disable RTCforzaprtc
Any help would be appreciated as I am a novice trying to work around a difficult situation. This is what the zaprtc helpfile says: zaprtc, getting zaptel timing out of your realtime clock Make sure that you _dont_ have rtc support compiled into your kernel! INSTALL: make USE: make load REMOVE: make unload I interpreted this as disabling support for RTC in the current kernel and loading new support. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Sunday, September 19, 2004 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Timing source on SMP system - Disable RTCforzaprtc It was my understanding that you don't 'disable' rtc, but recompile it as a kernel module. Again, just my understanding as I can't try it until monday. Matthew - Original Message - From: Chad Brown [EMAIL PROTECTED] To: Michael Bielicki [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 1:13 PM Subject: RE: [Asterisk-Users] Timing source on SMP system - Disable RTC forzaprtc Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Thanks, Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Bielicki Sent: Saturday, September 18, 2004 9:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Timing source on SMP system try zaprtc from www.junghanns.net. Works fine in my SMP systems - Original Message - From: Chad Brown [EMAIL PROTECTED] Date: Sat, 18 Sep 2004 20:23:54 -0700 Subject: [Asterisk-Users] Timing source on SMP system To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. I'm sure the DL360 G2 has a OHBI rather than UHBI controller. That said, I got this message during modprobe ztdummy: [EMAIL PROTECTED] zaptel]# modprobe ztdummy /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed I then moved on to zaprtc. However, I was told that this solution will not work with SMP systems. My DL360G2 is a dual proc machine. I'm running out of options here...please advise. Thanks, Chad M. Brown Infrastructure Architect identity mine, inc. - http://www.identitymine.com [EMAIL PROTECTED] 253.927.7737 - Office 866.4ID.MINE (866.443.6463) - Toll free 253.405.6726 - Cellular 253.444.5170 - Fax ___ 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 -- Michael Bielicki ___ 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 ___ 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] Timing source on SMP system - Disable RTCforzaprtc
On Sun, 2004-09-19 at 11:58 -0700, Chad Brown wrote: Any help would be appreciated as I am a novice trying to work around a difficult situation. This is what the zaprtc helpfile says: zaprtc, getting zaptel timing out of your realtime clock Make sure that you _dont_ have rtc support compiled into your kernel! Compiled in means as part of the kernel, you compile it as a module and then use zaprtc instead. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new ATA box for sale by Linksys
I had 2 senior level management people at linksys corp confirm that this would not be possible until December. They both told me that they are currently in development of a 'non-locked' version but that it would not be in stores until December. Did you find these PAP2-NA at Fry's as well? Online somewhere? Thanks, Matthew Until about three or four days ago, various online vendors listed the PAP2-NA on their websites but none had any in stock...they all showed an ETA of the 14th or 15th (that's from memory)...anyway when I checked on Friday, a handful of vendors show them as in stock. I found them via froogle (http://www.google.com/froogle?q=pap2-na). I haven't received one yet to test, but the vendors that I called to confim that this unit isn't locked swear to that fact...a couple of the vendors that I have seen sell both models (PAP2 and PAP2-NA) and have PDF spec sheets for each, with the PDF for the PAP2 having Vonage's name and logo in it, and tha PAP2-NA having no evidence of Vonage listed in it... Marty ___ 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] Timing source on SMP system - DisableRTCforzaprtc
I have had no success with a stock redhat install. The removal of rtc went fine, but I cannot compile the zaprtc. I posted the compile output a while back, also having NO rtc in my kernel and being an smp system, is this a problem ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: 19 September 2004 20:06 To: Asterisk List Subject: RE: [Asterisk-Users] Timing source on SMP system - DisableRTCforzaprtc On Sun, 2004-09-19 at 11:58 -0700, Chad Brown wrote: Any help would be appreciated as I am a novice trying to work around a difficult situation. This is what the zaprtc helpfile says: zaprtc, getting zaptel timing out of your realtime clock Make sure that you _dont_ have rtc support compiled into your kernel! Compiled in means as part of the kernel, you compile it as a module and then use zaprtc instead. -- Dave Cotton [EMAIL PROTECTED] ___ 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] new ATA box for sale by Linksys
Does anyone have one of these models? Can they confirm that it works with any other SIP server? How is the PAP2-NA configured? Web? Phone? The pdf I downloaded from the pap2-na page on shopblt.com says Model: PAP2. Thanks, Matthew - Original Message - From: Marty Mastera [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 2:10 PM Subject: RE: [Asterisk-Users] new ATA box for sale by Linksys I had 2 senior level management people at linksys corp confirm that this would not be possible until December. They both told me that they are currently in development of a 'non-locked' version but that it would not be in stores until December. Did you find these PAP2-NA at Fry's as well? Online somewhere? Thanks, Matthew Until about three or four days ago, various online vendors listed the PAP2-NA on their websites but none had any in stock...they all showed an ETA of the 14th or 15th (that's from memory)...anyway when I checked on Friday, a handful of vendors show them as in stock. I found them via froogle (http://www.google.com/froogle?q=pap2-na). I haven't received one yet to test, but the vendors that I called to confim that this unit isn't locked swear to that fact...a couple of the vendors that I have seen sell both models (PAP2 and PAP2-NA) and have PDF spec sheets for each, with the PDF for the PAP2 having Vonage's name and logo in it, and tha PAP2-NA having no evidence of Vonage listed in it... Marty ___ 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] new ATA box for sale by Linksys
Matthew Boehm wrote: I had 2 senior level management people at linksys corp confirm that this would not be possible until December. They both told me that they are currently in development of a 'non-locked' version but that it would not be in stores until December. Those kind of people only know what they are told: http://www.nufone.net/downloads/pap2.jpg Jeremy McNamara ___ 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[2]: [Asterisk-Users] new ATA box for sale by Linksys
Hi, http://www.costcentral.com/searchresult.php?keyword=PAP2searchin=1 Mfg Part # Stock Price -- -- -- PAP2 No $49.86 PAP2-NA Yes $49.76 Best regards, Miroslavmailto:[EMAIL PROTECTED] Sunday, September 19, 2004, 10:32:04 PM, you wrote: JM Matthew Boehm wrote: I had 2 senior level management people at linksys corp confirm that this would not be possible until December. They both told me that they are currently in development of a 'non-locked' version but that it would not be in stores until December. JM Those kind of people only know what they are told: JM http://www.nufone.net/downloads/pap2.jpg JM Jeremy McNamara JM ___ JM Asterisk-Users mailing list JM [EMAIL PROTECTED] JM http://lists.digium.com/mailman/listinfo/asterisk-users JM To UNSUBSCRIBE or update options visit: JMhttp://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] New Zealand Supplier
Does anyone in New Zealand have any ATA devices in stock I.e. Sipura SPA-2000? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new ATA box for sale by Linksys
Jeremy McNamara wrote: Matthew Boehm wrote: I had 2 senior level management people at linksys corp confirm that this would not be possible until December. They both told me that they are currently in development of a 'non-locked' version but that it would not be in stores until December. Those kind of people only know what they are told: http://www.nufone.net/downloads/pap2.jpg Jeremy McNamara ___ 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 Jeremy Hi, From your experience, could you give us the merits and demerits of these ATA devices as well as the IAXy. Thanks, ___ 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] Timing source on SMP system - Disable RTC for zaprtc
Chad Brown wrote: Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Thanks, Chad Chad, Real time clock under character devices in make menuconfig. I just did this on RHEL 3 WS (my laptop), so it should be similar. Also, good luck with this on an SMP box, because many people (on the list and elsewhere) can tell you that it is certainly not an optimal solution. If you have an SMP box, and want to do conferencing, buy some Digium hardware! -- Kristian Kielhofner ___ 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] AB1
On Saturday 18 September 2004 01:09, Matt Hohman wrote: Google? Hey thanks for the info I haven't seen that before. Wonders of modern technology. It's nice to use the list as a round table and get some insight. While I tend to agree with you the AB1's been discussed to death. Google really would have been the correct option here. So they don't have disconnect detection I've heard of people using busy detection is this sufficient am I going to be wishing I paid the extra for the adit 600? I certainly would not rely on busy detection, but that is just my opinion. Use the right hardware for the job. And IMO, the AB1 is not the right hardware for FXO. -A. ___ 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] Re: X100p on VIA EPIA-V
Date: Sun, 19 Sep 2004 12:59:52 -0600 From: Rich Adamson [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: [Asterisk-Dev] X100p on VIA EPIA-V problems To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 I'm running Asterisk on my (new) VIA EPIA-V motherboard. This seems to be the ideal platform for a home version of asterisk - its small, quiet, low power, and should have plenty of computing horsepower if only it would work! I'm running Redhat 9, Kernel 2.4.20-8, and Asterisk from CVS-HEAD-07/07/04-21:01:10 The phones in my house talk to asterisk via a Sipura SPA-2000. I have a X100p card (not from Digium, I regret, but one of the OEM cards sold by Diginetworks). Here' is a snipit from the boot log: I've been told very recently by a self-proclaimed linux expert (who happens to be involved with selling systems and motherboards, including the VIA) the VIA boards have a terrible PCI bus implementation that has caused lots of problems. The 'expert' has been involved with linux for years, is involved rather heavily in various audio apps, but has zero experience with asterisk. I don't have any experience at all with the VIA, so have no factual knowledge or experience. Simply passing on what I was told when I talked to him about a replacement motherboard. I wonder if I am seeing a similar issue, I am debugging a voice quality problem with a voiceronix openline4 on an VIA EPIA V mainboard. I get random tones chirps on calls through the FXO, otherwise it performs flawlessly. Anyone got any info on how to debug PCI issues? darren ___ 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] Using queue app with external members/destinations
Hi guys I've got a need to do some call queueing, with the slightly unusual caveat that the destination for the calls is not a phone or group of phones connected to my local asterisk box, but an external PSTN number. Can I setup a queue in asterisk and make the queue member an external address like SIP/[EMAIL PROTECTED] There will be a smaller number of PSTN lines available at the far end destination than there are inbound calls queueing, so after X number of calls, attempts to call that agent will receive a busy response back until a call in progress is finished and a line becomes available to take the next queued call (does that make sense?) It sounds simple enough, and doable too, I just wanted to check if anyone else had experience and/or thoughts on this kind of setup? Thanks in advance Paul ___ 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] Timing source on SMP system - Disable RTC forzaprtc
Kristian, I have 2 X100P cards but neither work on my Compaq DL360 G2. The system will not even boot! Take a look at my initial post and let me know if you have any other advice. Regardless, thanks for your post! - I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. I'm sure the DL360 G2 has a OHBI rather than UHBI controller. That said, I got this message during modprobe ztdummy: [EMAIL PROTECTED] zaptel]# modprobe ztdummy /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed I then moved on to zaprtc. However, I was told that this solution will not work with SMP systems. My DL360G2 is a dual proc machine. I'm running out of options here...please advise. Thanks, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Sunday, September 19, 2004 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Timing source on SMP system - Disable RTC forzaprtc Chad Brown wrote: Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Thanks, Chad Chad, Real time clock under character devices in make menuconfig. I just did this on RHEL 3 WS (my laptop), so it should be similar. Also, good luck with this on an SMP box, because many people (on the list and elsewhere) can tell you that it is certainly not an optimal solution. If you have an SMP box, and want to do conferencing, buy some Digium hardware! -- Kristian Kielhofner ___ 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] Using queue app with external members/destinations
What you want to do is connect the remote phone number to an internal extension. You can do this in a couple of ways, using the Manager interface and the Connect command. Alternatively, you can create a call file in Asterisk's call spool (usually /var/spool/asterisk or whatever) which has the same makeup as the Connect command. All you do then is specify in your extensions something like the following: [outboundqueue] exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Queue(...) All other queue stuff should work from there. -Ken Shaw... On Sun, 2004-09-19 at 14:42, Paul Crick wrote: Hi guys I've got a need to do some call queueing, with the slightly unusual caveat that the destination for the calls is not a phone or group of phones connected to my local asterisk box, but an external PSTN number. Can I setup a queue in asterisk and make the queue member an external address like SIP/[EMAIL PROTECTED] There will be a smaller number of PSTN lines available at the far end destination than there are inbound calls queueing, so after X number of calls, attempts to call that agent will receive a busy response back until a call in progress is finished and a line becomes available to take the next queued call (does that make sense?) It sounds simple enough, and doable too, I just wanted to check if anyone else had experience and/or thoughts on this kind of setup? Thanks in advance Paul ___ 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] Re: X100p on VIA EPIA-V
I'm running Asterisk on my (new) VIA EPIA-V motherboard. This seems to be the ideal platform for a home version of asterisk - its small, quiet, low power, and should have plenty of computing horsepower if only it would work! I'm running Redhat 9, Kernel 2.4.20-8, and Asterisk from CVS-HEAD-07/07/04-21:01:10 The phones in my house talk to asterisk via a Sipura SPA-2000. I have a X100p card (not from Digium, I regret, but one of the OEM cards sold by Diginetworks). Here' is a snipit from the boot log: I've been told very recently by a self-proclaimed linux expert (who happens to be involved with selling systems and motherboards, including the VIA) the VIA boards have a terrible PCI bus implementation that has caused lots of problems. The 'expert' has been involved with linux for years, is involved rather heavily in various audio apps, but has zero experience with asterisk. I don't have any experience at all with the VIA, so have no factual knowledge or experience. Simply passing on what I was told when I talked to him about a replacement motherboard. I wonder if I am seeing a similar issue, I am debugging a voice quality problem with a voiceronix openline4 on an VIA EPIA V mainboard. I get random tones chirps on calls through the FXO, otherwise it performs flawlessly. Anyone got any info on how to debug PCI issues? My understanding from the 'expert' is the PCI issues have something to do with a poor PCI chip design on the motherboard. The folks that are heavy into audio apps tend to swap out their VIA motherboards. Guess that implies there aren't any workarounds. ___ 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] Medium volume 100% SIP/IAX PBX.
Marconi Rivello [EMAIL PROTECTED] wrote: Hi, I have a curiosity: how much does a regular PBX system cost? I'm curious if using IP telephony in a building is cheaper than a regular PBX, because of the high cost of the IP phones. Take a look at http://www.buyerzone.com/telecom_equipment/phone_systems/buyers_guide7.html. As the article pointed out, TCO is important as well. Commercial PBXs usually require technicians with special software and training that can run US$100/hr. If you already have a sysadmin, the savings in labor alone can pay for an * system over a few years. The prices for cards and software modules can add up quick as well... even if you could get a cheap NEC/Panasonic/Nortel/Avaya for the same cost as an IP-PBX, you'd basically have a featureless switch with featureless telephones. At that point, you might as well call up the local telco and get Centrex service. Figure that very nice IP phone can be had for around $250, and nice * box can be built with dual processors for about $3000 that can handle 30 users. That equates to about $350/user for a somewhat over-provisioned all-VoIP setup. That's far less than buying a $900/user PBX. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100p on VIA EPIA-V problems
On Sun, 19 Sep 2004 07:29:16 -0500, Andy [EMAIL PROTECTED] wrote: Everything works perfectly, except for the following problem: Sporadically -- about once in 6 hrs, Asterisk reports a Red Alarm from the X100p. Thereafter, the X100p no longer works -- no outgoing calls can be placed; no incoming calls answered. The problem can be cleared in one of two ways - the phone line can be unplugged and plugged back in again, or asterisk can be shut down, the zaptel module uloaded from the kernel, and then reloaded. I had the same problem in an Athlon 900Mhz, with a MSI MoBo. Tried Linux 2.4 and 2.6... Then I changed to a dual Athlon MP 1400Mhz on a Tyan Tiger MoBo. Same problem... Then I moved to a Pentium 4 2.8Ghz, (I can check later, but I'm almost sure it's an Asus MB). It's up and running for 5 days without any problems (or module reloading, or reboot). I also replaced the phone wires with newer ones, but I don't believe that was causing the Red Alarm problem. This week, I'll hook up the old cable just to make sure, but I did read other people complaining about problems on AMD based systems, and also reporting 100% ok on Intel based... I'm not happy about this, but life isn't fair. Regards, Marconi. ___ 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 and Red Hat 9
Hi everyone, Im a newbie to Asterisk. Will Asterisk run on RH9, easily or does it have to run on FreeBSD? Will the drivers for the Digium cards work on RH9? ___ 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] Configuration on MySQL or not
Hi I am getting more hands on about Asterisk issues but I got a question to ask. What is the common factor, to put all configurations bind to MySQL or have them as they are originally on text configuration files. Maybe this questions can be out of focus, but it will clear up some ideas in the future All type of comments are welcome Regards John Fach ___ 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] Configuration on MySQL or not
I am working on a proper MySQL iaxfriends now just getting ready to post on bug site Tested and works great It loads all your iaxfriends into registry when you do a reload... So if you add new users to MySQL you have to do a reload -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Linux Dominicana Sent: Sunday, September 19, 2004 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Configuration on MySQL or not Hi I am getting more hands on about Asterisk issues but I got a question to ask. What is the common factor, to put all configurations bind to MySQL or have them as they are originally on text configuration files. Maybe this questions can be out of focus, but it will clear up some ideas in the future All type of comments are welcome Regards John Fach ___ 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] Configuration on MySQL or not
Depends. Do you have daily (sometimes hourly) configuration changes to your *.conf files? I do. Therefor for me its better for the conf files to be stored in database. I've even rewritten some of the * code to pull information out dynamically instead of having to reload each time. If most of your configuration will be static, it might not be worth the trouble and overhead of installing and maintaining and reprogramming asterisk to work better with mysql. My $0.02 Matthew - Original Message - From: Linux Dominicana [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 6:02 PM Subject: [Asterisk-Users] Configuration on MySQL or not Hi I am getting more hands on about Asterisk issues but I got a question to ask. What is the common factor, to put all configurations bind to MySQL or have them as they are originally on text configuration files. Maybe this questions can be out of focus, but it will clear up some ideas in the future All type of comments are welcome Regards John Fach ___ 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] Medium volume 100% SIP/IAX PBX.
On Sun, 19 Sep 2004 15:10:27 -0700 (PDT), Nick Bachmann [EMAIL PROTECTED] wrote: Take a look at http://www.buyerzone.com/telecom_equipment/phone_systems/buyers_guide7.html. As the article pointed out, TCO is important as well. Commercial PBXs usually require technicians with special software and training that can run US$100/hr. If you already have a sysadmin, the savings in labor alone can pay for an * system over a few years. The prices for cards and software modules can add up quick as well... even if you could get a cheap NEC/Panasonic/Nortel/Avaya for the same cost as an IP-PBX, you'd basically have a featureless switch with featureless telephones. At that point, you might as well call up the local telco and get Centrex service. Figure that very nice IP phone can be had for around $250, and nice * box can be built with dual processors for about $3000 that can handle 30 users. That equates to about $350/user for a somewhat over-provisioned all-VoIP setup. That's far less than buying a $900/user PBX. Nick Thank you very much... very illustrative... Marconi. ___ 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] Configuration on MySQL or not
Maybe is not right to ask, but those changes are GPL to the community? If yes, any README to integrate them, or detailed explanation If not, no problem, is nice to know that those changes exist Regards John Fach On Sun, 19 Sep 2004 18:06:56 -0500, Matthew Boehm [EMAIL PROTECTED] wrote: Depends. Do you have daily (sometimes hourly) configuration changes to your *.conf files? I do. Therefor for me its better for the conf files to be stored in database. I've even rewritten some of the * code to pull information out dynamically instead of having to reload each time. If most of your configuration will be static, it might not be worth the trouble and overhead of installing and maintaining and reprogramming asterisk to work better with mysql. My $0.02 Matthew ___ 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] Configuration on MySQL or not
http://bugs.digium.com/bug_view_page.php?bug_id=0002469 New Patch for MySql -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Linux Dominicana Sent: Sunday, September 19, 2004 7:19 PM To: Matthew Boehm Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Configuration on MySQL or not Maybe is not right to ask, but those changes are GPL to the community? If yes, any README to integrate them, or detailed explanation If not, no problem, is nice to know that those changes exist Regards John Fach On Sun, 19 Sep 2004 18:06:56 -0500, Matthew Boehm [EMAIL PROTECTED] wrote: Depends. Do you have daily (sometimes hourly) configuration changes to your *.conf files? I do. Therefor for me its better for the conf files to be stored in database. I've even rewritten some of the * code to pull information out dynamically instead of having to reload each time. If most of your configuration will be static, it might not be worth the trouble and overhead of installing and maintaining and reprogramming asterisk to work better with mysql. My $0.02 Matthew ___ 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] Configuration on MySQL or not
Yes. Mine are GPL to the community. I have already posted 1 such patch to the list. I will post my others once I believe they are stable. Matthew - Original Message - From: Linux Dominicana [EMAIL PROTECTED] To: Matthew Boehm [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 6:19 PM Subject: Re: [Asterisk-Users] Configuration on MySQL or not Maybe is not right to ask, but those changes are GPL to the community? If yes, any README to integrate them, or detailed explanation If not, no problem, is nice to know that those changes exist Regards John Fach On Sun, 19 Sep 2004 18:06:56 -0500, Matthew Boehm [EMAIL PROTECTED] wrote: Depends. Do you have daily (sometimes hourly) configuration changes to your *.conf files? I do. Therefor for me its better for the conf files to be stored in database. I've even rewritten some of the * code to pull information out dynamically instead of having to reload each time. If most of your configuration will be static, it might not be worth the trouble and overhead of installing and maintaining and reprogramming asterisk to work better with mysql. My $0.02 Matthew ___ 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] Configuration on MySQL or not
Have you also posted them at bugs.digium.com ? Thanks Duane Cox - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Linux Dominicana [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 6:44 PM Subject: Re: [Asterisk-Users] Configuration on MySQL or not Yes. Mine are GPL to the community. I have already posted 1 such patch to the list. I will post my others once I believe they are stable. Matthew - Original Message - From: Linux Dominicana [EMAIL PROTECTED] To: Matthew Boehm [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 6:19 PM Subject: Re: [Asterisk-Users] Configuration on MySQL or not Maybe is not right to ask, but those changes are GPL to the community? If yes, any README to integrate them, or detailed explanation If not, no problem, is nice to know that those changes exist Regards John Fach On Sun, 19 Sep 2004 18:06:56 -0500, Matthew Boehm [EMAIL PROTECTED] wrote: Depends. Do you have daily (sometimes hourly) configuration changes to your *.conf files? I do. Therefor for me its better for the conf files to be stored in database. I've even rewritten some of the * code to pull information out dynamically instead of having to reload each time. If most of your configuration will be static, it might not be worth the trouble and overhead of installing and maintaining and reprogramming asterisk to work better with mysql. My $0.02 Matthew ___ 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
Re: [Asterisk-Users] Re: X100p on VIA EPIA-V
Rich Adamson wrote: My understanding from the 'expert' is the PCI issues have something to do with a poor PCI chip design on the motherboard. The folks that are heavy into audio apps tend to swap out their VIA motherboards. Guess that implies there aren't any workarounds. From the Ardour (Linux Audio Editor) System requirements: Avoid VIA motherboards and chipsets wherever possible. This company has demonstrated an almost complete disregard for reasonable use of the PCI bus. Their hardware has repeatedly been implicated in a failure to achieve low latency performance. Here is one example of the kinds of problems you can expect. example points to: http://linux.derkeiler.com/Mailing-Lists/Kernel/2003-09/7761.html Stefan de Konink ___ 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] How does Asterisk interact with an h323 gateway
Hi, I don't know quite how to ask this question, because my knowledge is so limited at this time. I have an h323 phone that I am trying to use to do VOIP to phones on the PSTN. I want to sign up for a service and not have it go out my POTS line. I do have a Quicknet Line jack in my RH 9 box and it is fully confiugred. I have downloaded the latest drive from openh323.org and installed it and the module loads correctly when called. What I am confused about is how Asterisk will interact with an h323 gateway. Will I lose an functionality of the Asterisk PBX because of this. What I am ultimately trying to do is have a heterogeneous group of phones(Cisco, avaya, snome, etc) in my company that can use the Asterisk PBX for all their functionality not matter what protocol or Codec that use. Is this something that is wishful thinking? Also, do I need an h323 gateway for my h323 phone and Asterisk. If so, how does Asterisk use this h323 gateway? I thought Asterisk was the gateway for all calls into and out of the phone system, is this not correct? These questions may be trivial to most of you, but for someone new to this exciting technology it is not. TIA, Trevor ___ 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] No sound from IVR scripts, yet calls placed without any problem.
On 19 Sep 2004 at 18:19, Nick Barnes wrote: Thanks for your comments and suggestions though - keep them coming, there's sure to be something I missed. I think I may just recompile * and try it all from scratch again. Yeah. I'd delete the source and regrab it from CVS. Don't worry about deleting the conf's yet. It's easier just to grab CVS (not that you should need to on a ghosted machine). Does music on hold work? Are the phones SIP? I'm just wondering if maybe the reason you audio between them is that they do a reinvite and actually talk to each other rather than through Asterisk. Maybe also run an NMAP or similar against the box (not from the box - ideally somewhere close to the phones)... Can you make a call via the console? Then you're down to ethereal... nice Cheers, Matt Riddell http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Setting time on ADSI phones from Asterisk
Hi, Would anyone know of a way to set the time automatically on an ADSI capable phone from *? The phone in question is a Aastra 480e. While I am at it, does anyone have any helpful docs on the ADSI script programming? I have managed to do basic functions by modifying the asterisk.adsi file using stuff gleaned from the app_adsiprog.c file, but docs would be really helpful at this point. Tia, Dennis ___ 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] PC Requirements
HI all, I would like to build a 12 CO by 36 phone system with Voicemail, I am trying to decide which machine would be a cost effective solution. Would a Pentium 4 2.6 GHZ with 1G of Ram be suitable? ___ 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] passing octothorpe
some conferencing systems want you to hit octothorpe (aka pound, hash, etc.). once connected, i would have expected * to be transparent to all dtmf codes. it seems not to be. wiki has not been helpful, it seems to have most references to do with octothorpe in dial plan. so, what do i not understand? randy ___ 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: X100p on VIA EPIA-V
Rich Adamson wrote: My understanding from the 'expert' is the PCI issues have something to do with a poor PCI chip design on the motherboard. The folks that are heavy into audio apps tend to swap out their VIA motherboards. Guess that implies there aren't any workarounds. From the Ardour (Linux Audio Editor) System requirements: Avoid VIA motherboards and chipsets wherever possible. This company has demonstrated an almost complete disregard for reasonable use of the PCI bus. Their hardware has repeatedly been implicated in a failure to achieve low latency performance. Here is one example of the kinds of problems you can expect. example points to: http://linux.derkeiler.com/Mailing-Lists/Kernel/2003-09/7761.html Stefan de Konink Guess that pretty much says it all, and with a lot more authority then the individual I was talking to. :) Rich ___ 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] Configuration on MySQL or not
I will post my others once I believe they are stable. Matthew - Original Message - From: Duane Cox [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 7:15 PM Subject: Re: [Asterisk-Users] Configuration on MySQL or not Have you also posted them at bugs.digium.com ? Thanks Duane Cox - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Linux Dominicana [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 6:44 PM Subject: Re: [Asterisk-Users] Configuration on MySQL or not Yes. Mine are GPL to the community. I have already posted 1 such patch to the list. I will post my others once I believe they are stable. Matthew - Original Message - From: Linux Dominicana [EMAIL PROTECTED] To: Matthew Boehm [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 6:19 PM Subject: Re: [Asterisk-Users] Configuration on MySQL or not Maybe is not right to ask, but those changes are GPL to the community? If yes, any README to integrate them, or detailed explanation If not, no problem, is nice to know that those changes exist Regards John Fach On Sun, 19 Sep 2004 18:06:56 -0500, Matthew Boehm [EMAIL PROTECTED] wrote: Depends. Do you have daily (sometimes hourly) configuration changes to your *.conf files? I do. Therefor for me its better for the conf files to be stored in database. I've even rewritten some of the * code to pull information out dynamically instead of having to reload each time. If most of your configuration will be static, it might not be worth the trouble and overhead of installing and maintaining and reprogramming asterisk to work better with mysql. My $0.02 Matthew ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100p on VIA EPIA-V problems
I have the following system setup: ECS KT600-A motherboard Athlon XP 2.6 FSB 333 2x512 MB 333 DDR memory TDM400P (3 FXS, 1 FXO) X101P (Encore MD 3200) modem nVidia MX 4000 Works all right, I have been restarting asterisk about once a week, due to other issues. Here's the lspci: :00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400 AGP] Host Bridge (rev 80) :00:01.0 PCI bridge: VIA Technologies, Inc.: Unknown device b198 :00:09.0 Communication controller: Tiger Jet Network Inc. Intel 537 :00:0a.0 Communication controller: Tiger Jet Network Inc. Intel 537 :00:0f.0 RAID bus controller: VIA Technologies, Inc.: Unknown device 3149 (rev 80) :00:0f.1 IDE interface: VIA Technologies, Inc. VT82C586/B/686A/B PIPC Bus Master IDE (rev 06) :00:10.0 USB Controller: VIA Technologies, Inc. USB (rev 81) :00:10.1 USB Controller: VIA Technologies, Inc. USB (rev 81) :00:10.2 USB Controller: VIA Technologies, Inc. USB (rev 81) :00:10.3 USB Controller: VIA Technologies, Inc. USB (rev 81) :00:10.4 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 86) :00:11.0 ISA bridge: VIA Technologies, Inc.: Unknown device 3227 :00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233 AC97 Audio Controller (rev 60) :00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev 78) :01:00.0 VGA compatible controller: nVidia Corporation: Unknown device 0185 (rev c1) I've been using this box as my asterisk server, workstation, broadband router (sharing with a 768/256 ADSL with 10 LAN users), web server, e-mail server, gift (kazaa for linux) and running Windows games with Wine without any frequent issues. Considering that I'm running a snapshot conectiva linux (currently at 2.6.9 kernel), I'd say Asterisk itself has been rock solid. However Asterisk activity is very load, Asterisk is essentially single-user (myself). There have been sporadic lockups (once or twice a month), but I would atribute that to things other than Asterisk. Marcelo Pacheco Em Dom 19 Set 2004 19:14, Marconi Rivello escreveu: On Sun, 19 Sep 2004 07:29:16 -0500, Andy [EMAIL PROTECTED] wrote: I had the same problem in an Athlon 900Mhz, with a MSI MoBo. Tried Linux 2.4 and 2.6... Then I changed to a dual Athlon MP 1400Mhz on a Tyan Tiger MoBo. Same problem... Then I moved to a Pentium 4 2.8Ghz, (I can check later, but I'm almost sure it's an Asus MB). It's up and running for 5 days without any problems (or module reloading, or reboot). I also replaced the phone wires with newer ones, but I don't believe that was causing the Red Alarm problem. This week, I'll hook up the old cable just to make sure, but I did read other people complaining about problems on AMD based systems, and also reporting 100% ok on Intel based... I'm not happy about this, but life isn't fair. ___ 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 Red Hat 9
easily - Original Message - From: Henry Devito To: [EMAIL PROTECTED] Sent: Sunday, September 19, 2004 6:30 PM Subject: [Asterisk-Users] Asterisk and Red Hat 9 Hi everyone, Im a newbie to Asterisk. Will Asterisk run on RH9, easily or does it have to run on FreeBSD? Will the drivers for the Digium cards work on RH9? ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Timing source on SMP system - Disable RTC forzaprtc
Chad Brown wrote: Kristian, I have 2 X100P cards but neither work on my Compaq DL360 G2. The system will not even boot! Take a look at my initial post and let me know if you have any other advice. Regardless, thanks for your post! Oh that was you? I read that post earlier but obviously didn't put it together... I don't know about this, but have you tried an add-on usb-uhci controller (your motherboard is OHCI, right)? I don't know if that would work, but you could try it... What about a TDM400 with a module (any module FXS/FXO)? -- Kristian Kielhofner ___ 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 Red Hat 9
Henry Devito wrote: Hi everyone, Im a newbie to Asterisk. Will Asterisk run on RH9, easily or does it have to run on FreeBSD? Will the drivers for the Digium cards work on RH9? Last I heard RedHat was the development platform for *. It doesn't have to run on FreeBSD, and in my experience does not run nearly as well as it does on Linux. Especially with the lack of thorough hardware support under FreeBSD. -- Kristian Kielhofner ___ 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: X100p on VIA EPIA-V
I believe the problem being refered to is described below, which affects VIA chipsets made between 1997-2002. The VT8233 used in the Mini-ITX is on the 'hit list'. I understand there was a patch to the linux kernel which was supposed to have been merged into kernel 2.4.25 to address this issue. Personally, I've not encountered serious problems with the Mini-ITX. I suppose not running X helps reduce the likelihood of a bus mastering problem. http://adsl.cutw.net/dlink-dsl200-via.html Stefan de Konink wrote: Rich Adamson wrote: My understanding from the 'expert' is the PCI issues have something to do with a poor PCI chip design on the motherboard. The folks that are heavy into audio apps tend to swap out their VIA motherboards. Guess that implies there aren't any workarounds. From the Ardour (Linux Audio Editor) System requirements: Avoid VIA motherboards and chipsets wherever possible. This company has demonstrated an almost complete disregard for reasonable use of the PCI bus. Their hardware has repeatedly been implicated in a failure to achieve low latency performance. Here is one example of the kinds of problems you can expect. example points to: http://linux.derkeiler.com/Mailing-Lists/Kernel/2003-09/7761.html Stefan de Konink ___ 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] new ATA box for sale by Linksys
Quoting Matthew Boehm [EMAIL PROTECTED]: Does anyone have one of these models? Can they confirm that it works with any other SIP server? How is the PAP2-NA configured? Web? Phone? The pdf I downloaded from the pap2-na page on shopblt.com says Model: PAP2. The product manager for this devices sent us one and the first thing I did was configure it for my home Asterisk box. It works just as an SPA2000 would. The voice prompts are the same (except no mention of the word Sipura). The web interface looks like it has a different style sheet. ___ 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] English vs American voice files
OK, I've posted the orignal WAV files in 44.1KHZ x 16bit mono format here http://g7ltt.dyndns.org:8010/VoIP/vmukmale-wav.tgz (26MB!) Mark Mark Phillips said: Erm, didn't think of that. Stupidly I deleted the individual wav files. Not a problem though as I have the 3 master files that I recorded them all into. I'll just have to slice it up again. That'll be a few days as I've got family arriving today. Mark Linus Surguy said: I've spent the afternoon recording all the files for the English speaking VM etc. I've parked the file here http://www.g7ltt.com/VoIP/vmukmale.tgz I did it with Audacity at 44.1KHz x 16bit and thenused sox to raise the levels to -3db and then again to down sample them into 8KHz GSM files. The few that I've listened to sound fine. Hi Mark, If you're going to publish these for public use it would be great if you could make them available in two versions, both the Asterisk 'standard' .gsm format, but also either in 8KHz/8bit/alaw raw or wav and/or 32Kbit ADPCM format - these do give a noticable increase in quality for local/PSTN users of telephony applications over GSM format. Either that, or if you could make the original 44.1K 16bit masters available so others could create the alternatives. Unfortunatly *'s ability to play these cleanly seems a bit broken at the moment, but at least we'll have them for when its fixed! Linus -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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
Re: [Asterisk-Users] passing octothorpe
On 19 Sep 2004 at 17:56, Randy Bush wrote: some conferencing systems want you to hit octothorpe (aka pound, hash, etc.). once connected, i would have expected * to be transparent to all dtmf codes. it seems not to be. wiki has not been helpful, it seems to have most references to do with octothorpe in dial plan. so, what do i not understand? The hash can be used to tranfer in Asterisk (the t and T options at the end of the dial line). The standard way to get around this is to use the doublehash (or maybe doublepound but unlikely to be doubleoctothorpe) patch which will allow you to press hash twice for transfer or once to send it to the remote end. IIRC you can also specify the timeout for it to wait for the second hash. Cheers, Matt Riddell http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Re:Asterisk and Red Hat 9 (Henry Devito)
Hi everyone, I'm a newbie to Asterisk. Will Asterisk run on RH9, easily or does it have to run on FreeBSD? Will the drivers for the Digium cards work on RH9? I am running Asterisk on RH9. I did not encounter any problems with both Asterisk and the drivers for the Digium card. _ Keep track of Singapore Malaysia stock prices. http://www.msn.com.sg/money/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new ATA box for sale by Linksys
I have 1 PAP2-NA. Configuration is done thru Phone (IVR) and Web. Im wondering if this ATA supports auto-provisioning. Matthew Boehm wrote: Does anyone have one of these models? Can they confirm that it works with any other SIP server? How is the PAP2-NA configured? Web? Phone? The pdf I downloaded from the pap2-na page on shopblt.com says Model: PAP2. Thanks, Matthew ___ 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 Red Hat 9
we run Asterisk on RedHat 9 with no problems. Works great! Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC665 Third StreetSuite 100San Francisco, CA94107-1901Asterisk Services and Training From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry DevitoSent: Sunday, September 19, 2004 2:30 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk and Red Hat 9 Hi everyone, Im a newbie to Asterisk. Will Asterisk run on RH9, easily or does it have to run on FreeBSD? Will the drivers for the Digium cards work on RH9? signate small logo.gif___ 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] Timing source on SMP system - DisableRTC forzaprtc
Yes, I tried adding an adaptec that I thought was uhci. This didn't work. I didn't check the specs but figured that they would be uhci given the history of ohci. Regardless, I will confirm with adaptec. Thanks, Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Sunday, September 19, 2004 6:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Timing source on SMP system - DisableRTC forzaprtc Chad Brown wrote: Kristian, I have 2 X100P cards but neither work on my Compaq DL360 G2. The system will not even boot! Take a look at my initial post and let me know if you have any other advice. Regardless, thanks for your post! Oh that was you? I read that post earlier but obviously didn't put it together... I don't know about this, but have you tried an add-on usb-uhci controller (your motherboard is OHCI, right)? I don't know if that would work, but you could try it... What about a TDM400 with a module (any module FXS/FXO)? -- Kristian Kielhofner ___ 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] new ATA box for sale by Linksys
Antonio Rabena wrote: I have 1 PAP2-NA. Configuration is done thru Phone (IVR) and Web. Im wondering if this ATA supports auto-provisioning. Can you confirm if under Advanced Settings there is a Provisioning Tab, and under it there is a space for a Profile Rule? -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new ATA box for sale by Linksys
Yes there is.. the default setting is /init.cfg. Not sure about these parameters.. I cant find this provisioning setting on the user-guide. maybe anyone can help? Regards, Antonio Rabena Andres wrote: Antonio Rabena wrote: I have 1 PAP2-NA. Configuration is done thru Phone (IVR) and Web. Im wondering if this ATA supports auto-provisioning. Can you confirm if under Advanced Settings there is a Provisioning Tab, and under it there is a space for a Profile Rule? ___ 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 and Linux 2.6 Kernel
I ran the LiveCD version of Asterisk on my hardware and it worked. I am trying to run it natively on a 2.6 kernel (Gentoo distro), but it keeps getting a seg fault using the sample configuration files. Does Asterisk not work with the 2.6.8 kernel? TIA Chuck Wegrzyn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new ATA box for sale by Linksys
So here is the big question.. just how similar is the hardware in the Linksys PAP2-NA to the Sipura 2000? It would be interesting to see if a PAP2-NA will take the Sipura 2000 firmware That would be great, since I'm guessing there will be quite a bit of lag time between the Sipura updates and Linksys releasing their rebranded version. -jeremy Antonio Rabena wrote: I have 1 PAP2-NA. Configuration is done thru Phone (IVR) and Web. Im wondering if this ATA supports auto-provisioning. Matthew Boehm wrote: Does anyone have one of these models? Can they confirm that it works with any other SIP server? How is the PAP2-NA configured? Web? Phone? The pdf I downloaded from the pap2-na page on shopblt.com says Model: PAP2. Thanks, Matthew ___ 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] Setting time on ADSI phones from Asterisk
Would anyone know of a way to set the time automatically on an ADSI capable phone from *? just call the phone its part of the adv callerid fsk tones it will set it as soon as ypou call the adsi phone i have some 350's and 480's and the the time is set The phone in question is a Aastra 480e. While I am at it, does anyone have any helpful docs on the ADSI script programming? as usual http://www.google.com/custom?hl=enlr=ie=ISO-8859-1cof=sitesearch=lists.d igium.comq=adsi+script http://lists.digium.com/pipermail/asterisk-dev/2004-June/004891.html from a previous post i made the HUGE issue is that v1 adsi is so limited you to write a adsi v2 parser / compiler to replace app_adsiprog-adsi_process the orginal was done for the v1 350's phones but at this point with inexpensive 480i ip phones starting to come on the market why bother using adsi to build an interface/app on the icd display its soo slow compared to ip based phones 480i and the memory is so limited its like stuffing an elephant in a phone booth to make an interesting adsi apps ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new ATA box for sale by Linksys
From your experience, could you give us the merits and demerits of these ATA devices as well as the IAXy. They are essentially a Sipura SPA-2000. One of my customers uses the Sipura exclusively for his customers and they work very well. Setup is easy, and they support the CLASS type features superbly. Thanks to everyone who cleared up the PAP2 versus PAP2-NA. I am not sure if the one bought at Fry's is the NA version or not. I didn't buy it, my customer did. If it's the Vonage-locked version I'm sure he'll return it. I do know that the only version Frys had on the shelf had a Vonage sticker pasted on it, but the side of the box seems to indicate that it will work with any SIP phone service provider. Matthew N. Simpson TxLink Communications SIP/IAX VoIP Origination and Termination Minutes as low as 0.005/minute www.txlink.net/ ___ 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] Effectively using a telco Type 102 Milliwatt Test line with ztmon itor -v to set txgain/rxgain in zapata?
I am trying to obtain optimum gain settings for a bank of analog lines connected to a channel bank. My telco has provided a 'Type 102' test line to use for incoming level calibration. This is functionally equivalent to app Milliwatt(), but provides tone from the CO inwards. Question is, how should one use this a 0dbm test source with ztmonitor? Am I correct in understanding that a 0dbm level should provide a 100% drive in the level monitor, not a 50% drive as is otherwise 'optimum' for regular voice traffic? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District ___ 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] Cisco Phones (7960, 7905G) and tdm400p with FXS for sale
Hi there, I am selling my 7960, 7905g and tdm400p with one fxs module. E-mail me for more info. Just for info the phones both have 3 year service contracts (smart-net) on them. [EMAIL PROTECTED] ___ 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
