Re: [Asterisk-Users] ILS - Internet Locator Service ?
On Wednesday 28 May 2003 15:30, Gary wrote: Has anyone consider or found an opensource ILS ?? doubt it. from http://www.collaborium.org/onsite/peru/christian/h323/tsld027.htm: - ILS is a Microsoft extension to the LDAP protocol - It is used to make it easier for Netmeeting users to find each other - It is not hierarchical - It is not included in the h323 arch. - It can NOT interoperate with a gatekeeper - It is disappearing just thinking of same and possible link with asterisk . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ILS - Internet Locator Service ?
Gary wrote: Has anyone consider or found an opensource ILS ?? just thinking of same and possible link with asterisk . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users IIRC there was a shell-backend to OpenLDAP that would allow to use slapd as an ILS-Server. Ask Google for details. I think it was called Netmeeting Directory Kit. Holger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialogic DIALOG/4
how you mean "board is not full duplex"? ofcourse is full duplex it has been used for iptelephony "long" time ago. have you try this board in linux? you have drivers? i have two cards ;) Thomas Benjamin Miller wrote: This board is not full duplex and will not work with *. I have 3 of them :-( Buy some Digium boards. They will serve you much better and are easier. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Sent: Wednesday, May 28, 2003 3:22 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dialogic DIALOG/4 Good question. This board works well with * ? Can I get the caller-id(ANI) using this board? Isamar On Wed, 28 May 2003, Tomaz Izanc wrote: hi .. anyone using dialogic isa board DIALOG / 4 ? Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialogic DIALOG/4
Title: Message "Full Duplex" means that it has enough voice resources to send and receive data at the same time. It has been used in telephony for a long time but usually as a call recorder or an IVR application that requires only half duplex. I have installed the GlobalCall driver and gotten it to work under Linux. However because it is _not_ full duplex it does not work with Asterisk. -Original Message-From: Tomaz Izanc [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 28, 2003 9:05 AMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] dialogic DIALOG/4how you mean "board is not full duplex"? ofcourse is full duplex it has been used for iptelephony "long" time ago.have you try this board in linux? you have drivers?i have two cards ;)ThomasBenjamin Miller wrote: This board is not full duplex and will not work with *. I have 3 of them :-( Buy some Digium boards. They will serve you much better and are easier. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Sent: Wednesday, May 28, 2003 3:22 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dialogic DIALOG/4 Good question. This board works well with * ? Can I get the caller-id(ANI) using this board? Isamar On Wed, 28 May 2003, Tomaz Izanc wrote: hi .. anyone using dialogic isa board DIALOG / 4 ? Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem w/ Zaptel HDLC mode cisco Data Stability
What network card are you using ? (model and vendor) Martin On Tue, 27 May 2003, Nick Eggleston wrote: We are using the zaptel driver to deliver a combined voice/data T1 circuit. The data channel-group is using the cisco hdlc protocol (on the linux side) and connects with a cisco router on the far side. Everything comes up and works fine for a while. A while later, though, IP communication fails completely. The cisco side shows up/up. The Linux side has a message in the log file stating: hdlc0: Link down. Just prior to the link down message are numerous messages that say New offset: followed by different numbers. That may or may not be relevent. I am curious to know other people results with this configuration, if any. If you had the same problem, how did you solve it? We are planning to switch to PPP and see if that improves the situation. --Nick (implementation details follow:) CISCO: interface Serial1/0:1 ip address 1.0.0.102 255.255.255.252 encapsulation hdlc end LINUX: sethdlc hdlc0 mode cisco ifconfig hdlc0 1.0.0.101 netmask 255.255.255.252 OS: Redhat 8.0 Kernel: 2.4.18-27.8.0 /etc/zaptel.conf: loadzone=us defaultzone=us span=1,1,0,esf,b8zs nethdlc=1-6 bchan=13-23 dchan=24 ZAPTEL driver CVS diff: cvs server: Diffing . Index: Makefile === RCS file: /usr/cvsroot/zaptel/Makefile,v retrieving revision 1.5 diff -r1.5 Makefile 36c36 #KFLAGS+=-DCONFIG_ZAPTEL_MMX --- KFLAGS+=-DCONFIG_ZAPTEL_MMX 54c54 #KFLAGS+=-DCONFIG_ZAPATA_NET --- KFLAGS+=-DCONFIG_ZAPATA_NET ZAPTEL driver CVS/Entries: /.cvsignore/1.1.1.1/Wed Feb 12 13:59:20 2003// /ChangeLog/1.1.1.1/Wed Feb 12 13:59:20 2003// /README.fxsusb/1.1.1.1/Wed Feb 12 13:59:20 2003// /arith.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /biquad.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /bittest.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /complex.cc/1.1.1.1/Mon Mar 17 18:11:45 2003// /complex.h/1.1.1.1/Mon Mar 17 18:11:45 2003// /digits.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /ecdis.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /fasthdlc.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /fir.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /fxstest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /genconst.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /gendigits.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /hdlcgen.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /hdlcstress.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /hdlctest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /hdlcverify.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /ifcfg-hdlc0/1.1.1.1/Wed Feb 12 13:59:20 2003// /ifup-hdlc/1.1.1.1/Wed Feb 12 13:59:20 2003// /makefw.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /mec.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /mec2.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /mec2_const.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /mec3.h/1.3/Wed Apr 16 03:31:07 2003// /mkfilter.h/1.1.1.1/Mon Mar 17 18:11:45 2003// /mknotch.cc/1.1.1.1/Mon Mar 17 18:11:45 2003// /orig.ee/1.1.1.1/Wed Feb 12 13:59:20 2003// /patgen.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /patlooptest.c/1.1.1.2/Sat Mar 15 06:00:30 2003// /pattest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /proslic.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /sec-2.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /sec.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /sethdlc.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /timertest.c/1.1/Thu Mar 20 07:00:04 2003// /tonezone.c/1.2/Mon Apr 14 16:14:55 2003// /tonezone.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /tor.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /tor2-hw.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /tor2.c/1.1.1.5/Mon Mar 17 18:11:45 2003// /tor2.ee/1.1.1.1/Wed Feb 12 13:59:20 2003// /tor2ee.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /torisa.c/1.1.1.2/Mon Mar 17 18:11:45 2003// /torisa.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /torisatool.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /tormenta2.rbt/1.1.1.1/Wed Feb 12 13:59:20 2003// /tormenta2.ucf/1.1.1.1/Wed Feb 12 13:59:20 2003// /tormenta2.vhd/1.1.1.1/Wed Feb 12 13:59:20 2003// /usbfxstest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /wcfxo.c/1.5/Thu Apr 10 21:02:54 2003// /wcfxs.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /wcfxsusb.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /wcfxsusb.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /wct1xxp.c/1.1.1.4/Mon Mar 17 18:11:45 2003// /wcusb.c/1.1.1.2/Fri Mar 14 06:00:34 2003// /wcusb.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /zaptel.conf.sample/1.1.1.2/Mon Mar 17 18:11:45 2003// /zaptel.init/1.1.1.2/Sat Mar 15 06:00:31 2003// /zaptel.sysconfig/1.1.1.1/Wed Feb 12 13:59:20 2003// /zonedata.c/1.3/Fri May 9 22:41:51 2003// /ztcfg-dude.c/1.1.1.1/Tue Mar 18 06:00:29 2003// /ztcfg.c/1.1.1.3/Tue Mar 18 06:00:29 2003// /ztcfg.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /ztd-eth.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /ztdiag.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /ztdummy.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /ztdummy.h/1.1.1.2/Mon Feb 24 06:00:31 2003// /ztdynamic.c/1.1.1.2/Mon Mar 17 18:11:45 2003// /ztmonitor.c/1.1.1.2/Mon Feb 24 06:00:31 2003//
Re: [Asterisk-Users] Problem w/ Zaptel HDLC mode cisco Data Stability
Digium T100P On Wed, 28 May 2003, Martin Pycko wrote: What network card are you using ? (model and vendor) Martin On Tue, 27 May 2003, Nick Eggleston wrote: We are using the zaptel driver to deliver a combined voice/data T1 circuit. The data channel-group is using the cisco hdlc protocol (on the linux side) and connects with a cisco router on the far side. Everything comes up and works fine for a while. A while later, though, IP communication fails completely. The cisco side shows up/up. The Linux side has a message in the log file stating: hdlc0: Link down. Just prior to the link down message are numerous messages that say New offset: followed by different numbers. That may or may not be relevent. I am curious to know other people results with this configuration, if any. If you had the same problem, how did you solve it? We are planning to switch to PPP and see if that improves the situation. --Nick (implementation details follow:) CISCO: interface Serial1/0:1 ip address 1.0.0.102 255.255.255.252 encapsulation hdlc end LINUX: sethdlc hdlc0 mode cisco ifconfig hdlc0 1.0.0.101 netmask 255.255.255.252 OS: Redhat 8.0 Kernel: 2.4.18-27.8.0 /etc/zaptel.conf: loadzone=us defaultzone=us span=1,1,0,esf,b8zs nethdlc=1-6 bchan=13-23 dchan=24 ZAPTEL driver CVS diff: cvs server: Diffing . Index: Makefile === RCS file: /usr/cvsroot/zaptel/Makefile,v retrieving revision 1.5 diff -r1.5 Makefile 36c36 #KFLAGS+=-DCONFIG_ZAPTEL_MMX --- KFLAGS+=-DCONFIG_ZAPTEL_MMX 54c54 #KFLAGS+=-DCONFIG_ZAPATA_NET --- KFLAGS+=-DCONFIG_ZAPATA_NET ZAPTEL driver CVS/Entries: /.cvsignore/1.1.1.1/Wed Feb 12 13:59:20 2003// /ChangeLog/1.1.1.1/Wed Feb 12 13:59:20 2003// /README.fxsusb/1.1.1.1/Wed Feb 12 13:59:20 2003// /arith.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /biquad.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /bittest.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /complex.cc/1.1.1.1/Mon Mar 17 18:11:45 2003// /complex.h/1.1.1.1/Mon Mar 17 18:11:45 2003// /digits.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /ecdis.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /fasthdlc.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /fir.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /fxstest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /genconst.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /gendigits.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /hdlcgen.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /hdlcstress.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /hdlctest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /hdlcverify.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /ifcfg-hdlc0/1.1.1.1/Wed Feb 12 13:59:20 2003// /ifup-hdlc/1.1.1.1/Wed Feb 12 13:59:20 2003// /makefw.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /mec.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /mec2.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /mec2_const.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /mec3.h/1.3/Wed Apr 16 03:31:07 2003// /mkfilter.h/1.1.1.1/Mon Mar 17 18:11:45 2003// /mknotch.cc/1.1.1.1/Mon Mar 17 18:11:45 2003// /orig.ee/1.1.1.1/Wed Feb 12 13:59:20 2003// /patgen.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /patlooptest.c/1.1.1.2/Sat Mar 15 06:00:30 2003// /pattest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /proslic.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /sec-2.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /sec.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /sethdlc.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /timertest.c/1.1/Thu Mar 20 07:00:04 2003// /tonezone.c/1.2/Mon Apr 14 16:14:55 2003// /tonezone.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /tor.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /tor2-hw.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /tor2.c/1.1.1.5/Mon Mar 17 18:11:45 2003// /tor2.ee/1.1.1.1/Wed Feb 12 13:59:20 2003// /tor2ee.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /torisa.c/1.1.1.2/Mon Mar 17 18:11:45 2003// /torisa.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /torisatool.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /tormenta2.rbt/1.1.1.1/Wed Feb 12 13:59:20 2003// /tormenta2.ucf/1.1.1.1/Wed Feb 12 13:59:20 2003// /tormenta2.vhd/1.1.1.1/Wed Feb 12 13:59:20 2003// /usbfxstest.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /wcfxo.c/1.5/Thu Apr 10 21:02:54 2003// /wcfxs.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /wcfxsusb.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /wcfxsusb.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /wct1xxp.c/1.1.1.4/Mon Mar 17 18:11:45 2003// /wcusb.c/1.1.1.2/Fri Mar 14 06:00:34 2003// /wcusb.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /zaptel.conf.sample/1.1.1.2/Mon Mar 17 18:11:45 2003// /zaptel.init/1.1.1.2/Sat Mar 15 06:00:31 2003// /zaptel.sysconfig/1.1.1.1/Wed Feb 12 13:59:20 2003// /zonedata.c/1.3/Fri May 9 22:41:51 2003// /ztcfg-dude.c/1.1.1.1/Tue Mar 18 06:00:29 2003// /ztcfg.c/1.1.1.3/Tue Mar 18 06:00:29 2003// /ztcfg.h/1.1.1.1/Wed Feb 12 13:59:20 2003// /ztd-eth.c/1.1.1.1/Wed Feb 12 13:59:20 2003// /ztdiag.c/1.1.1.1/Wed Feb 12 13:59:20 2003//
Re: [Asterisk-Users] Bridging two iconnect calls
1) you need two accounts in iconnecthere 2) you need to register with two accounts 3) then simply receive the call using one and send it over another account Martin On Wed, 28 May 2003, pradeep kumar wrote: Hi All. I am trying to setup asterisk so that I can place two outbound calls via iconnecthere and connect them. Is this possible ? If this is the case, please let me know what I need in the extension conf to accomodate this feature. Thanks in advance PS : I did send a similar mail yesterday, but it does not seem to have reached the mailing list. If this appears to be a duplicate please ignore. Pradeep _ Reconnect with old pals. Relive the happy times. http://www.batchmates.com/msn.asp With just one click. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Conferencing
If you don't have any hardware for conferencing than you could use the ztdummy from zaptel package. Check the archives. look for ztdummy Martin On Tue, 27 May 2003, Rahul Gupta wrote: Hello , I am a newbie to * and have just been able to call a sip User Agent on a different machine thru *. I was trying to set up conferencing between 3 sip useragent on different macines at my worplace but was not able to figure out the procedure. I made the changes in meetme.conf and extension.conf as specified by someone in this mailing list, but * giving some error, No ISA Tormeta card found. Does conferencing require some special hardware on the machine on which * is running ?? thanks rahul __ Do you Yahoo!? Yahoo! Calendar - Free online calendar with sync to Outlook(TM). http://calendar.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] immediate on fxo
When immediate is set on a port that is an fxo, what is the meaning of this ? Will it go immediately to the s extension of the context when the line first rings, or something else ? I am looking for a way to stop the line ringing two extra times before being answered by the channel bank. (caller id is set to off I believe, and the lines in question don't have it anyway.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ILS - Internet Locator Service ?
Has anyone consider or found an opensource ILS ?? just thinking of same and possible link with asterisk IIRC there was a shell-backend to OpenLDAP that would allow to use slapd as an ILS-Server. Ask Google for details. I think it was called Netmeeting Directory Kit. There is information on creating a NetMeeting compatible ILS server using OpenLDAP in my LDAP presentation - ftp://ftp.kalamazoolinux.org/pub/pdf/ldapv3.pdf I've tested it with GNOME Meeting as well, which also worked. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] immediate on fxo
FXO ports don't get DID numbers usually so they'll always go to 's' Martin On Wed, 28 May 2003, Jon Pounder wrote: When immediate is set on a port that is an fxo, what is the meaning of this ? Will it go immediately to the s extension of the context when the line first rings, or something else ? I am looking for a way to stop the line ringing two extra times before being answered by the channel bank. (caller id is set to off I believe, and the lines in question don't have it anyway.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 4-port T1 cards
1. Voicemail, and the voicemail itself will be stored on another box, NFS mounted, or I might use mysql. There will be a little bit of call routing via iax to a separate * box with a channel bank on it. 2. I don't disagree with you, they do throw in a lot, but redhat does have its advantages, IMHO. I've always been able to get things to work quickly with redhat, and there is that whole 24 hour support contract we have with them... 3. Mmm, ok. 4. Does the ati radeon 9000 have a frame buffer? That's the card I was going to use for all the * boxes. Thank you very much. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Tuesday, May 27, 2003 6:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 4-port T1 cards On Tue, 2003-05-27 at 16:05, Joe Antkowiak wrote: Are there any known issues with putting 2 4-port T1 cards in a single box and having all ports and all channels in use at the same time? Planning on 4 of these boxes, dual AMD cpu MB from MSI, 512m, redhat 9, agp video, on board NICs, serial ata raid. Newbie 101 (Not deragatory) 1. What are you doing with these ports? If you are routing calls from one side of the cards to the other, then you should have no problems with a 1gig P3 or so. But if you are doing more than routing, it will depend on what that something is, and what kind of overhead it is going to impose. 2. RH blows chunks. (Personal opinion) RH is known to make kitchen sink installs when you don't need them, and would be better off without most of the install base. 3. Dual MB won't help much in pure telephony. In pure telephony, you are basically dealing with serial line IO. A T1 is little more than I long distance serial line. 8 T1s is just 11.7megs per second each way, or 23.4 megs in and out. Not too much for a good machine to do. Granted, if you are doing VoIP then you add another set of ins and outs with compression inthe middle of it too. This is where the second CPU comes in handy. 4. AGP Video. Make sure not to use the frame buffer, it has been reported that the frame buffer generates large amounts of interupts and will degrade the performance. Here is for discussion as it is parts I don't know real well. Will the serial ATA buy you any flexibilty or lowered CPU load while accessing the disk? Don't take this question as shooting down the SATA, just don't know if there is real benefit in it yet. Also what chipset is the onboard nics? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 4-port T1 cards
Cool, dual CPU it is =) I'll watch out for the frame buffers. Other than that, are there any other known issues with doing this? I will also be using dual-channel memory... Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: Wednesday, May 28, 2003 12:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 4-port T1 cards 3. Dual MB won't help much in pure telephony. In pure telephony, you are basically dealing with serial line IO. A T1 is little more than I long distance serial line. 8 T1s is just 11.7megs per second each way, or 23.4 megs in and out. Not too much for a good machine to do. Granted, if you are doing VoIP then you add another set of ins and outs with compression in the middle of it too. This is where the second CPU comes in handy. Actually, with Zaptel, and the T400P especially, dual CPU makes a *big* difference. The T400P and E400P are slave-only designs, so the CPU spends a lot of time just cramming I/O down the PCI bus. Having a second CPU free to do work will definitely help. 4. AGP Video. Make sure not to use the frame buffer, it has been reported that the frame buffer generates large amounts of interupts and will degrade the performance. Don't underestimate this effect or think that a fast CPU will get around it. frame buffer is a definite no-no because it disables interrupts during screen redraws which take an enormous amount of time. If your call quality drops while you're playing quake on your PBX, don't come crying to us ;-) Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicetronix support
Check the archives, someone got the openline card working. I had been talking with voicetronix about writing a driver for the openswitch cards but they haven't been interested recently. dave On Wed, 28 May 2003, Daniel ANDRE wrote: Hello, I would like to know if voicetronix card (specially openswitch6 and 12) can be used with asterisk. Is there any driver for this card? Best regards, Daniel -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] per virtual pbx VM storage
You may be able to solve this with either multiple mount points or symlinks to alternate disk partitions. Or, a really ugly method would be to write an quickie AGI that parses the output of du -s /var/spool/asterisk/voicemail/blah and if that number is higher than the permitted value, new voicemail is not allowed (divert before sending to voicemail app.) JT Is there any way to control the file system location of where Voicemail is stored per virtual PBX? I'm thinking in the context of hosting Virtual PBX customers who would have a disk quota for voicemail and the need to enforce a hard limit. Thanks, Steve Bourg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] immediate on fxo
Nope. usecallerid=no should work for it. If not you might try to modify the code in chan_zap.c Martin On Wed, 28 May 2003, Jon Pounder wrote: my question was - will immediate put an end to the extra 2 rings before pickup ? (I know they go to s eventually.) At 10:40 AM 5/28/2003 -0500, you wrote: FXO ports don't get DID numbers usually so they'll always go to 's' Martin On Wed, 28 May 2003, Jon Pounder wrote: When immediate is set on a port that is an fxo, what is the meaning of this ? Will it go immediately to the s extension of the context when the line first rings, or something else ? I am looking for a way to stop the line ringing two extra times before being answered by the channel bank. (caller id is set to off I believe, and the lines in question don't have it anyway.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie: Soundsdirectory
Sorry, resolved it myself. Because I need to load all modules manually I did not load enough codecs and formats. - Original Message - From: Jan Boon To: [EMAIL PROTECTED] Sent: Wednesday, May 28, 2003 5:46 PM Subject: [Asterisk-Users] Newbie: Soundsdirectory I am trying to implement MeetMeCount but got the error could not find digits/0 in any form. I know modulesshould bein /usr/lib/asterisk and soundsin /var/lib/asterisk and they are. The path to modules is specified in asterisk.conf (samples). The path to sounds not. Could not find the syntax for it in the handbook or anywhere else. Could somebody give me a hint?
[Asterisk-Users] REMOVE
--- Gary [EMAIL PROTECTED] wrote: I just had a thought, but haven't tried it yet we are able to include one conf file into another... this leads to the thought can we use the same file to include into iax.conf as well as sif.conf ? does the various differences actually have an effect of the application reading the file ? or another way will nonexisitant controlcommands have an adverse effect eg: in SIP.conf having nat=yes, what happens if that was in the same user stuff in iax.conf ?? Gary . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Calendar - Free online calendar with sync to Outlook(TM). http://calendar.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 4-port T1 cards
On Wed, 2003-05-28 at 11:02, Joe Antkowiak wrote: 1. Voicemail, and the voicemail itself will be stored on another box, NFS mounted, or I might use mysql. There will be a little bit of call routing via iax to a separate * box with a channel bank on it. 2. I don't disagree with you, they do throw in a lot, but redhat does have its advantages, IMHO. I've always been able to get things to work quickly with redhat, and there is that whole 24 hour support contract we have with them... 3. Mmm, ok. 4. Does the ati radeon 9000 have a frame buffer? That's the card I was going to use for all the * boxes. Yes, it does., but just don't use that driver in the kernel and your okay. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about Voicemail and Voicemail2
I would like to know if maxsilence and silencethreshold parameters in voicemail.conf work only for voicemail2 application and what are the main differences that exist between voicemail and voicemail2. What possible values silencethreshold can take? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 4-port T1 cards
On Wed, 2003-05-28 at 11:02, Joe Antkowiak wrote: 1. Voicemail, and the voicemail itself will be stored on another box, NFS mounted, or I might use mysql. There will be a little bit of call routing via iax to a separate * box with a channel bank on it. Don't use Mysql. if you ever have had to deal with it in a production environment that works it over, you will know that as it reaches it's limits, it starts a death spiral that is very difficult to recover from. For our software on a dual P3 866 with a gig of ram, the limit was around 1.5 queries a second fairly mixed update, inserts, and selects. Total file size of the database was under 200meg, and was fully cached so even though we had hardware raid 5 across 4 10K rpm ultra160 drives, it shouldn't have mattered for the selects. Also, NFS mounting of the voicemail for such a large install is probably not the best idea. Unless you really need it available to another machine, you _may_ want to rethink this idea. NFS can be a major speed hit on a machine, especially if the client is overworked. Also if you are planning on running most all the channels to voicemail, then do you think you are going to be able to have your NFS server keep high speed writing going so as not to slow you asterisk machine down with it's 96 channels running full tilt? 2. I don't disagree with you, they do throw in a lot, but redhat does have its advantages, IMHO. I've always been able to get things to work quickly with redhat, and there is that whole 24 hour support contract we have with them... 3. Mmm, ok. 4. Does the ati radeon 9000 have a frame buffer? That's the card I was going to use for all the * boxes. Thank you very much. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Tuesday, May 27, 2003 6:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 4-port T1 cards On Tue, 2003-05-27 at 16:05, Joe Antkowiak wrote: Are there any known issues with putting 2 4-port T1 cards in a single box and having all ports and all channels in use at the same time? Planning on 4 of these boxes, dual AMD cpu MB from MSI, 512m, redhat 9, agp video, on board NICs, serial ata raid. Newbie 101(Not deragatory) 1. What are you doing with these ports? If you are routing calls from one side of the cards to the other, then you should have no problems with a 1gig P3 or so. But if you are doing more than routing, it will depend on what that something is, and what kind of overhead it is going to impose. 2. RH blows chunks. (Personal opinion) RH is known to make kitchen sink installs when you don't need them, and would be better off without most of the install base. 3. Dual MB won't help much in pure telephony. In pure telephony, you are basically dealing with serial line IO. A T1 is little more than I long distance serial line. 8 T1s is just 11.7megs per second each way, or 23.4 megs in and out. Not too much for a good machine to do. Granted, if you are doing VoIP then you add another set of ins and outs with compression in the middle of it too. This is where the second CPU comes in handy. 4. AGP Video. Make sure not to use the frame buffer, it has been reported that the frame buffer generates large amounts of interupts and will degrade the performance. Here is for discussion as it is parts I don't know real well. Will the serial ATA buy you any flexibilty or lowered CPU load while accessing the disk? Don't take this question as shooting down the SATA, just don't know if there is real benefit in it yet. Also what chipset is the onboard nics? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 4-port T1 cards
On Wed, 2003-05-28 at 13:30, Steven Critchfield wrote: Don't use Mysql. if you ever have had to deal with it in a production environment that works it over, you will know that as it reaches it's limits, it starts a death spiral that is very difficult to recover from. For our software on a dual P3 866 with a gig of ram, the limit was around 1.5 queries a second fairly mixed update, inserts, and selects. Total file size of the database was under 200meg, and was fully cached so even though we had hardware raid 5 across 4 10K rpm ultra160 drives, it shouldn't have mattered for the selects. I'd check what configuration problem you have with mysql then. Uptime: 48 days 1 hour 42 min 6 sec Threads: 15 Questions: 376005206 Slow queries: 22 Opens: 214 Flush tables: 1 Open tables: 64 Queries per second avg: 90.531 This is a mixture of inserts updates and selects as well. slashdot.org runs mysql at loads way over this, and it continues to run, although its very select heavy most likely. signature.asc Description: This is a digitally signed message part
RE: [Asterisk-Users] 2 4-port T1 cards
On Wed, 2003-05-28 at 14:30, Steven Critchfield wrote: Don't use Mysql. if you ever have had to deal with it in a production environment that works it over, you will know that as it reaches it's limits, it starts a death spiral that is very difficult to recover from. For our software on a dual P3 866 with a gig of ram, the limit was around 1.5 queries a second fairly mixed update, inserts, and selects. Total file size of the database was under 200meg, and was fully cached so even though we had hardware raid 5 across 4 10K rpm ultra160 drives, it shouldn't have mattered for the selects. What table type? The default MyISAM tables don't support row-level locking and thus are horrible if you do a lot of inserts or updates. InnoDB tables however are much, much better. Hell SlashDot runs on MySQL with InnoDB. :) -- Joshua M. Thompson [EMAIL PROTECTED] Planet Jurai ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 4-port T1 cards
I've run tests with MySQL. It is very fast unless you have many processes that need to access it at once and then the weak point is it's locking method which let's only one process in at a time So you run into a wall as the size/complexity of your project grows past some point. But for a PBX all you are doing is using MySQL as a log file. It's a triveal application. Just one process doing a few inserts. This is where MySQL is at it's best. For the triveal case of one table and one process doing inserts I've done hundres per second but with a dozen processes and a doxen related tables we saw signs of the death spiral and even incorrect results and lock ups. The bigger, higher end DBMSes start out slower but scale better to more complex tasks and more concurent users. The fastest way to get between two points in a city using ground transport might be a motorcycle. But what if your job is to move 10,000 cases of beer between those two points. fastest is the wrong benchmark in that case. Even a slow and plodding horse drawn wagon would beat the bike at that job. It's kind of the same with databases. --- Ryan Butler [EMAIL PROTECTED] wrote: On Wed, 2003-05-28 at 13:30, Steven Critchfield wrote: Don't use Mysql. if you ever have had to deal with it in a production environment that works it over, you will know that as it reaches it's limits, it starts a death spiral that is very difficult to recover from. For our software on a dual P3 866 with a gig of ram, the limit was around 1.5 queries a second fairly mixed update, inserts, and selects. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Calendar - Free online calendar with sync to Outlook(TM). http://calendar.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 4-port T1 cards
On Wed, 2003-05-28 at 14:11, Ryan Butler wrote: On Wed, 2003-05-28 at 13:30, Steven Critchfield wrote: Don't use Mysql. if you ever have had to deal with it in a production environment that works it over, you will know that as it reaches it's limits, it starts a death spiral that is very difficult to recover from. For our software on a dual P3 866 with a gig of ram, the limit was around 1.5 queries a second fairly mixed update, inserts, and selects. Total file size of the database was under 200meg, and was fully cached so even though we had hardware raid 5 across 4 10K rpm ultra160 drives, it shouldn't have mattered for the selects. I'd check what configuration problem you have with mysql then. Uptime: 48 days 1 hour 42 min 6 sec Threads: 15 Questions: 376005206 Slow queries: 22 Opens: 214 Flush tables: 1 Open tables: 64 Queries per second avg: 90.531 This is a mixture of inserts updates and selects as well. This is partly why I qualified it with the statement with our application. I admit there was several things I inherited that are way sub optimal. Mysql didn't help matters when it went into it's death spiral though. Not all of it was the fault of Mysql, it just doesn't handle it gracefully. My biggest grip beyond table level locking was busy idling a process while the table was locked instead of giving up the processor so it could service the thread that had the table locked. I'd end up with 5+ processes queued up waiting for a table to be free using as much CPU as they could blocking on the table. slashdot.org runs mysql at loads way over this, and it continues to run, although its very select heavy most likely. Actually it is mostly cached. For the case of slashdot, 95% or more only read the front page, and the values of messages do not need to be accurate. Of the 5% that read deeper, very few actually post, and only the posts require anything more than selects. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 4-port T1 cards
On Wed, 2003-05-28 at 14:36, Ron Gage wrote: On Wednesday 28 May 2003 02:30 pm, Steven Critchfield wrote: On Wed, 2003-05-28 at 11:02, Joe Antkowiak wrote: 1. Voicemail, and the voicemail itself will be stored on another box, NFS mounted, or I might use mysql. There will be a little bit of call routing via iax to a separate * box with a channel bank on it. Don't use Mysql. if you ever have had to deal with it in a production environment that works it over, you will know that as it reaches it's limits, it starts a death spiral that is very difficult to recover from. For our software on a dual P3 866 with a gig of ram, the limit was around 1.5 queries a second fairly mixed update, inserts, and selects. Total file size of the database was under 200meg, and was fully cached so even though we had hardware raid 5 across 4 10K rpm ultra160 drives, it shouldn't have mattered for the selects. Um... I suppose that if MySQL can't handle more than 1.5 operations a second, someone should tell the folks at Slashdot and Yahoo that their choice of a DB engine isn't going to scale too well. See other post about why Slashdot isn't a good argument here. It applies equally well for Yahoo. They both are select heavy, and Mysql is able to parallelize these just fine. I also doubt Yahoo does it's updates to the production databases, they probably do it to a backend that then gets replicated to multiple less backends that do the real serving of data. For that matter, I suppose that GIS database I have here (the entire Tiger census data for the state of Michigan - 1.2 million type A records alone) isn't capable of handling more than 1.5 transactions a second. So, when I generate an AutoCAD script from the database records (generates a full-scale road map of the entire state of Michigan), it shouldn't be capable of running in under 10 minutes (at 1.5 transactions a second, it would take 13,333 minutes to run the script). Thats great for a single user. Now put 30 processes doing random updates and inserts while 5 users try and generate that map. I would strongly suggest that something is seriously messed up with your MySQL implementation if you are only capable of getting 1.5 transactions a second before the spiral of death. For comparison, I am running Mysql 4.0.12 on Slackware 9.0. AMD XP1800, 1 gig memory, single 73 gig SCSI-Wide drive, Adaptec 29160 controller. It can do a select distinct across the 1.2 million records in under 4 seconds. Again, 1 user on decent hardware, whoopeee. Try scaling that out and watch it die. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 4-port T1 cards
I'll save my typing fingers somewhat on this one - you are doing great arguing about all the crappiness of mysql and actually backing it up with real examples. It is nice to see that for a change in comparison to all the mysql lovers that love it just because but have no basis to compare it to something with heavy load. I, like you don't consider massive amounts of selects heavy at all. My example of heavy load where mysql could not even begin to handle the situation was a project with real time stock market data streamed in as bids and offers and trades happened, statistics computed from that in real time, database kept in sync live, and charts and graphs plotted in real time for users on the site. Now that situation had more than its share of inserts and updates, and a massive wad of historical data being kept just to add to the fun. Might I add for record that postgres did just fine. At 02:56 PM 5/28/2003 -0500, you wrote: On Wed, 2003-05-28 at 14:36, Ron Gage wrote: On Wednesday 28 May 2003 02:30 pm, Steven Critchfield wrote: On Wed, 2003-05-28 at 11:02, Joe Antkowiak wrote: 1. Voicemail, and the voicemail itself will be stored on another box, NFS mounted, or I might use mysql. There will be a little bit of call routing via iax to a separate * box with a channel bank on it. Don't use Mysql. if you ever have had to deal with it in a production environment that works it over, you will know that as it reaches it's limits, it starts a death spiral that is very difficult to recover from. For our software on a dual P3 866 with a gig of ram, the limit was around 1.5 queries a second fairly mixed update, inserts, and selects. Total file size of the database was under 200meg, and was fully cached so even though we had hardware raid 5 across 4 10K rpm ultra160 drives, it shouldn't have mattered for the selects. Um... I suppose that if MySQL can't handle more than 1.5 operations a second, someone should tell the folks at Slashdot and Yahoo that their choice of a DB engine isn't going to scale too well. See other post about why Slashdot isn't a good argument here. It applies equally well for Yahoo. They both are select heavy, and Mysql is able to parallelize these just fine. I also doubt Yahoo does it's updates to the production databases, they probably do it to a backend that then gets replicated to multiple less backends that do the real serving of data. For that matter, I suppose that GIS database I have here (the entire Tiger census data for the state of Michigan - 1.2 million type A records alone) isn't capable of handling more than 1.5 transactions a second. So, when I generate an AutoCAD script from the database records (generates a full-scale road map of the entire state of Michigan), it shouldn't be capable of running in under 10 minutes (at 1.5 transactions a second, it would take 13,333 minutes to run the script). Thats great for a single user. Now put 30 processes doing random updates and inserts while 5 users try and generate that map. I would strongly suggest that something is seriously messed up with your MySQL implementation if you are only capable of getting 1.5 transactions a second before the spiral of death. For comparison, I am running Mysql 4.0.12 on Slackware 9.0. AMD XP1800, 1 gig memory, single 73 gig SCSI-Wide drive, Adaptec 29160 controller. It can do a select distinct across the 1.2 million records in under 4 seconds. Again, 1 user on decent hardware, whoopeee. Try scaling that out and watch it die. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] About Channel Banks
Hello I'm starting to learn about Asterisk and trying to install the first one. I've a doubt. Here in Brazil the Telecom Standards uses E1. So, I'll use a Wildcard E100P to inter-connect with 30 POTS. Should I use a Channel Bank or simply a Multiplexer? Whats the difference between a channel bank and a multiplexer?? Can I use any? The real question is: How to connect Analog Phones to Asterisk using by output an E1? Thanks in advance Ricardo Saar Gemignani Porto Alegre - RS - Brazil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Channel Banks
Make sure that you don't have a R2 signalling. Since then you'll have problems EuroISDN PRI is all right. Martin On Wed, 28 May 2003, Ricardo Saar Gemignani wrote: Hello I'm starting to learn about Asterisk and trying to install the first one. I've a doubt. Here in Brazil the Telecom Standards uses E1. So, I'll use a Wildcard E100P to inter-connect with 30 POTS. Should I use a Channel Bank or simply a Multiplexer? Whats the difference between a channel bank and a multiplexer?? Can I use any? The real question is: How to connect Analog Phones to Asterisk using by output an E1? Thanks in advance Ricardo Saar Gemignani Porto Alegre - RS - Brazil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 4-port T1 cards
On Wed, 2003-05-28 at 15:09, Jon Pounder wrote: I'll save my typing fingers somewhat on this one - you are doing great arguing about all the crappiness of mysql and actually backing it up with real examples. It is nice to see that for a change in comparison to all the mysql lovers that love it just because but have no basis to compare it to something with heavy load. I, like you don't consider massive amounts of selects heavy at all. My example of heavy load where mysql could not even begin to handle the situation was a project with real time stock market data streamed in as bids and offers and trades happened, statistics computed from that in real time, database kept in sync live, and charts and graphs plotted in real time for users on the site. Now that situation had more than its share of inserts and updates, and a massive wad of historical data being kept just to add to the fun. Might I add for record that postgres did just fine. While I'm on the postgres bandwagon for now, I wouldn't want it in the middle of a phone system doing heavy call loads either. Postgres also has some downsides too. As I understand it, postgres doesn't understand prepared statements, or at least it doesn't via the perl DBI. Regardless I've seen our postgres database eat +2600 updates in under 2 seconds from a remote host on the same exact hardware that mysql choked on and not cause any degredation of access times for any other user. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Channel Banks
Sorry, what is R2 signalling? When I've that? - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 28, 2003 5:43 PM Subject: Re: [Asterisk-Users] About Channel Banks Make sure that you don't have a R2 signalling. Since then you'll have problems EuroISDN PRI is all right. Martin On Wed, 28 May 2003, Ricardo Saar Gemignani wrote: Hello I'm starting to learn about Asterisk and trying to install the first one. I've a doubt. Here in Brazil the Telecom Standards uses E1. So, I'll use a Wildcard E100P to inter-connect with 30 POTS. Should I use a Channel Bank or simply a Multiplexer? Whats the difference between a channel bank and a multiplexer?? Can I use any? The real question is: How to connect Analog Phones to Asterisk using by output an E1? Thanks in advance Ricardo Saar Gemignani Porto Alegre - RS - Brazil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Channel Banks
On Wed, 2003-05-28 at 15:34, Ricardo Saar Gemignani wrote: Hello I'm starting to learn about Asterisk and trying to install the first one. I've a doubt. Here in Brazil the Telecom Standards uses E1. So, I'll use a Wildcard E100P to inter-connect with 30 POTS. Should I use a Channel Bank or simply a Multiplexer? Whats the difference between a channel bank and a multiplexer?? A channel bank is a multiplexer. It also is a demultiplexer. It depends only on which direction you are looking from. From a E1, it is a demultiplexer. Can I use any? As long as it meets your needs. The real question is: How to connect Analog Phones to Asterisk using by output an E1? First, you need to get FXO ports for the converting of the POTS lines to E1 to be imported into asterisk. Then you need one with FXS ports to be connected to asterisk to drive analog extensions. You may want to make a smaller step first. Check with your telco and see if they will drop a E1 into your location to replace the 30 POTS lines. If so, they may even provide a channel bank to ease the transition. You may even be able to get them to either lease it to you, or give it to you for a certain contract length. This would reduce your needs to get started as you are only looking at a single DA conversion and less signalling problems. Also you put the cost off onto your telco for part of the install. Your E1 hopefully is cheaper than 30 POTS lines if you are close to a CO. Thanks in advance Ricardo Saar Gemignani Porto Alegre - RS - Brazil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 827-4v SIP config
has anyone ever used an 827-4v DSL router to do SIPtoPOTS conversion? how would I set up my 4 pots lines to be SIP extensions/phones? any ideas? Cisco site is a little lacking on sample configs. I would like to set up 3 of the ports for FXS analog sets and one port for FXO. convert all the analog to SIP and send it to a SIp gateway like FWD or Asterisk. Any thought or suggestions/examples would be a great help Thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 4-port T1 cards
Don't use Mysql. if you ever have had to deal with it in a production environment that works it over, you will know that as it reaches it's limits, it starts a death spiral that is very difficult to recover from. For our software on a dual P3 866 with a gig of ram, the limit was around 1.5 queries a second fairly mixed update, inserts, and selects. Total file size of the database was under 200meg, and was fully cached so even though we had hardware raid 5 across 4 10K rpm ultra160 drives, it shouldn't have mattered for the selects. Wow! I thought it was just me. I actually went through the quite painfull migration process from mysql to postgresql in order to recover from mysql's death spiral. Ouch! Good advice. Mike Diehl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 4-port T1 cards
My example of heavy load where mysql could not even begin to handle the situation was a project with real time stock market data streamed in as bids and offers and trades happened, statistics computed from that in real time, database kept in sync live, and charts and graphs plotted in real time for users on the site. Now that situation had more than its share of inserts and updates, and a massive wad of historical data being kept just to add to the fun. ...and I can even top that. I had an Intrusion Detection System proof of concept which logged to a Postgresql database. This database received 10-15 Million inserts a day and I was required to keep 90 day's worth of data. Most of the elects were done in batch during slow times at night. Some of the queries were expected to run in near-real time. And might I also add that postgres did just fine... Might I add for record that postgres did just fine. Mike Diehl. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 4-port T1 cards
was that mysql 3.23.x or 4.0.x ? michael *** REPLY SEPARATOR *** On 28.05.2003 at 16:18 [EMAIL PROTECTED] wrote: My example of heavy load where mysql could not even begin to handle the situation was a project with real time stock market data streamed in as bids and offers and trades happened, statistics computed from that in real time, database kept in sync live, and charts and graphs plotted in real time for users on the site. Now that situation had more than its share of inserts and updates, and a massive wad of historical data being kept just to add to the fun. ...and I can even top that. I had an Intrusion Detection System proof of concept which logged to a Postgresql database. This database received 10-15 Million inserts a day and I was required to keep 90 day's worth of data. Most of the elects were done in batch during slow times at night. Some of the queries were expected to run in near-real time. And might I also add that postgres did just fine... Might I add for record that postgres did just fine. Mike Diehl. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 4-port T1 cards
Steven Critchfield wrote: Don't use Mysql. if you ever have had to deal with it in a production environment that works it over, you will know that as it reaches it's limits, it starts a death spiral that is very difficult to recover from. For our software on a dual P3 866 with a gig of ram, the limit was around 1.5 queries a second fairly mixed update, inserts, and selects. Total file size of the database was under 200meg, and was fully cached so even though we had hardware raid 5 across 4 10K rpm ultra160 drives, it shouldn't have mattered for the selects. You must have a major flaw in your database architecture. We very easily run 5-7 mixed queries a second on a nothing special Dell 1U PowerEdge server using MySQL-Max-3.23 and it doesn't even break a sweat. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 4-port T1 cards
was that mysql 3.23.x or 4.0.x ? michael I did most of my mysql work some time ago with 3.x. I have, however, installed mysql 4.0.12 since I'm working on a CASE tool which needs to support both mysql and postgresql. I'll be the first to admit that mysql has probably improved (a lot?) since then. I was simply relating my experience with both mysql and postgresql. *** REPLY SEPARATOR *** On 28.05.2003 at 16:18 [EMAIL PROTECTED] wrote: My example of heavy load where mysql could not even begin to handle the situation was a project with real time stock market data streamed in as bids and offers and trades happened, statistics computed from that in real time, database kept in sync live, and charts and graphs plotted in real time for users on the site. Now that situation had more than its share of inserts and updates, and a massive wad of historical data being kept just to add to the fun. ...and I can even top that. I had an Intrusion Detection System proof of concept which logged to a Postgresql database. This database received 10-15 Million inserts a day and I was required to keep 90 day's worth of data. Most of the elects were done in batch during slow times at night. Some of the queries were expected to run in near-real time. And might I also add that postgres did just fine... Might I add for record that postgres did just fine. Mike Diehl. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 4-port T1 cards
On 28 May 2003, Steven Critchfield waxed: 8's While I'm on the postgres bandwagon for now, I wouldn't want it in the middle of a phone system doing heavy call loads either. Postgres also has some downsides too. As I understand it, postgres doesn't understand prepared statements, or at least it doesn't via the perl DBI. Regardless I've seen our postgres database eat +2600 updates in under 2 seconds from a remote host on the same exact hardware that mysql choked on and not cause any degredation of access times for any other user. I've been using my own PostgreSQL CDR backend -- patches submitted -- without a hitch for months now, peak load is ~1000 calls/hr. The server, an Athlon XP1800+ w/256MB and a 40gig IDE HD, also serves up data for a calling card service with PGSQL, and hosts 3-6 clients for data entry. Up for 100+ days. I was going to put * on it, too ;) I'm running PostgreSQL 7.2, but 7.3 does provide for PREPARE'd queries on a per-connection basis, although they are dropped after the connection is closed and I'm not sure what fancy footwork Perl-DBI does. --Chris ps. While on this thread, I'm attaching my PG CDR patches again. -- Chris Maj cmaj_hat_freedomcorpse_hot_info 0xC0051F6A 5EB8 2035 F07B 3B09 5A31 7C09 196F 4126 C005 1F6A cdr_postgres.patches.tar.gz Description: Binary data
Re: [Asterisk-Users] dialogic DIALOG/4
I think what Benjamin is trying to say is that the Dialogic D/4 is not capable of full duplex operation when reading/writing from the PCI bus. It's capable of full-duplex operations only when using it's own SC-BUS. Yes, you're right these cards have been used for VOIP. But, to achieve that you'll need to get special VOIP compression cards that talks to the D/4 via the SC-BUS, e.g. Dialogic DM-3 and some Audiocodes cards. Those cards are definitely not cheap. Tomaz Izanc wrote: how you mean board is not full duplex? ofcourse is full duplex it has been used for iptelephony long time ago. have you try this board in linux? you have drivers? i have two cards ;) Thomas Benjamin Miller wrote: This board is not full duplex and will not work with *. I have 3 of them :-( Buy some Digium boards. They will serve you much better and are easier. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 28, 2003 3:22 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dialogic DIALOG/4 Good question. This board works well with * ? Can I get the caller-id(ANI) using this board? Isamar On Wed, 28 May 2003, Tomaz Izanc wrote: hi .. anyone using dialogic isa board DIALOG / 4 ? Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF problems with Zaptel T100P
We've got an asterisk system hooked up to a number of telephones via a channel bank. [*]T100P---CAC(access bank)---Phones... What we are seeing is problems with DTMF tone detection on some (but not all) of the FXS ports. In our testing, we plugged a simple analog phone into each of the first 10 ports, called outside the syatem, and listed while the user depressed the 5 key for four seconds. What we expect to hear on the remote-end is four seconds of DTMF tone. That is what the local user hears. On some of the ports we do indeed hear four seconds of continuous DTMF. On one port, we only hear about 1/2 second of DTMF followed by 3.5 seconds of silence. On anohter port, we hear about 3.5 seconds of silence, followed by 0.5 seconds of DTMF. On another port, we hear 4 seconds of DTMF interrupted by blips of silence. When not playing DTMF tones, we can hear just fine. Has anyone else had these symptoms? Can anyone suggest a way to debug this? Thanks! -- Nick Eggleston Consultant Data Communications Consulting, Inc. 6320 Rucker Road, Suite E Indianapolis, IN 46220 317/726-0295 x18 317/202-2445 (fax) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ANI matching trouble
Hi. I need to send calls to different programs depending on where the call originates. For example, I need calls from San Diego (NPA 619 and 858) to to be routed differently than L.A. calls. I tried entries like: exten = 4044633/_619.,1,OurApp,sandiego-queue exten = 4044633/_858.,1,OurApp,sandiego-queue exten = 4044633/_213.,1,OurApp,losangeles-queue exten = 4044633/_.,1,OurApp,default-queue but it didn't seem to work. Every call went to the default queue. I also tried exten = 4044633/_619XXX,1,OurApp,sandiego-queue to no avail. It did work if I put a specific number in there: exten = 4044633/6193644788,1,OurApp,sandiego-queue but of course I can't list every possible number. What am I doing wrong? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANI matching trouble
I don't think you can use wildcards in the ANI matching areas, though I'd be happy if this were the case. You'll probably need to write an AGI that hands back an appropriate variable set to something that a Goto can parse. The use of wildcards in ANI matches would be darn handy, though. JT Hi. I need to send calls to different programs depending on where the call originates. For example, I need calls from San Diego (NPA 619 and 858) to to be routed differently than L.A. calls. I tried entries like: exten = 4044633/_619.,1,OurApp,sandiego-queue exten = 4044633/_858.,1,OurApp,sandiego-queue exten = 4044633/_213.,1,OurApp,losangeles-queue exten = 4044633/_.,1,OurApp,default-queue but it didn't seem to work. Every call went to the default queue. I also tried exten = 4044633/_619XXX,1,OurApp,sandiego-queue to no avail. It did work if I put a specific number in there: exten = 4044633/6193644788,1,OurApp,sandiego-queue but of course I can't list every possible number. What am I doing wrong? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many X100P's in a system..
On Thu, 22 May 2003, Robert Hajime Lanning wrote: Shared IRQs are just bad. Just the PC hardware does not have enough IRQs to handle all the devices we want to connect to them, nowadays. No, that's not the reason at all. Most systems are not short of IRQs. The reason is that standard PCI bus only has 4 interrupt lines. Shared IRQs are not bad at all. It's only bad if the card is badly designed. -Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANI matching trouble
I was just thinking that. Shouldn't this be a feature? I'm sure coding it would be a cut and past job. :) Another one for the TO-DO list Mark. :) Jamie On Wed, 28 May 2003 16:45:44 -0700 John Todd [EMAIL PROTECTED] wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* I don't think you can use wildcards in the ANI matching areas, though I'd be happy if this were the case. You'll probably need to write an AGI that hands back an appropriate variable set to something that a Goto can parse. The use of wildcards in ANI matches would be darn handy, though. JT Hi. I need to send calls to different programs depending on where the call originates. For example, I need calls from San Diego (NPA 619 and 858) to to be routed differently than L.A. calls. I tried entries like: exten = 4044633/_619.,1,OurApp,sandiego-queue exten = 4044633/_858.,1,OurApp,sandiego-queue exten = 4044633/_213.,1,OurApp,losangeles-queue exten = 4044633/_.,1,OurApp,default-queue but it didn't seem to work. Every call went to the default queue. I also tried exten = 4044633/_619XXX,1,OurApp,sandiego-queue to no avail. It did work if I put a specific number in there: exten = 4044633/6193644788,1,OurApp,sandiego-queue but of course I can't list every possible number. What am I doing wrong? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gastman windows build?
This build is about 200 years old. I've tried it and it actually doesn't work for me. CVS builds work fine though. Jamie On Wed, 28 May 2003 00:40:44 +1000 Shaun Ewing [EMAIL PROTECTED] wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* - Original Message - From: Dave Packham [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 28, 2003 12:03 AM Subject: [Asterisk-Users] Gastman windows build? is the win32 binary on * website suilt from the latest gastman code? Thanks Dave This is the latest prebuilt binary I could find: ftp://ftp.digium.com/pub/gastman/gastman-win32-0.2.1.zip --Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many X100P's in a system..
On Fri, 23 May 2003, Stephen R. Besch wrote: own card to determine if it is the interrupting hardware. In point of fact, this same strategy was possible on the ISA bus, it just wasn't used. I dont think level-triggered interrupts ever reliably worked on ISA, it was always edge. I never heard of any ISA cards that could share interrupts correctly. -Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM40B Problems
I have just a second case where after * had been up for a few weeks working perfectly, it suddenly stopped talking to all the extensions. The I4L was still working perfectly though... I tried to stop asterisk, rmmod the modules, and insmod, and re-start asterisk but it did this: Using /lib/modules/2.4.21-pre5/misc/zaptel.o Using /lib/modules/2.4.21-pre5/misc/wcfxo.o Using /lib/modules/2.4.21-pre5/misc/wcfxs.o /lib/modules/2.4.21-pre5/misc/wcfxs.o: init_module: No such device and on the console (it crashed, even CTRL-ALT-DEL didn't work) it had a lot of lines about register or something and at the end said something about Freshmaker or Bread config failed. (sorry, I should have copied some of it down, but I kinda needed the pbx to work again... Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP INVITE and ACK go to different ports
Title: SIP INVITE and ACK go to different ports Greetings, CVS 05/23/03. 10.50.4.140 is an * box. I see SIP INVITE to port 5060 and ACK (after OK) to port 32824. The log is attached. tcpdump shows 18:31:48.380006 10.50.4.140.5060 wmssqa02.webley.5060: udp 615 (DF) 18:31:48.390007 wmssqa02.webley.32824 10.50.4.140.5060: udp 331 (DF) 18:31:48.500018 wmssqa02.webley.32824 10.50.4.140.5060: udp 554 (DF) 18:31:48.540022 wmssqa02.webley.32824 10.50.4.140.5060: udp 540 (DF) 18:31:48.540022 10.50.4.140.5060 wmssqa02.webley.32824: udp 369 (DF) 17.1.1.2 Formal Description ...The ACK MUST be sent to the same address, port, and transport to which the original request was sent... I don't see configuration problems but cannot be 100% sure. I think it was working before (on a different * box). Thank you. Alex Zarubin z z Description: Binary data
Re: [Asterisk-Users] Who would use Asterisk SS7?
On 24 May 2003, Thilo Salmon wrote: The other issue is a legal one. In order to connect to the incumbent telco your equipment has to be certified. I believe unless quite a few of us get together, this one might be a real problem. Easy solution -- Have * talk to SS7-certified equipment. Cisco comes to mind. They have SS7 gateways that could talk with * as do many others. You can use * to cut out the expensive hardware and only use the bare minimum of the vendor's setup to talk to SS7. -Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 827-4v SIP config
From Reading http://www.cisco.com/univercd/cc/td/doc/pcat/827.htm that device is H323 and only has FXS ports. So any of the Cisco Docs on H323 would probably point you in the right direction. Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dave Packham Sent: Wednesday, May 28, 2003 17:49 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 827-4v SIP config has anyone ever used an 827-4v DSL router to do SIPtoPOTS conversion? how would I set up my 4 pots lines to be SIP extensions/phones? any ideas? Cisco site is a little lacking on sample configs. I would like to set up 3 of the ports for FXS analog sets and one port for FXO. convert all the analog to SIP and send it to a SIp gateway like FWD or Asterisk. Any thought or suggestions/examples would be a great help Thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Phantom Call.. T1 card too
Then why bring it up? What has any of this to do with the original topic? Jamie On 27 May 2003 12:35:57 -0500 Steven Critchfield [EMAIL PROTECTED] wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* I had a similar problem when there was timing problems with my T100P. You could also hear lots of popping and generally bad audio during the dial tone. After we fixed the timing problem, the audio was clear as could be, and the problems went away. Of course this doesn't fix a X100P as it is strictly analog. On Tue, 2003-05-27 at 12:03, Joe Antkowiak wrote: I've had the same thing happen, only on the single port T1 card and a channel bank, and one of the FXO channels also having a phone attached elsewhere... I just wound up putting that channel in a different context and running Exten = s,1,Hangup (I'm just using the line for outbound dialing) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tamas Levente Sent: Tuesday, May 27, 2003 11:45 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Phantom Call.. Same thing happened with me too. X100P. Same US tones Sometimes it gets into the voicemail too:)) And the voicemail record 3 minutes tone, after 1.5minutes it's service not available or something similar. Is there a fix for that? - Original Message - From: Mark Street [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 27, 2003 5:39 PM Subject: Re: [Asterisk-Users] The Phantom Call.. Funny, I just noticed this happening on my box with 2 X101P's installed and a phone connected to the same line as one of the X101P's. I pick up the phone after 1 ring, or call someone. After a minute or two * picks up the line and starts the greeting. I pull the plug on the asterisk box to continue the conversation. I just noticed it happening a couple of weeks ago. US dialtone here... On Tuesday 27 May 2003 08:13, Mark Spencer wrote: Could it be that the X100P is detecting the UK dial tone as a ring?? or Has anyone else had a similar problem when using the X100P/S100U combination?? It's possible there is *something* on the line that is confusing Asterisk into thinking a ring takes place. You might try adjusting the value of PEGCOUNT in wcfxo.c to a higher value (say, 10). -- Mark Street, D.C. Red Hat Certified Engineer Cert# 807302251406074 -- Key fingerprint = 3949 39E4 6317 7C3C 023E 2B1F 6FB3 06E7 D109 56C0 GPG key http://www.streetchiro.com/pubkey.asc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] REMOVE
No! You're outnumbered and trapped! On Wed, 28 May 2003 10:22:33 -0700 (PDT) Patrick Tabor [EMAIL PROTECTED] wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* --- Gary [EMAIL PROTECTED] wrote: I just had a thought, but haven't tried it yet we are able to include one conf file into another... this leads to the thought can we use the same file to include into iax.conf as well as sif.conf ? does the various differences actually have an effect of the application reading the file ? or another way will nonexisitant controlcommands have an adverse effect eg: in SIP.conf having nat=yes, what happens if that was in the same user stuff in iax.conf ?? Gary . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Calendar - Free online calendar with sync to Outlook(TM). http://calendar.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Australia users?
I've got an end-user in Australia that needs some gear .. I can ship it over there, but it'd be a bit of a pain to find adapters localized to AU, even though I know stuff is all 110/220 these days. Anyone know a good cheap place to buy a Cisco ATA186 and maybe a Netgear/etc dsl router/switch in Queensland (or close to)? Any advice would be helpful -d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANI matching trouble
exten = 4044633/_619.,1,OurApp,sandiego-queue exten = 4044633/_858.,1,OurApp,sandiego-queue exten = 4044633/_213.,1,OurApp,losangeles-queue exten = 4044633/_.,1,OurApp,default-queue but it didn't seem to work. Every call went to the default queue. Take out the _. rule and just leave it 4044633 and it should work fine. Not postive the _ is required on matching the callerid part, but honestly i just don't remember. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many X100P's in a system..
No, that's not the reason at all. Most systems are not short of IRQs. The reason is that standard PCI bus only has 4 interrupt lines. Shared IRQs are not bad at all. It's only bad if the card is badly designed. All the Zaptel hardware *should* be able to share IRQ's. Each of the cards has a register that can tell us if the IRQ was real or not. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP INVITE and ACK go to different ports
Be sure you do *not* have NAT mode turned on. Mark On Wed, 28 May 2003, Alex Zarubin wrote: Greetings, CVS 05/23/03. 10.50.4.140 is an * box. I see SIP INVITE to port 5060 and ACK (after OK) to port 32824. The log is attached. tcpdump shows 18:31:48.380006 10.50.4.140.5060 wmssqa02.webley.5060: udp 615 (DF) 18:31:48.390007 wmssqa02.webley.32824 10.50.4.140.5060: udp 331 (DF) 18:31:48.500018 wmssqa02.webley.32824 10.50.4.140.5060: udp 554 (DF) 18:31:48.540022 wmssqa02.webley.32824 10.50.4.140.5060: udp 540 (DF) 18:31:48.540022 10.50.4.140.5060 wmssqa02.webley.32824: udp 369 (DF) 17.1.1.2 Formal Description ...The ACK MUST be sent to the same address, port, and transport to which the original request was sent... I don't see configuration problems but cannot be 100% sure. I think it was working before (on a different * box). Thank you. Alex Zarubin z ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] vmail.cgi contexts
Mark Spencer wrote:how is this passed to the script (I know 0 perl) I would have though something like ./vmail.cgi?context=mycontext but that makes the script barf try logging in as [EMAIL PROTECTED] Same results for me: 1. I can see things in my INBOX, which is in the default context, but the Old messages that Allison can see don't seem visible to the web interface. 2. New mails coming in show up right away in the INBOX. 3. Clicking on the message (in INBOX) yields the error message I sent to the list earlier today. 4. The same mails sent as attachments play just fine. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] vmail.cgi contexts
yeah logging in like that works, however the stored mail seems to be in the default context not the one the mailboxes are really in. eg: if I move my mailbox entry to default, I see my mail when I login, if I leave in the context it was always in, I see no mail after I login. is this some sort of growing pain with voicemail / voicemail2 ? should I delete and recreate all the mailboxes and then switch to app_voicemail2 from app_voicemail and then the contexts will be obeyed when recording as well ? can the script be modified to accept parameters if it does not now so that I can have a separate url for each context, and not require users to know what context they are in, just how to link to the right url to login for their context. Is there someone that knows enough perl to clear this up since it looks to me that is how the script is actually supposed to work. At 11:02 PM 5/28/2003 -0500, you wrote: Mark Spencer wrote:how is this passed to the script (I know 0 perl) I would have though something like ./vmail.cgi?context=mycontext but that makes the script barf try logging in as [EMAIL PROTECTED] Same results for me: 1. I can see things in my INBOX, which is in the default context, but the Old messages that Allison can see don't seem visible to the web interface. 2. New mails coming in show up right away in the INBOX. 3. Clicking on the message (in INBOX) yields the error message I sent to the list earlier today. 4. The same mails sent as attachments play just fine. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANI matching trouble
Shouldn't the priority also be different for each entry? This would make it: exten = 4044633/_619.,1,OurApp,sandiego-queue exten = 4044633/_858.,2,OurApp,sandiego-queue exten = 4044633/_213.,3,OurApp,losangeles-queue exten = 4044633,4,OurApp,default-queue This should work I would think. Give it a shot, if it doesn't remove the _'s from the ANI pattern. Jamie On Wed, 28 May 2003 22:39:19 -0500 (CDT) Mark Spencer [EMAIL PROTECTED] wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* exten = 4044633/_619.,1,OurApp,sandiego-queue exten = 4044633/_858.,1,OurApp,sandiego-queue exten = 4044633/_213.,1,OurApp,losangeles-queue exten = 4044633/_.,1,OurApp,default-queue but it didn't seem to work. Every call went to the default queue. Take out the _. rule and just leave it 4044633 and it should work fine. Not postive the _ is required on matching the callerid part, but honestly i just don't remember. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many X100P's in a system..
On Wed, 28 May 2003, Mark Spencer wrote: No, that's not the reason at all. Most systems are not short of IRQs. The reason is that standard PCI bus only has 4 interrupt lines. Shared IRQs are not bad at all. It's only bad if the card is badly designed. All the Zaptel hardware *should* be able to share IRQ's. Each of the cards has a register that can tell us if the IRQ was real or not. I would hope so.. but didnt someone say the x100p docs stated the card required its own non-shared irq? BTW does the x100p use dma? There are some motherboards with specific slots which cannot do dma (eg Abit BP6). -Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Who would use Asterisk SS7?
On Thu, 2003-05-29 at 02:36, [EMAIL PROTECTED] wrote: On 24 May 2003, Thilo Salmon wrote: The other issue is a legal one. In order to connect to the incumbent telco your equipment has to be certified. I believe unless quite a few of us get together, this one might be a real problem. The SS7 equipment from Lucent, Nortel, Alcatel are likely already certified with the carrier you want to link to. If not, they are happy to make that happen for you. Also, in Europe you will not get an SS7 link to a carrier unless you are a licensed carrier yourself. Easy solution -- Have * talk to SS7-certified equipment. Cisco comes to mind. They have SS7 gateways that could talk with * as do many others. You can use * to cut out the expensive hardware and only use the bare minimum of the vendor's setup to talk to SS7. -Dan Whatever * is able to cut out, you still need a serious telco budget to actually get the SS7 solution. Given customer requirements, you pass the $500,000 mark in the blink of an eye. And that does not include a service contract for the kit for as long as it is in service. This may still make sense to some though. If I were to make such investments I would: * become a licensed carrier * install SS7 interconnection gear with all major carriers in the designated area * negotiate termination service fees as high as possible * get tons of traffic to my network by offering ??? to customers * profit! Suggestions on the ??? part are most welcome :) Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting up fax on *
Hello All, I am using an E100P card on a PRI line. I need to setup a FAX extension. Can somebody help me please? Marco
Re: [Asterisk-Users] What is the going rate for the Snom 100 in the UK?
That's about right, inlcuding postage. - Original Message - From: nathan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 12:44 PM Subject: [Asterisk-Users] What is the going rate for the Snom 100 in the UK? Hi All, What is the going rate for the Snom 100 in the UK? I've found a couple of suppliers with prices around the £170 (exc vat) mark. Regards, Nathan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF problem, need help!
Very weird problem, got a channelized T1 with SBC. There is a toll free number that points to a local number on that T, when the local number is called, * recognizes dtmf tones, when the toll free number is called, * does not pick up dtmf.. any thoughts? lizardbox ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the going rate for the Snom 100 in the UK?
Nathan, Get in touch with www.provu.co.uk ask to speak to Tim, and tell him you heard from me (Andy Powell) that they had a deal running where you could get Snom 100's for 140 gbp... HTH Andy *** REPLY SEPARATOR *** On 29/05/2003 at 12:44 nathan wrote: Hi All, What is the going rate for the Snom 100 in the UK? I've found a couple of suppliers with prices around the £170 (exc vat) mark. Regards, Nathan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
