Re: [Asterisk-Users] Re: Tormenta ISA E1 card
Hi John, [EMAIL PROTECTED] wrote: Hi!Steve, Glad to receive your message. Could you tell me if the card can work properly in Asterisk? and what is the different bettween the Tormenta ISA E1 card and the ISA T1 card (I refer to the hardware). Thanks. john The differences between the T1 and E1 versions are the frequency of the two VCXO crystals, and part number of the two framer chips. Change those to the E1 type and the card functions OK as a clock slave. A hardware limitation means it will not perform properly as a clock master. As a T1 card clock master mode is OK. As an E1 card it isn't. Most people need to slave most of the time to the great atomic clock in the PSTN, so this isn't such a big limitation. I never tried the card with Asterisk, but it should work for CTR4/EuroISDN. I don't think there is support for any other protocols. It works OK for the things I do with it, using libPRI and not using Asterisk. The differences between Steve Underwood writes: [EMAIL PROTECTED] wrote: Has any one used it? Yes! Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 32 E1 or 64 E1 Configuration ?
At 20:16 4-3-2003 +0900, you wrote: Is it possible to support 32 or 64 E1 in a linux box with Wildcard E400P board ? I'd like to make large scale PPS system. Hmm, 8 to 16 PCI slots ? That'll be a challenge. I think the current practical maximum would be 12 or maybe 16 ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 32 E1 or 64 E1 Configuration ?
On Tue, Mar 04, 2003 at 02:35:04PM +0100, Florian Overkamp wrote: At 20:16 4-3-2003 +0900, you wrote: Is it possible to support 32 or 64 E1 in a linux box with Wildcard E400P board ? I'd like to make large scale PPS system. Hmm, 8 to 16 PCI slots ? That'll be a challenge. I think the current practical maximum would be 12 or maybe 16 ? A better question might be, why do you need them all to be in a single box? -SteveK -- Steve Kann - Chief Engineer - 520 8th Ave #2300 NY 10018 - (212) 533-1775 HorizonLive.com - collaborate . interact . learn The box said 'Requires Windows 95, NT, or better,' so I installed Linux. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX calls drop suddenly?
I've got asterisk working as a PSTN gateway between two sites. The IP connection is pretty good and bandwidth exceeds 640Kbps all the time. I'm using X100P hardware on 300Mhz and 333MHz systems. One of them has a sound card, but the slower system does not have a sound card. Both are connected to PBXes and I have busydetect=yes Do the calls drop after they are bridged to PSTN ?? If so busydetect=yes or callprogress=yes is suspect the logic on these flag can some times misinterpret the channel status try to hang up a channel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] a problem with MeetMe
Hi, I try the application MeeMe but i Have a problem when I call a conference. It show me : Unable to open pseudo channel Does anyone can help me ? regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error in tor2
On Mon, 3 Mar 2003 07:15:03 +0100 Victor Sanchez [EMAIL PROTECTED] wrote: when i have used modprobe tor2 i halt my PC. and i need to reset it. What gcc You are using? I found out that after building tor2 with new gcc-3.2 it halts the system The solution is: in Makefile edit CC variabe and set it to gcc-2.95 or something like this... (Your old gcc) make clean; make - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED]; Greg Vance [EMAIL PROTECTED] Sent: Friday, February 28, 2003 6:52 PM Subject: Re: [Asterisk-Users] error in tor2 Use modprobe instead of insmod. If you use insmod then you have to first insmod zaptel. regards Martin On Fri, 28 Feb 2003, Victor Sanchez wrote: i have error in install module of tor2. but it compile good. what happen ? ivr2:/usr/src/zaptel # make clean; make install rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN DALONE_ZAPATA -c zaptel.c cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATAmakefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN DALONE_ZAPATA -c tor2.c gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN DALONE_ZAPATA -c torisa.c gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN DALONE_ZAPATA -c wcusb.c gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN DALONE_ZAPATA -c wcfxo.c gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN DALONE_ZAPATA -c wcfxs.c gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTAN DALONE_ZAPATA -c ztdynamic.c gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototype s -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include usr/src/linux/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd
Re: [Asterisk-Users] 32 E1 or 64 E1 Configuration ?
But you can connect several asterisk boxes as one system. regards Martin On Tue, 4 Mar 2003, Sphyrna wrote: NO, THE ASTERISK HAS A PRATICAL LIMIT OF 8 E1S CURRENTLY. THE I/O ERRORS STOP EVERYTHING - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, March 04, 2003 8:35 AM Subject: Re: [Asterisk-Users] 32 E1 or 64 E1 Configuration ? At 20:16 4-3-2003 +0900, you wrote: Is it possible to support 32 or 64 E1 in a linux box with Wildcard E400P board ? I'd like to make large scale PPS system. Hmm, 8 to 16 PCI slots ? That'll be a challenge. I think the current practical maximum would be 12 or maybe 16 ? Florian ___ 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] a problem with MeetMe
Uncomment ztdummy from zaptel/Makefile make clean ; make install modprobe ztdummy. Restart asterisk, all fixed. Rattana BIV wrote: Hi, I try the application MeeMe but i Have a problem when I call a conference. It show me : Unable to open pseudo channel Does anyone can help me ? regards Rattana ___ 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] Brooktrout T1/E1 cards and Asterisk
I have wanted to setup an Asterisk system for several months but have been unable to do so because I lacked the funds for a T1 card to connect my channel bank to the server. Recently, I acquired several Brooktrout T1/E1 interface (PRI-PCI48VC/PRI-PCI64V-C) cards for the PCI bus. According to the Brooktrout web site http://www.brooktrout.com/products/netaccess_pri_pci/specs.html Linux drivers are available for these cards but I dont as yet know if they are open source. Brooktrout makes several references to the software development package that comes with the cards (I only received the hardware) so I think there is a good chance that the drivers are open source. Assuming that I can obtain the Brooktrout Linux drivers and they are open source, can I use these cards with Asterisk or would Asterisk specific drivers be required? Regards, George Bean
[Asterisk-Users] Mailing lists and spam
Hello, everyone. I've proven to myself, via throw-away addresses, that individuals and robots sift thru archives and mailing list subscription lists, gathering addresses to sell on CD's to spammers. No list is immune, especially if it or its archives are available over the web. I've been spammed within a very short while of my address appearing on a posted message. If you go to http://lists.digium.com you will find that you are probably on all of the 3 lists specified: announce, users, and dev. I suggest you all take a moment, and Visit subscriber list for each one, and click on your own email, and have the password for each one sent to you. Then, edit your options; some may wish digests; I suggest all of you Conceal yourself from subscriber list. Because, in this case, even if you don't post, you'll get spammed. It may already be too late, but, better late than never, right? It'd be nice if the mailman guys (and most other list server software), would disguise or hide people's emails in the letters sent out and in the archives. There are several methods for doing this, and perhaps a rotating mix of them could be used. The collection software will probably easily decode all these methods soon, if they haven't already. Perhaps mailman someday will include some sort of mediation capability where, if you wish to go offline on a subject with someone, you will be able to post to an alias on the list server, which will forward the message to you. If it only does this for list members, these aliases would be useless to spammers. And, make sure to leave out your signatures if you want to keep your spam levels low. Right now, it doesn't much matter, as the from address is included in the header for each message... but, then again, I guess web robots could collect phone numbers for telemarketers and addresses for postal marketing, couldn't they? Wouldn't that be worth something? Mark-- I suggest that you hide the subscription lists from the web pages... just as a service to your list members. Also, if monthly reminders are sent to all the members, they may find it easier to access their settings on the 3 mailing lists. murf -- signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Brooktrout T1/E1 cards and Asterisk
On Tue, 2003-03-04 at 10:31, George Bean wrote: I have wanted to setup an Asterisk system for several months but have been unable to do so because I lacked the funds for a T1 card to connect my channel bank to the server. Recently, I acquired several Brooktrout T1/E1 interface (PRI-PCI48VC/PRI-PCI64V-C) cards for the PCI bus. According to the Brooktrout web site http://www.brooktrout.com/products/netaccess_pri_pci/specs.html Linux drivers are available for these cards but I dont as yet know if they are open source. Brooktrout makes several references to the software development package that comes with the cards (I only received the hardware) so I think there is a good chance that the drivers are open source. Assuming that I can obtain the Brooktrout Linux drivers and they are open source, can I use these cards with Asterisk or would Asterisk specific drivers be required? Last time I dealt with Brooktrout their drivers where definately not open source. That is why they make reference to the development package. Althought they probably have been upgraded, they only where intending to release a driver against a RH production kernel. Basically the just of this is, you will run into licensing issues if you write a channel driver and then distribute it as your copy of asterisk is GPL, and the library and drivers for the card are decidedly not GPL. You may well be able to place those cards on Ebay and with some luck get more than the cost of Mark's T1 cards. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Mailing lists and spam
On Tuesday 04 March 2003 10:58 am, Steve Murphy wrote: I've proven to myself, via throw-away addresses, that individuals and robots sift thru archives and mailing list subscription lists, gathering addresses to sell on CD's to spammers. No list is immune, especially if it or its archives are available over the web. I've been spammed within a very short while of my address appearing on a posted message. One comment on this: one of NLUG's programmers has created a very workable solution to this problem. You can browse the NLUG archives here: http://www.nlug.org/mail/. All email addresses are obscured from bots by replacing them with PNG images. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] devkit gone?
I was considering buying a devkit, bit it looks like it's not longer available. Will there be a replacement devkit? If not, where to buy a reasonable priced channel bank? Which channel bank was included in the devkit and where can it be bought? (btw I live in Holland) Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] devkit gone?
On Tue, 2003-03-04 at 11:52, Chris Wetemans wrote: I was considering buying a devkit, bit it looks like it's not longer available. Will there be a replacement devkit? If not, where to buy a reasonable priced channel bank? Which channel bank was included in the devkit and where can it be bought? (btw I live in Holland) Ebay is always a good choice for looking for channel banks. It is also about the only place to be looking for them if you are doing this as either a feasibility study, or as a home system. At the time you wan't to roll out as a business system that requires more than one, you will want to get one from a reputable dealer, maybe even from Digium. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR Output
I am pushing all the cdr info to a MySQL database on a separate machine. I have noticed that the duration times for all calls are recorded in seconds, by Asterisk. Is there a way to set the recorded call duration to a decimal representation of minutes? ie 90 sec = 1.5 min. My extraction process would be a little simpler if the data dumped into the database were minutes and not seconds. Thanks, Matthew S. Hill Systems Engineer CHG Companies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Output
On Tuesday 04 March 2003 01:12 pm, Steven Critchfield wrote: On Tue, 2003-03-04 at 12:57, Matthew S. Hill wrote: I am pushing all the cdr info to a MySQL database on a separate machine. I have noticed that the duration times for all calls are recorded in seconds, by Asterisk. Is there a way to set the recorded call duration to a decimal representation of minutes? ie 90 sec = 1.5 min. My extraction process would be a little simpler if the data dumped into the database were minutes and not seconds. It is opensource, and all you are looking at is changing a printf statement to output duration/60 and change the type so it can handle the decimal. Just a friendly jab, not a start to a flame war. If you where on postgres you could just create a view that showed duration as duration/60. I believe there is even another function in postgres that would allow you to round it to the precision you are comfortable with seeing. Not to mention it would have been possible to create a trigger on insert that did the conversion for you so that by the time the data was stored it was the way you wanted it. Or just change the field in the SELECT clause to: duration/60 AS duration -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cellsocket update
When I posted my last message about the cellsocket, I hadn't thought to try incoming calls. (I bought the cellsocket to use only for outgoing calls.) Unfortunately, I cannot get the cellsocket to work for inbound calls. CallerID does not work even though I have a GSM phone. The cellsocket answers the phone and then starts ringing any attached phones after answering. Most importantly, levels are so low once the call is completed that neither party on the call can hear the other. Very strange, since outbound calls work so well. Hopefully I just have a defective cellsocket. Regards, Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] S100U == DEAD !
Picking up the phone now results in a brief dialtone followed by bursts of random static and crackling noises and then a second or two of fast-busy followed by silence and more crackling noises. I've tried the hardware attached to a couple of different machines as well as a couple of different phones. Having eliminated the machine itself and the phones I can only conclude that the S100U is the cause. Ours always was problematic being subject to frequently disconnections which I traced to an abnormally high sensitivity to static-electric shock. I can only assume that it got zapped once two many times and now no longer works. I'm not really sure what to do now. We can't afford a channel bank and it hardly seems sensible to purchase another S100U. Your S100U has to still be under warranty since we haven't even been selling them for two years yet :) We'll be happy to trade it out for you. Contact Greg ([EMAIL PROTECTED]) or support ([EMAIL PROTECTED]) and be sure to tell them you're down so they'll send it overnight. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Patch to avoid sporadic hangup during normal conversation
Hi, Mark. I'm posting a little patch that permits to set the busycount value in dsp (via chan_zap), for systems that have "busydetect=yes" in zapata.conf and have noticed sporadic hangup, caused by incorrect cadence detection during normal conversation. With this patch, it's possible to increase the busycount from 3 (default) to 5 (max, becauseDSP_HISTORY=5), simplyadding a "busycount=5" linein zapata.conf, andleaving the dsp.c code unchanged. In my system, buycount=5 works great! I'd like to see this patch commited if you think it's useful. best regards -Eduardo chan_zap_busycount.diff Description: Binary data
Re: [Asterisk-Users] CDR Output
I tested both MySQL and PostgreSQL under various loads, Actually got paid to do it. In the simple case where one process just stuffs data into the database with INSERT MySQL was about 4x faster in terms of INSERT/second However if you have five or six processes that are doing lots of transactions, mixed INSERT and SELECT, MySQL grinds to a halt The trouble is locking. MySQL only lets one process at a table at a time so everyone else is shut out waiting. On a large PBX with high call volume you can't afford to have a table locked for 10 or 30 seconds while some guy does an ad-hoc join. PostgreSQL is more stable but can only do a few hundred INSERT per second on PC hardware vs. well over 1000/sec for MySQL. --- Matteo Brancaleoni [EMAIL PROTECTED] wrote: I agree. postgres could be too slow if used into a big system and you can't afford a rather good machine. mysql is very fast and simple. and I don't see where's the problem into the extraction... a simple division could be done in any language, with any program... matteo. Il mar, 2003-03-04 alle 20:27, [EMAIL PROTECTED] ha scritto: Why would anyone use such a big axe for a small problem, a trigger to do simple math. Bah. MySql is perfectly suitable for CDR logging on any practical phone system short of telco main office eqpt. Karl Putland wrote: On Tue, 2003-03-04 at 11:57, Matthew S. Hill wrote: I am pushing all the cdr info to a MySQL database on a separate machine. I have noticed that the duration times for all calls are recorded in seconds, by Asterisk. Is there a way to set the recorded call duration to a decimal representation of minutes? ie 90 sec = 1.5 min. My extraction process would be a little simpler if the data dumped into the database were minutes and not seconds. Write a trigger for on insert or a view that converts sec-min. Oh wait your using MySQL. Use PostgreSQL ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Tax Center - forms, calculators, tips, more http://taxes.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ringing
Hi All... Can Asterick detect distinctive ringing on a POTS line and answer with different configurations? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: PRI costs in US
- Original Message - From: Brian Johnson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, March 04, 2003 11:46 AM Subject: RE: [Asterisk-Users] OT: PRI costs in US I'm right here ... where are you? Right about here.. Just kidding, I'm in Stratford, Ontario, Canada I'm in Lethbridge Alberta. As an aside, I looked at switching to your company for internet access from Bell HSE but switched to istop.com instead. My company? What made you choose istop.com? We have a branch office in Kemptville (currently using Bell HSE - but only until I get time to get them switched to something else) that I'd like to connect to here via VoIP I have ordered two digium dev lite kits to try to connect the two (with asterisk) I'm looking at doing something similar, but I can't my asterisk box to work right. For two small offices, it's questionable whether it's worth the effort ... but it's a learning experience that will hopefully pay off by easing the opening of additional branch offices and providing an integrated whole It might be. Let me know how that goes. _ Kris Cote Director of Internet Engineering StormServer Internet / Storm3 Communications [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Storm3 Communications Sent: Monday, March 03, 2003 22:39 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT: PRI costs in US Brian, where are you? _ Kris Cote Director of Internet Engineering StormServer Internet / Storm3 Communications [EMAIL PROTECTED] - Original Message - From: Brian Johnson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 03, 2003 7:02 AM Subject: Re: [Asterisk-Users] OT: PRI costs in US I don't think I'd be stepping on toes if I told you guys we're paying about $35/month per analog line with callerid, line hunting, call waiting, etc, etc (basically the works minus voice mail) through ATT That's Canadian so you'll have to work out the currency conversion ie $1 USD = $1.65 CND Jon Pounder ([EMAIL PROTECTED]) wrote*: Hey I hear ya, I am in much the same situation. PRI or channelized T1 is not really a cost effective solution until you are over at least 10 lines, and need features like the DID or callerid, hunt, etc. that you would pay a few bucks more each per line so the per line analog cost would have been driven up over $50/line. My thoughts were BRI's as well since they work out to about $40/B channel, and you can add and delete in fairly inexpensive groups of 2. You get all the CLASS features of the PRI, but without having to buy a ton of lines to get them. BRI channel bank is one option I guess, but fairly rare and expensive. I would love to find a bri to pri T1 multiplexer at a reasonable cost. I have seen a few for sale minus the BRI cards so no deal on them. The ones I have seen do much more than is actually needed here so likely the cost is excessive when bought new. I am still not really clear why the isdn modems that seem so common can't/won't work in this situation. Is it simply an issue of software does not exist ? Does the modem do something other than simply dump the composite digital datastream into the serial port ? I am familar with how a channelized T1 can be output/input via a v35 HSSI port - Does this same concept not extend to the BRI isdn and a regular speed serial port ? Is the issue that a conventional serial port is topped out at 115k and bri would be 64+64+16 ? If so, how does the modem work in bonded bearer channel mode for data ? Anyone got answers ? At 06:40 AM 3/2/2003 -0500, you wrote: Hello! Several of my customers would like to add a backup to their Internet connection. ISDN is a good solution: decently fast for a dial-up-type connection, yet still faily affordable. While I was at it, I decided to look at a couple of more creative telephone service options to possibly improve their service or lower costs at the same time. These customers range from having just a couple of POTS lines without a key system up to as many as 18 POTS lines into systems such as ATT Merlin or Nortel Norstar systems. My first thought on the higher end was a PRI line. However, the cost seems *very* prohibitive. The average cost for a PRI line was $550/month, just for dial tone! I've heard others say that PRI becomes cost effective in the 8 line range, but the cost for the office with 14 lines ($25/line * 14 lines) is only $350. It would take 22 lines before the PRI would equal the cost of individual POTS lines! As an aside, I did a survey of several of my customers with 10-30 employees. Most of them have POTS lines
[Asterisk-Users] Tiger Jet card X100P
To the guy who wanted to know if the Tiger 560 card was a X100P - I don't think so. Here's the lspci -vv from a fairly new X100P: 00:09.0 Communication controller: Tiger Jet Network Inc. Model 300 128k Subsystem: Unknown device 8085:0003 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: 64 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 11 Region 0: I/O ports at 7000 [size=256] Region 1: Memory at e4001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=0mA PME(D0+,D1-,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Hope this helps. PS It's interesting to see the difference between this and an older X100P: 00:0a.0 Communication controller: Motorola: Unknown device 5608 Subsystem: Motorola: Unknown device 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: 64 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 5 Region 0: I/O ports at dc00 [size=256] Region 1: Memory at e8002000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=0mA PME(D0+,D1-,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Motorola is 1057 so the PCI ID is 1057:5608 for the older card, and 8085:0003 for the newer card. -- Jim Ockers ([EMAIL PROTECTED]) Contact info: please see http://www.ockers.net/ Fight Spam! Join CAUCE (Coalition Against Unsolicited Commercial Email) at http://www.cauce.org/ . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CERT advisory (SIP)
In a quick search of the list archives, I found no mention of the recent CERT advisory concerning vulnerabilities in some implementations of the SIP protocol (i.e. whether or not * users were impacted by it, and if so, to what extent and/or in what configurations), so I figured it would be worthwhile to toss the question out there ... Link: http://www.cert.org/advisories/CA-2003-06.html As someone in the early stages of investigating *'s potential usefulness for both my own needs and those of my clients - and as one who readily admits of possessing little knowledge of any but the most rudimentary aspects of telephony and CTI - I would be grateful if someone familiar with the nuts and bolts of * and SIP could provide a brief assessment on this point to the list. TIA! -- Bill Mullen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
