[asterisk-users] Hardware Server Configuration/8 or 4 port PRI Card
Hi, Can someone please recommend me the Hardware Server Configuration/8 or 4 port PRI Card to make Outbound Call at the rate of around 320 outbound Calls/min ? Thanks and Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware recommendation needed
Hi, We are planning to set up a prototype IVR system in Urdu language using Asterisk. For speech recognition, we will be using our own engine built using Sphinx, and for text to speech synthesis (for run time generation of responses based on user queries), we have a system for Urdu built in C++ that can be used as an API. My question is, can the Linksys SPA400 telephony gateway be used with Asterisk to develop the IVR system described? And if not, what other options should we explore? We have looked into the following options: 1. The Linksys SPA400 telephony gateway: we have used this previously with Trixbox to collect speech data over a telephone line, but we are not sure if it would support an IVR system such as the one described. 2. Digium telephony cards: we may have to rule these out because of cost issues if we have other options available. Also, most of these seem to be internal cards, and we would prefer to use an external device due to some equipment related limitations. 3. Dialogic cards: these were also ruled out due to cost issues. 4. We have also looked at Asterisk documentation and it seems that an IVR system setup should be possible with any of these devices, but could find no recommendations for IVR applications in particular. Any suggestions will be much appreciated. -- Thanks regards, Huda Sarfraz www.cle.org.pk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware Compatibility HP Proliant - Sangoma PCI Express
Hi to all, I am in the process of setup a new asterisk server, I think in the HP Proliant ML350 G6 Server (aprox. 100 SIP Users), and Sangoma A102DE Card. The specs of the Proliant (HP PART 487932-001) about PCI are the next. 1 ( 1 ) x PCI Express 2.0 x16 ( x8 mode ) , 1 ( 1 ) x PCI Express 2.0 x8 ( x8 mode ) , 4 ( 3 ) x PCI Express 2.0 x8 ( x4 mode ) The question is, if the card is compatible with the PCI slots in the server? And. If there is a known issue with this combination? Thanks a lot. Ricardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Compatibility HP Proliant - Sangoma PCI Express
Am 21.10.2010 19:30, schrieb Ricardo Melendez: Hi to all, I am in the process of setup a new asterisk server, I think in the HP Proliant ML350 G6 Server (aprox. 100 SIP Users), and Sangoma A102DE Card. The specs of the Proliant (HP PART 487932-001) about PCI are the next. 1 ( 1 ) x PCI Express 2.0 x16 ( x8 mode ) , 1 ( 1 ) x PCI Express 2.0 x8 ( x8 mode ) , 4 ( 3 ) x PCI Express 2.0 x8 ( x4 mode ) The question is, if the card is compatible with the PCI slots in the server? And. If there is a known issue with this combination? Thanks a lot. Ricardo Hello, i dont have a ML360 but several DL380 G5 with Sangoma A108D cards in it and i dont have any problems with this even if all 240 channels are used. best regards stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware clock drift and CDR
--- On Sun, 4/25/10, Gordon Henderson gordon+aster...@drogon.net wrote: Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. Hi Gordon, Are you sure about this? ntpd is a daemon and adjusts the time in a continuous manner. ntp-client or ntpdate or whatever are one-time clients that reset the system clock. I don't see why an ntp-client can't be run while ntpd is working (it shouldn't be necessary but may come in handy when the time difference is big and ntpd refuses to sync). Anyway, I've noticed that my ntpd log messages don't say anything when trying to sync to my Windows PDC LAN time server. Curiously, ntp-client DOES sync to this Windows server. So I decided to sync to pool.ntp.org and now I see syslog messages that actually show that the system time gets adjusted by ntpd. I'd rather sync to my LAN time server but this is off-topic on this ML. How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Who cares. Just fix ntpd then your worys are gone. Well, I still have doubts about that. I could look at * source code but I'd rather hear from someone here. My ntp log shows this: 26 Apr 13:06:30 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:21:24 ntpd[534]: time reset +2.318647 s 26 Apr 13:21:44 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:37:46 ntpd[534]: time reset +2.325417 s 26 Apr 13:38:06 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:54:11 ntpd[534]: time reset +2.327974 s 26 Apr 13:55:19 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:09:16 ntpd[534]: time reset +2.177572 s 26 Apr 14:10:08 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:26:07 ntpd[534]: time reset +2.357017 s That kind of scares me because if I'm not mistaken it means that about every 20 seconds, my ntpd adjusts the system time by about 2 seconds forward. So my clock is going back 2 seconds every 20... That's a significant drift. And it would definitely make a difference in my CDR records IF Asterisk were to compare the start and end system times. Should I worry about this? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware clock drift and CDR
On 4/26/2010 7:33 AM, Vieri wrote: --- On Sun, 4/25/10, Gordon Hendersongordon+aster...@drogon.net wrote: Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. Hi Gordon, Are you sure about this? ntpd is a daemon and adjusts the time in a continuous manner. ntp-client or ntpdate or whatever are one-time clients that reset the system clock. I don't see why an ntp-client can't be run while ntpd is working (it shouldn't be necessary but may come in handy when the time difference is big and ntpd refuses to sync). Anyway, I've noticed that my ntpd log messages don't say anything when trying to sync to my Windows PDC LAN time server. Curiously, ntp-client DOES sync to this Windows server. So I decided to sync to pool.ntp.org and now I see syslog messages that actually show that the system time gets adjusted by ntpd. I'd rather sync to my LAN time server but this is off-topic on this ML. How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Who cares. Just fix ntpd then your worys are gone. Well, I still have doubts about that. I could look at * source code but I'd rather hear from someone here. My ntp log shows this: 26 Apr 13:06:30 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:21:24 ntpd[534]: time reset +2.318647 s 26 Apr 13:21:44 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:37:46 ntpd[534]: time reset +2.325417 s 26 Apr 13:38:06 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:54:11 ntpd[534]: time reset +2.327974 s 26 Apr 13:55:19 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:09:16 ntpd[534]: time reset +2.177572 s 26 Apr 14:10:08 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:26:07 ntpd[534]: time reset +2.357017 s That kind of scares me because if I'm not mistaken it means that about every 20 seconds, my ntpd adjusts the system time by about 2 seconds forward. So my clock is going back 2 seconds every 20... That's a significant drift. And it would definitely make a difference in my CDR records IF Asterisk were to compare the start and end system times. Should I worry about this? Vieri If it is NTP that you are worried about, you can see what your servers look like by doing an ntpq -p which should show you the clocks in the pool, which ones it is using etc. Example: remote refid st t when poll reach delay offset jitter == *clock.trit.net 192.12.19.20 2 u 385 512 377 50.2203.094 0.558 +blue.nonexiste. 91.189.94.4 3 u 339 512 377 49.154 -16.663 4.596 +216.45.57.38216.218.254.202 2 u 155 512 377 50.2381.419 0.481 With my system synchronized to clock.trit.net. That is off my master clock, and everything else is synced to it by +/- 1 second. To fix this the easiest way, that I have seen at least, stop ntpd, do an ntpdate to your primary chosen clock (ntpdate clock.trit.net in my example) and restart ntpd and verify that your clock is sync'ed accurately. Also verify that it isn't hitting your hardware dummy clock in ntpd.conf, and if it is, and you can't force it out, you can remove it temporarily. Your CDR's will be screwy in terms of timestamps based on the system time constantly changing, as well as your log files being slightly off, and if you are doing anything remote to another box in terms of logging or database, it will be even more screwy. ~Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware clock drift and CDR
On Mon, 26 Apr 2010, Vieri wrote: --- On Sun, 4/25/10, Gordon Henderson gordon+aster...@drogon.net wrote: Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. Hi Gordon, Are you sure about this? Yes. ntpd is a daemon and adjusts the time in a continuous manner. ntp-client or ntpdate or whatever are one-time clients that reset the system clock. I don't see why an ntp-client can't be run while ntpd is working (it shouldn't be necessary but may come in handy when the time difference is big and ntpd refuses to sync). ntp binds to the ntp port (123) and prevents anything else binding to it, or listening on it - which ntpdate needs to do. Example here: Desktop is running ntpd: yakko:/home/gordon# ps ax | fgrep ntp 22064 ?Ss 0:14 /usr/sbin/ntpd -p /var/run/ntpd.pid -u 106:107 -g 30340 pts/29 R+ 0:00 fgrep ntp I try to run ntpdate: yakko:/home/gordon# ntpdate essen.drogon.net 26 Apr 14:20:47 ntpdate[30341]: the NTP socket is in use, exiting Anyway, I've noticed that my ntpd log messages don't say anything when trying to sync to my Windows PDC LAN time server. Curiously, ntp-client DOES sync to this Windows server. So I decided to sync to pool.ntp.org and now I see syslog messages that actually show that the system time gets adjusted by ntpd. I'd rather sync to my LAN time server but this is off-topic on this ML. Using pool and your LAN server would be the best way forward - there are pool server avalable for most countries too, so us.pool.ntp.org, uk.pool.ntp.org, and so on. Your /etc/ntp.conf file can be very simple indeed - my workstation one is nothing more than: server essen.drogon.net server uk.pool.ntp.org You can check your servers ntp daemon with: ntpq -c peers and ntpq -c rl The key thing to look for in the 'rl' command is 'stratum'. If it's 16 then it's not synchronised and anything less than 16 is good. yakko:/home/gordon# ntpq -c rl | fgrep stratum processor=i686, system=Linux/2.6.29.2, leap=00, stratum=4, Don't get too hung-up on how close to zero the stratum is. How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Who cares. Just fix ntpd then your worys are gone. Well, I still have doubts about that. I could look at * source code but I'd rather hear from someone here. Might be easier to read the code ;-) My ntp log shows this: 26 Apr 13:06:30 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:21:24 ntpd[534]: time reset +2.318647 s 26 Apr 13:21:44 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:37:46 ntpd[534]: time reset +2.325417 s 26 Apr 13:38:06 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:54:11 ntpd[534]: time reset +2.327974 s 26 Apr 13:55:19 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:09:16 ntpd[534]: time reset +2.177572 s 26 Apr 14:10:08 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:26:07 ntpd[534]: time reset +2.357017 s That kind of scares me because if I'm not mistaken it means that about every 20 seconds, my ntpd adjusts the system time by about 2 seconds forward. So my clock is going back 2 seconds every 20... That's a significant drift. And it would definitely make a difference in my CDR records IF Asterisk were to compare the start and end system times. Should I worry about this? If ntpd can't keep the kernel time in-sync then it will step abput every 900 seconds - which is what appears to be happening here. (the intervals are typically much longer than 20 seconds - e.g. 13:06:30 to 12:21:24 is ~15 minutes - 900 seconds. I don't think I've ever had a server a bad as that before, so have never looked further... Still, it's 2 seconds in 900 seconds, not 2 in 20 as you thought. Which I think is odd - the Linux clock is software derived based on a hardware interrupt - it only consults the hardware battery-backed clock at boot time (and is supposed to write the current time to it at shutdown time) so I wonder if your server is missing interrupts, or otherwise mis-behaving. Is there anything else odd in the log-files? Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list
Re: [asterisk-users] hardware clock drift and CDR
--- On Mon, 4/26/10, Gordon Henderson gordon+aster...@drogon.net wrote: --- On Sun, 4/25/10, Gordon Henderson gordon+aster...@drogon.net wrote: Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. Hi Gordon, Are you sure about this? Yes. ntpd is a daemon and adjusts the time in a continuous manner. ntp-client or ntpdate or whatever are one-time clients that reset the system clock. I don't see why an ntp-client can't be run while ntpd is working (it shouldn't be necessary but may come in handy when the time difference is big and ntpd refuses to sync). ntp binds to the ntp port (123) and prevents anything else binding to it, or listening on it - which ntpdate needs to do. Example here: Desktop is running ntpd: yakko:/home/gordon# ps ax | fgrep ntp 22064 ? Ss 0:14 /usr/sbin/ntpd -p /var/run/ntpd.pid -u 106:107 -g 30340 pts/29 R+ 0:00 fgrep ntp I try to run ntpdate: yakko:/home/gordon# ntpdate essen.drogon.net 26 Apr 14:20:47 ntpdate[30341]: the NTP socket is in use, exiting Anyway, I've noticed that my ntpd log messages don't say anything when trying to sync to my Windows PDC LAN time server. Curiously, ntp-client DOES sync to this Windows server. So I decided to sync to pool.ntp.org and now I see syslog messages that actually show that the system time gets adjusted by ntpd. I'd rather sync to my LAN time server but this is off-topic on this ML. Using pool and your LAN server would be the best way forward - there are pool server avalable for most countries too, so us.pool.ntp.org, uk.pool.ntp.org, and so on. Your /etc/ntp.conf file can be very simple indeed - my workstation one is nothing more than: server essen.drogon.net server uk.pool.ntp.org You can check your servers ntp daemon with: ntpq -c peers and ntpq -c rl The key thing to look for in the 'rl' command is 'stratum'. If it's 16 then it's not synchronised and anything less than 16 is good. yakko:/home/gordon# ntpq -c rl | fgrep stratum processor=i686, system=Linux/2.6.29.2, leap=00, stratum=4, Don't get too hung-up on how close to zero the stratum is. How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Who cares. Just fix ntpd then your worys are gone. Well, I still have doubts about that. I could look at * source code but I'd rather hear from someone here. Might be easier to read the code ;-) My ntp log shows this: 26 Apr 13:06:30 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:21:24 ntpd[534]: time reset +2.318647 s 26 Apr 13:21:44 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:37:46 ntpd[534]: time reset +2.325417 s 26 Apr 13:38:06 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:54:11 ntpd[534]: time reset +2.327974 s 26 Apr 13:55:19 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:09:16 ntpd[534]: time reset +2.177572 s 26 Apr 14:10:08 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:26:07 ntpd[534]: time reset +2.357017 s That kind of scares me because if I'm not mistaken it means that about every 20 seconds, my ntpd adjusts the system time by about 2 seconds forward. So my clock is going back 2 seconds every 20... That's a significant drift. And it would definitely make a difference in my CDR records IF Asterisk were to compare the start and end system times. Should I worry about this? If ntpd can't keep the kernel time in-sync then it will step abput every 900 seconds - which is what appears to be happening here. (the intervals are typically much longer than 20 seconds - e.g. 13:06:30 to 12:21:24 is ~15 minutes - 900 seconds. I don't think I've ever had a server a bad as that before, so have never looked further... Still, it's 2 seconds in 900 seconds, not 2 in 20 as you thought. Which I think is odd - the Linux clock is software derived based on a hardware interrupt - it only consults the hardware battery-backed clock at boot time (and is supposed to write the current time to it at shutdown time) so I wonder if your server is missing interrupts, or otherwise mis-behaving. Is there anything else odd in the log-files? I ran the following and it supposedly updated my system time
Re: [asterisk-users] hardware clock drift and CDR
On Mon, 26 Apr 2010, Vieri wrote: I ran the following and it supposedly updated my system time while ntpd was running: # ps ax | fgrep ntp 1256 ?Ss 0:00 /usr/sbin/ntpd -p /var/run/ntpd.pid -u ntp:ntp 1623 pts/14 S+ 0:00 fgrep ntp # ntpdate -b -u pool.ntp.org 26 Apr 19:41:18 ntpdate[2791]: step time server 163.117.131.239 offset 0.142263 sec From the ntpdate man page: -u Direct ntpdate to use an unprivileged port for outgoing packets. This is most useful when behind a firewall that blocks incoming traffic to privileged ports, and you want to synchronise with hosts beyond the firewall. Note that the -d option always uses unprivileged ports. So ntpdate does not try and use 123 -- which is in use by ntpd. Does: sudo netstat -a -n -p | grep ntpd show something like: udp0 0 192.168.0.xx:1230.0.0.0:* 1693/ntpd udp0 0 127.0.0.1:123 0.0.0.0:* 1693/ntpd udp0 0 0.0.0.0:123 0.0.0.0:* 1693/ntpd udp6 0 0 fe80::222:68ff:fe36:123 :::* 1693/ntpd udp6 0 0 ::1:123 :::* 1693/ntpd udp6 0 0 :::123 :::* 1693/ntpd unix 2 [ ] DGRAM6635 1693/ntpd -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware clock drift and CDR
On Mon, 26 Apr 2010, Vieri wrote: I ran the following and it supposedly updated my system time while ntpd was running: # ps ax | fgrep ntp 1256 ?Ss 0:00 /usr/sbin/ntpd -p /var/run/ntpd.pid -u ntp:ntp 1623 pts/14 S+ 0:00 fgrep ntp # ntpdate -b -u pool.ntp.org 26 Apr 19:41:18 ntpdate[2791]: step time server 163.117.131.239 offset 0.142263 sec Steves posted the reason - the -u flag causes it to bypass the normal ports, and so does work in this instance. By the way, as a side question, on another server I see this: # ntpq -c peers remote refid st t when poll reach delay offset jitter == inf-srv1.hospit .LOCL. 1 u 56 64 3770.314 21755.8 7.634 Not sure what LOCL means but I'll refer to the NTP docs (inf-srv1 is my LAN Windoze time server). It may mean that it's using it's internal clock as the master source. If so, then it's trust it as far as I could spit a rat... Try this: ntpq host inf-srv1 (or it's IP addresS) peers and find out what peers it's using. It's just possible that your server is actually more accurate that your LAN server... Give your server a few more peers and find out - just list pool.ntp.org in the /etc/ntp.conf file a few times (and restart ntpd) Anyway, back to the faulty new server (which reports a stratum of 3 after ntpd has been running for a while and sync'ing to pool.ntp.org): The stratus is just how far it is away from stratum 1 - which is deemed to be synchronised to true time - usually derived from GPS, local atomic clock or MSF type radio. (I used to run an MSF clock synced NTP server for a while) So a host synchronised to a stratum 1 server will be at stratum 2, and hosts synchronised to a stratum 2 server will be at stratum 3. If you synchronise to a mixture, then your host will be somewhere in the range, depending on how good it reckons the other are... it's supposed to be a good motherboard (Asus) but I'm running a relatively old kernel (2.6.23). Googling around suggests me to try to boot with noapic if I keep seeing my clock drift so much. # more /proc/interrupts CPU0 CPU1 CPU2 CPU3 0:103 0 0 1 IO-APIC-edge timer 1: 2151 0 0 9 IO-APIC-edge i8042 4: 12772543 1321793296030647661766 IO-APIC-edge serial That's a rather high number of serial interrupts... Do you have a serial console, or using the serial link with Linux HA? In-general, I like ASUS motherboards though and use them a lot myself. 8: 1 0 1 0 IO-APIC-edge rtc 9: 0 0 0 1 IO-APIC-fasteoi acpi 12: 0 0 0 4 IO-APIC-edge i8042 14: 2234 73664 0 2470 IO-APIC-edge ide0 16: 28322780 51914617 40744985 39615361 IO-APIC-fasteoi eth0 17: 63242610 42157366 43790794 48255583 IO-APIC-fasteoi eth1 18:1348544 0 0 1 IO-APIC-fasteoi eth2 20:9006839824429560765954923525 IO-APIC-fasteoi ahci 21: 162750903 140985080 176469550 166839225 IO-APIC-fasteoi wcte12xp0 22: 16662710 18210608 12053147 12739782 IO-APIC-fasteoi HFC-multi NMI: 0 0 0 0 LOC: 64546905 64546897 64546897 64546897 ERR: 0 MIS: 0 I have 3 PCI cards: 1 PRI, 1 quad BRI, 1 dual ethernet. Could booting with noapic help? Doubt it, but iy's worth a try. Personally, I'd try more NTP hosts first. (Especially knowing you're syncing to a windoze host ;-) What about my PCI devices? Will they be stable even with noapic? The reason I got this new mobo is that the previous hardware froze the system with a kernel crash. In fact, I rsync'ed to this new hardware (so identical system software) and it has been running flawlessly for more than a week now, while it used to crash/freeze once a day (another Asus board, by the way). My only problem now is with the d...@!mned clock... As far as syslog messages, I don't see anything wrong. No errors whatsoever. Thanks for your time. I'll try to boot with noapic and cross my fingers. Good luck.. What may also help is compiling a custom kernel for your hardware - it's what I do by default, but I appreciate that's not for everyone, however it is the best way to make sure you have the kernel tuned exactly to your hardware needs. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] hardware clock drift and CDR
Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. However, suppose I update system time at every hour and it sets +1 minute (due to a -1 minute drift). Suppose a call is dialed at 03:58 and lasts 4 real minutes. According to the updated system time, the call will have lasted 5 minutes (4+1 drift). How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Thanks Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware clock drift and CDR
On Sun, 25 Apr 2010, Vieri wrote: Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. So I'd start by fixing ntpd. It really is the best way forward. However, suppose I update system time at every hour and it sets +1 minute (due to a -1 minute drift). Suppose a call is dialed at 03:58 and lasts 4 real minutes. According to the updated system time, the call will have lasted 5 minutes (4+1 drift). How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Who cares. Just fix ntpd then your worys are gone. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware requirements question.
I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? -- Thanks, David Little MM Technology, Inc. da...@mandm-tech.com 704.882.9432 x3 704.882.0405 FAX -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements question.
- David Little da...@mandm-tech.com wrote: I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? whistle Zenons?!? Those must be brand new on the market... :-) In all seriousness, yes, I would think that hardware should handle the calls. BUT, how much will you be spending on power? My quick Googling shows thats a pretty beefy box. For what you could save in power, buy a shiny little Intel Atom based or similar low power system. You'll save on your monthly electrical costs plus, you'll have headroom to do other telephony tasks and not have to worry about your system load causing poor voice quality. My $0.02 USD. I accept cash only. :-) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements question.
On Fri, 5 Mar 2010, David Little wrote: I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? Since I'm happy doing that (or something similar) on a 1GHz processor with 256MB of RAM, I'd suggest that your box is somewhat over-specced It'll keep the room warm though. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements question.
On Fri, 5 Mar 2010, Gordon Henderson wrote: On Fri, 5 Mar 2010, David Little wrote: I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? Since I'm happy doing that (or something similar) on a 1GHz processor with 256MB of RAM, I'd suggest that your box is somewhat over-specced It'll keep the room warm though. What's a MHz? This sounds like a really old box he just happens to have laying around... -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements question.
On Fri, 5 Mar 2010, Steve Edwards wrote: On Fri, 5 Mar 2010, Gordon Henderson wrote: On Fri, 5 Mar 2010, David Little wrote: I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? Since I'm happy doing that (or something similar) on a 1GHz processor with 256MB of RAM, I'd suggest that your box is somewhat over-specced It'll keep the room warm though. What's a MHz? This sounds like a really old box he just happens to have laying around... Doh! :) Looks like I missed that bit! Wow - 1GB of RAM in an old 550 MHz Xeon box. I've just given one of these away too - only had 256MB of RAM though! Actually, I reckon it'll work just fine though - I do all my testing on a very old 550MHz VIA system, and have production boxes on 500MHz Geode boxes, so make sure the distro is as lean as possible and off you go... Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements question.
Yes, this machine will be enough for that task. Performance wise. The other good thing is that it is not very likely that someone will steal your PBX. As far as I remember it is a 7 rack unit box which weights approx. one metric ton. ;-) But remember - if anything dies in the box and you have to get spare parts quick you will pay more than you want to. Chris 2010/3/5 David Little da...@mandm-tech.com: I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? -- Thanks, David Little MM Technology, Inc. da...@mandm-tech.com 704.882.9432 x3 704.882.0405 FAX -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware
Thanks Again steve . Actually I feel that is expensive for my initial requirement of maling Asterisk and Zaptel work on my Linux box. I saw this in ebay. only 1 FXO. Asterisk X100P(B2) FXO PCI For IP-PBX From U.S http://cgi.ebay.com/Asterisk-X100P-B2-FXO-PCI-For-IP-PBX-From-U-S_W0QQitemZ160331750263QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item2554845b77#ht_3934wt_1165 This is just 14$.. looks like the distributer sold 600 pieces. Did any one try this with Astersik? how did it work with the ZAPTEL(DHADI)?? were there any issues pl let me know... if it works than I want t order that :-) I am in USA west coast. if u know any one else who can give working pieces f a better deal please let me know. From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sun, February 28, 2010 10:13:13 AM Subject: Re: [asterisk-users] Hardware On Sun, 28 Feb 2010, Aditya Kumar wrote: Can any one please suggest me a Card which is economical.. My requirement is one FXO and one FSO. (FXS) Also, as steve suggested I cannot use ATA because the out put to ATA-SPA is SIP. I want to make use of DHAHII(interface) so looking f card I've never used this vendor and I don't know which corner of the world you're in, but this seems like a pretty good deal: http://www.cetusvoip.com/product_info.php?cPath=1_18_19products_id=2780 Digium TDM411B 1FXS / 1FXO Analog TDM PCI Card for US$220. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users link is : -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware
On 4 Mar 2010, at 17:22, Aditya Kumar wrote: I saw this in ebay. only 1 FXO. Asterisk X100P(B2) FXO PCI For IP-PBX From U.S link is : http://cgi.ebay.com/Asterisk-X100P-B2-FXO-PCI-For-IP-PBX-From-U-S_W0QQitemZ160331750263QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item2554845b77#ht_3934wt_1165 This is just 14$.. looks like the distributer sold 600 pieces. http://www.voip-info.org/wiki/view/X100P+clone Might not be worth the $14.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware
Hi, Im one of the user for this card. It works like charm. in my country i have to set the signalling to fxs_ls and it works. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware
Hi All, Can any one please suggest me a Card which is economical.. My requirement is one FXO and one FSO. Also, as steve suggested I cannot use ATA because the out put to ATA-SPA is SIP. I want to make use of DHAHII(interface) so looking f card - a card, a Digium TDM410p. I've used the TDM400p and it worked fine. For an ATA, a Linksys SPA3102. I have a SPA3000 I still use on occasion. For a USB, Sangoma has a cute little FXO adapter. I've never used it but Sangoma has a good reputation. Also take a look at Xorcom and see if they have anything that fits. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware
On Sun, 28 Feb 2010, Aditya Kumar wrote: Can any one please suggest me a Card which is economical.. My requirement is one FXO and one FSO. (FXS) Also, as steve suggested I cannot use ATA because the out put to ATA-SPA is SIP. I want to make use of DHAHII(interface) so looking f card I've never used this vendor and I don't know which corner of the world you're in, but this seems like a pretty good deal: http://www.cetusvoip.com/product_info.php?cPath=1_18_19products_id=2780 Digium TDM411B 1FXS / 1FXO Analog TDM PCI Card for US$220. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware issue, was Using HASH() and REALTIME_HASH()
On 22/01/2010 19:10, Benoit wrote:
Le 13/01/2010 09:57, Benoit a écrit :
Le 12/01/2010 16:35, Tilghman Lesher a écrit :
On Tuesday 12 January 2010 04:44:36 Benoit wrote:
I just experienced another problem however i have two rnis cards, one
b410p and one te220,
while the later works prefectly i can't really make the first one work,
using DAHDI or mISDN
under asterisk 1.6.
If you're having trouble with any Digium hardware, the best thing to do is
to
call Digium support and get your free installation support provided with our
hardware.
Hi,
I didn't think of this, since it looked like more of an asterisk problem
(asterisk 1.4/misdn = ok asterisk 1.6/misdn = fail, asterisk 1.6/dahdi
= fail).
Audio (both way) is working (voicemail/playback), but it fail when
Dial'ing a device.
Looks like a problem with signalling ...
But anyway i just opened a support case, thanks
Well, in fact it wasn't an hardware issue: when calling thru the B410p
the callerid string is prepended with
an Id, looks like the length of the resulting string is a problem to
initial a SIP call.
Hell, it's even more simple, it was the double quote in the Set() (
Set(CALLERID(name)=- ID - ${CALLERID(name)}) ) that rendered
the sip message invalid ...
--
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Re: [asterisk-users] Hardware issue, was Using HASH() and REALTIME_HASH()
Le 13/01/2010 09:57, Benoit a écrit : Le 12/01/2010 16:35, Tilghman Lesher a écrit : On Tuesday 12 January 2010 04:44:36 Benoit wrote: I just experienced another problem however i have two rnis cards, one b410p and one te220, while the later works prefectly i can't really make the first one work, using DAHDI or mISDN under asterisk 1.6. If you're having trouble with any Digium hardware, the best thing to do is to call Digium support and get your free installation support provided with our hardware. Hi, I didn't think of this, since it looked like more of an asterisk problem (asterisk 1.4/misdn = ok asterisk 1.6/misdn = fail, asterisk 1.6/dahdi = fail). Audio (both way) is working (voicemail/playback), but it fail when Dial'ing a device. Looks like a problem with signalling ... But anyway i just opened a support case, thanks Well, in fact it wasn't an hardware issue: when calling thru the B410p the callerid string is prepended with an Id, looks like the length of the resulting string is a problem to initial a SIP call. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware issue, was Using HASH() and REALTIME_HASH()
Le 12/01/2010 16:35, Tilghman Lesher a écrit : On Tuesday 12 January 2010 04:44:36 Benoit wrote: I just experienced another problem however i have two rnis cards, one b410p and one te220, while the later works prefectly i can't really make the first one work, using DAHDI or mISDN under asterisk 1.6. If you're having trouble with any Digium hardware, the best thing to do is to call Digium support and get your free installation support provided with our hardware. Hi, I didn't think of this, since it looked like more of an asterisk problem (asterisk 1.4/misdn = ok asterisk 1.6/misdn = fail, asterisk 1.6/dahdi = fail). Audio (both way) is working (voicemail/playback), but it fail when Dial'ing a device. Looks like a problem with signalling ... But anyway i just opened a support case, thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware issue, was Using HASH() and REALTIME_HASH()
On Tuesday 12 January 2010 04:44:36 Benoit wrote: I just experienced another problem however i have two rnis cards, one b410p and one te220, while the later works prefectly i can't really make the first one work, using DAHDI or mISDN under asterisk 1.6. If you're having trouble with any Digium hardware, the best thing to do is to call Digium support and get your free installation support provided with our hardware. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware echo cancellation
If I get an echo cancellation module for my Digium TE121 card, will I need to do any adjustments/configuration in Asterisk? You should probably still set the gain using rxgain and txgain. IME, it's much easier setting gains on a PRI than it is on a POTS line, though. I've worked with a couple of PRIs that need no adjustments at all. Is the hardware better than the software version? The hardware version is the same algorithm as the HPEC echo canceler. It's quite a bit better than the MG2 algorithm that comes free with asterisk and maybe slightly better than OSLEC. The convergence time of the hardware algorithm is pretty fast (time it takes for the EC to effectively get rid of echo on a call). FYI: If you're considering running the software-based HPEC for all channels on a T1/E1, you should use a reasonably fast machine, as it uses quite a bit of CPU. That's one big reason to get the hardware module. - Noah TIA! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hardware echo cancellation
I got a few newbie questions. If I get an echo cancellation module for my Digium TE121 card, will I need to do any adjustments/configuration in Asterisk? Is the hardware better than the software version? TIA! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware Requirement for asterisk
i am going to set up asterisk for pbx purpose in my office. i am having 2 PSTN lines and will be configuring 10 extentions in my office. plz tell me which hardware will be needed for this. thx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Requirement for asterisk
On Nov 15, 2009, at 2:27 PM, aster...@opensourcesolution.in wrote: i am going to set up asterisk for pbx purpose in my office. i am having 2 PSTN lines and will be configuring 10 extentions in my office. plz tell me which hardware will be needed for this. thx Have you read this page? http://www.asterisk.org/applications/pbx --- fred http://qxork.com___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Requirement for asterisk
i am going to set up asterisk for pbx purpose in my office. i am having 2 PSTN lines and will be configuring 10 extentions in my office. plz tell me which hardware will be needed for this. Can someone please throw that moron of the list?? _ Looking for a place to manage all your online stuff? Download the new Windows Live http://download.live.com___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Requirement for asterisk
On Nov 15, 2009, at 2:27 PM, aster...@opensourcesolution.in wrote: i am going to set up asterisk for pbx purpose in my office. i am having 2 PSTN lines and will be configuring 10 extentions in my office. plz tell me which hardware will be needed for this. On Sun, 15 Nov 2009, Andreas Anderson wrote: Can someone please throw that moron of the list?? Moron may be a bit strong. Lazy, inconsiderate, and entitled come to mind. As long as people feed the pigeons, the pigeons will return to feed. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Requirement for asterisk
Throwing him off the list would not achieve anything - he still has our email addresses, and will still be able to send you email. Unless of course, you pop his email address on the DENY list of your gateway...*whistles innocently* From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andreas Anderson Sent: Monday, 16 November 2009 06:34 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hardware Requirement for asterisk i am going to set up asterisk for pbx purpose in my office. i am having 2 PSTN lines and will be configuring 10 extentions in my office. plz tell me which hardware will be needed for this. Can someone please throw that moron of the list?? Windows 7: Make your own home movies. Learn more.http://download.live.com/moviemaker IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hardware requirements for asterisk
hello friends friend i had just finished my chapters of asterisk. ill be configuring asterisk in for home for r/d purpose. i am having p4 machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt which i am having is which hardware ill have to buy to configure asterisk. i think analog card ? plz clear my doubt. n be with me from beginning till end, of the journey of asterisk. Regards, Pawan___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware requirements for asterisk
aster...@opensourcesolution.in wrote: hello friends friend i had just finished my chapters of asterisk. ill be configuring asterisk in for home for r/d purpose. i am having p4 machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt which i am having is which hardware ill have to buy to configure asterisk. i think analog card ? plz clear my doubt. n be with me from beginning till end, of the journey of asterisk. Depending on what you intend to accomplish, you may not need any additional hardware; you do not need PSTN connectivity to use Asterisk. If you want it anyway, you can get PSTN origination (calls from the PSTN-VoIP) and termination (VoIP-PSTN) over IP without any need for physical lines. If you have a fixed analog line and are determined to interface it with Asterisk, you would need an FXO card. TDM hardware that interfaces with T1/E1 circuits (ISDN PRI, typically) is also available. -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware requirements for asterisk
On Mon, 2009-11-02 at 09:37 +, aster...@opensourcesolution.in wrote: hello friends friend i had just finished my chapters of asterisk. ill be configuring asterisk in for home for r/d purpose. i am having p4 machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt which i am having is which hardware ill have to buy to configure asterisk. i think analog card ? plz clear my doubt. n be with me from beginning till end, of the journey of asterisk. Regards, Pawan Hi Pawan, It vey much depend on what you expect the box to be handling As you wrote: soho + RD, i presume it will be anoccasional call. Personally, i would recommend to leave the analogue stuff out of your PC. (no hassle with pci-slots, shared-IRQ's, PSU, ) Leave the handling of analogue-parts to an ATA-box. Linksys (and others) are making those at reasonable prices (Cheaper than an analogue card) hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware suggestions
On Thu, 19 Mar 2009 16:38:02 -0300, David fire ddf...@gmail.com wrote: dive in the mailing list archive in February a very nice user sent an email about how to do load balancing using opensip. I don't suppose you know the Subject line, do you David? I can't find it! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware suggestions
Hi, I`m looking for reliable and redundant hardware for Asterisk. I`ve been leaning towards buying one of these (HP 360 G5 with everything as redundant as possible), which I know will be good enough for a few months before needing to upgrade: http://h10010.www1.hp.com/wwpc/us/en/en/WF05a/15351-15351-3328412-241644-241 475-1121486.html Questions: 1) Any reason why I shouldn't? (bad past experience with HP hardware and Asterisk for example) 2) Should I go Quad core or Dual-core? I will certainly go with two processors (to start, simply for redundancy). 3) When installing the OS (CentOS is what I generally use) should I install it 64 bits or 32 bits? (does it even matter for Asterisk?) I will possibly be running a very little used Apache and FTP server. The only notable thing running with Asterisk will be MySQL for CDR and other dialplan data. Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware suggestions
On Thu, 19 Mar 2009, Mike wrote: Hi, I`m looking for reliable and redundant hardware for Asterisk. I`ve been leaning towards buying one of these (HP 360 G5 with everything as redundant as possible), which I know will be good enough for a few months before needing to upgrade: http://h10010.www1.hp.com/wwpc/us/en/en/WF05a/15351-15351-3328412-241644-241 475-1121486.html You can reliably run asterisk on just about any x86 hardware. You don't mention what kind of stresses you are going to put on it, so your sizing questions are impossible to answer. How many extensions? How many simultaneous calls? Will you be transcoding? Routing to/from the PSTN? What cards will you be putting in the box? Some cards don't play nicely together if forced to share interrupts, for example. Questions: 1) Any reason why I shouldn't? (bad past experience with HP hardware and Asterisk for example) 2) Should I go Quad core or Dual-core? I will certainly go with two processors (to start, simply for redundancy). I'm shooting from the hip here, but I don't think dual CPU gives you redundancy. If one chip fries I am pretty sure the machine will crash. 3) When installing the OS (CentOS is what I generally use) should I install it 64 bits or 32 bits? (does it even matter for Asterisk?) Totally depends on what you are planning to do with this box. If you are running for a small office with a handful of extensions and a couple of analog POTS lines, you could potentially use a Celeron with 128MB of RAM and a 1GB hard drive (I have a few of these running myself!). If you are planning to serve several hundred simultaneous calls you have a lot more to think about. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware suggestions
You can reliably run asterisk on just about any x86 hardware. You don't mention what kind of stresses you are going to put on it, so your sizing questions are impossible to answer. How many extensions? How many simultaneous calls? Will you be transcoding? Routing to/from the PSTN? What cards will you be putting in the box? Some cards don't play nicely together if forced to share interrupts, for example. I wasn't worried about sizing (let's imagine that this is more than enough for now and less than I'll need later). More about whether this was the right BRAND more than the right hardware. Does HP make Asterisk friendly hardware? I know Dells was problems a few years back. As for CPU, the question is mostly one about more GHz or more cores? Dual cores are cheaper by GHz. What`s best for Asterisk? I am doing only SIP to SIP calls. Some transcoding (half calls are G711 to G729, the other half are G729 both ways). [snip] I'm shooting from the hip here, but I don't think dual CPU gives you redundancy. If one chip fries I am pretty sure the machine will crash. This was sort of a question disguised as a statement. Can a CPUs function when it's neighbour is fried? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware suggestions
i am very far away to be an expert in my experience i prefer to use a cluster of normal computers instead of an expensive one. if one go down you can trhow it and buy a new one any where very fast. using opensip and *Heartbeat* you you can have an failsafe system. dive in the mailing list archive in February a very nice user sent an email about how to do load balancing using opensip. regards David 2009/3/19 Mike l...@virtutel.ca Hi, I`m looking for reliable and redundant hardware for Asterisk. I`ve been leaning towards buying one of these (HP 360 G5 with everything as redundant as possible), which I know will be good enough for a few months before needing to upgrade: http://h10010.www1.hp.com/wwpc/us/en/en/WF05a/15351-15351-3328412-241644-241475-1121486.html Questions: 1) Any reason why I shouldn't? (bad past experience with HP hardware and Asterisk for example) 2) Should I go Quad core or Dual-core? I will certainly go with two processors (to start, simply for redundancy). 3) When installing the OS (CentOS is what I generally use) should I install it 64 bits or 32 bits? (does it even matter for Asterisk?) I will possibly be running a very little used Apache and FTP server. The only notable thing running with Asterisk will be MySQL for CDR and other dialplan data. Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware suggestions
Mike wrote: You can reliably run asterisk on just about any x86 hardware. You don't mention what kind of stresses you are going to put on it, so your sizing questions are impossible to answer. How many extensions? How many simultaneous calls? Will you be transcoding? Routing to/from the PSTN? What cards will you be putting in the box? Some cards don't play nicely together if forced to share interrupts, for example. Sizing is important. Take your company's projected growth rate, double it, and work it out for 3-5 years. I recommend 5 years for the sizing. As much as its fun to tinker, once it goes into production you want to have it as stable as possible. Look at all the apps you want to use and figure out how much they are going to cost you in terms of resources. In the company I work for, we put in Asterisk to replace our Nortel system which reached the limits. So we expected standard usage rates and growth etc. However, once we introduced meetme application our Asterisk usage spiked. We figured on average 2-3 meetme meetings a week (based on the usage of a third party conference bridge we had before), and now its at 2-3 a day. We had it setup so that every person has their own conference bridge. Other features are also taking up more resources. I'm currently modigying meetime and writing an AGI so that once the meetme conference ends, it will take the recording and conver it to an mp3 and then emails it to the leader. I wasn't worried about sizing (let's imagine that this is more than enough for now and less than I'll need later). More about whether this was the right BRAND more than the right hardware. Does HP make Asterisk friendly hardware? I know Dells was problems a few years back. As for CPU, the question is mostly one about more GHz or more cores? Dual cores are cheaper by GHz. What`s best for Asterisk? I am doing only SIP to SIP calls. Some transcoding (half calls are G711 to G729, the other half are G729 both ways). [snip] I'm shooting from the hip here, but I don't think dual CPU gives you redundancy. If one chip fries I am pretty sure the machine will crash. This was sort of a question disguised as a statement. Can a CPUs function when it's neighbour is fried? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware suggestions
On Thu, 19 Mar 2009, Mike wrote: You can reliably run asterisk on just about any x86 hardware. You don't mention what kind of stresses you are going to put on it, so your sizing questions are impossible to answer. How many extensions? How many simultaneous calls? Will you be transcoding? Routing to/from the PSTN? What cards will you be putting in the box? Some cards don't play nicely together if forced to share interrupts, for example. I wasn't worried about sizing (let's imagine that this is more than enough for now and less than I'll need later). More about whether this was the right BRAND more than the right hardware. Does HP make Asterisk friendly hardware? I know Dells was problems a few years back. AFAIK (there's that acronym again :):) ), the Dell issues were related to interrupt sharing and multiple PSTN interface cards. You mention below SIP/SIP only, so I wouldn't worry. The HP should be fine. As for CPU, the question is mostly one about more GHz or more cores? Dual cores are cheaper by GHz. What`s best for Asterisk? That's actually a decent question. Anyone have any benchmarks? It is the transcoding that will eat your CPU. I think with minimal transcoding you would have a hard time overloading a 2.4GHz machine before other factors came into play. I am doing only SIP to SIP calls. Some transcoding (half calls are G711 to G729, the other half are G729 both ways). [snip] I'm shooting from the hip here, but I don't think dual CPU gives you redundancy. If one chip fries I am pretty sure the machine will crash. This was sort of a question disguised as a statement. Can a CPUs function when it's neighbour is fried? Certainly the machine will crash, and I doubt it would boot on one CPU if the other is still installed and shorting out its pins :) I think David is completely correct. If you want a redundant setup, run multiple smaller cheaper machines with a load balancing front end. Stay away from single points of failure. j Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware suggestions
On Thu, 19 Mar 2009, Mike wrote: Hi, I`m looking for reliable and redundant hardware for Asterisk. I`ve been leaning towards buying one of these (HP 360 G5 with everything as redundant as possible), which I know will be good enough for a few months before needing to upgrade: http://h10010.www1.hp.com/wwpc/us/en/en/WF05a/15351-15351-3328412-241644-241 475-1121486.html Hm. Expensive, but ... Questions: 1) Any reason why I shouldn't? (bad past experience with HP hardware and Asterisk for example) 2) Should I go Quad core or Dual-core? I will certainly go with two processors (to start, simply for redundancy). xxx-CORE. Both cores in the same physical chip. The chances of one failing and the other not... slim, I reckon, and while Linux does have support for hot-plug CPUs, I doubt it's intended to work at the chip level like that. 3) When installing the OS (CentOS is what I generally use) should I install it 64 bits or 32 bits? (does it even matter for Asterisk?) Use the one you are most familair with. I will possibly be running a very little used Apache and FTP server. The only notable thing running with Asterisk will be MySQL for CDR and other dialplan data. The hardware is overkill for that. For that price you can get 2 Atom motherboards and run them in Linux HA mode if you want redundancy. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware suggestions
I'm shooting from the hip here, but I don't think dual CPU gives you redundancy. If one chip fries I am pretty sure the machine will crash. This was sort of a question disguised as a statement. Can a CPUs function when it's neighbour is fried? Dualcore means two cores on one CPU. Quadcore is 4 cores on one CPU. There are not multiple CPUs, unless you start going into specifically multi-processor (SMP) systems. These were popular on higher end PC grade hardware before Dualcores came into existence, but are now redundant. So if you bought a dual-dualcore machine, you may possibly have redundancy (depending on how the board is designed to handle one CPU popping). CPUs don't tend to fry themselves unless something else like the CPU fan has first gone and allowed the CPU to overheat. Worrying about processor redundancy is overkill, IMO. Have a backup machine or a spare motherboard/CPU+fan, or have a good support contract with your hardware vendor. -- Spiro Harvey Knossos Networks Ltd 021-295-1923www.knossos.net.nz signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware suggestions
I did mean multiple chips, not multiple cores. Thanks Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Spiro Harvey Sent: Thursday, March 19, 2009 16:36 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hardware suggestions I'm shooting from the hip here, but I don't think dual CPU gives you redundancy. If one chip fries I am pretty sure the machine will crash. This was sort of a question disguised as a statement. Can a CPUs function when it's neighbour is fried? Dualcore means two cores on one CPU. Quadcore is 4 cores on one CPU. There are not multiple CPUs, unless you start going into specifically multi-processor (SMP) systems. These were popular on higher end PC grade hardware before Dualcores came into existence, but are now redundant. So if you bought a dual-dualcore machine, you may possibly have redundancy (depending on how the board is designed to handle one CPU popping). CPUs don't tend to fry themselves unless something else like the CPU fan has first gone and allowed the CPU to overheat. Worrying about processor redundancy is overkill, IMO. Have a backup machine or a spare motherboard/CPU+fan, or have a good support contract with your hardware vendor. -- Spiro Harvey Knossos Networks Ltd 021-295-1923www.knossos.net.nz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware that can accomondate 2 TDM24
Are you locked into the 3U form factor? We're running Asterisk on a Dell PowerEdge 1950 (1U, 2 full height PCI-E slots [one home to an AEX-804E], 3 drive bays, redundant power). I both the 2950 and 2970 (both are 2U, variable number of drive bays based on the config you choose, the 2950 shares firmware with the 1950) can be ordered with PCI-E risers because we have a handful in our datacenter, but I have no idea how many slots -- I want to say 3. I think the TDM24 is too long to fit in a 1950, but I'm pretty sure (you'd have to check) that the 2950/70 has at least two full-length slots. HTH, 3U leaves me more room for further expansion if my customer wanted to. Dell 2950 looks like what I wanted but the starting price is quite high. That's why I prefer supermicro since it has a way lower starting price. Cheers, Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St. 604-585-2...@104 (main) | Surrey, BC 604-585-3056 (fax)| Canada, V3W 1R1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hardware that can accomondate 2 TDM24
Hi guys, I'm building a server that need to host 2 digium TDM24 cards. I know any 3U server with 2 PCI-E slots would do. Since I do prefer supermicro server, but getting one configured is pretty darn hard. Any suggestions here? Cheers, Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St. 604-585-2...@104 (main) | Surrey, BC 604-585-3056 (fax)| Canada, V3W 1R1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware that can accomondate 2 TDM24
Are you locked into the 3U form factor? We're running Asterisk on a Dell PowerEdge 1950 (1U, 2 full height PCI-E slots [one home to an AEX-804E], 3 drive bays, redundant power). I both the 2950 and 2970 (both are 2U, variable number of drive bays based on the config you choose, the 2950 shares firmware with the 1950) can be ordered with PCI-E risers because we have a handful in our datacenter, but I have no idea how many slots -- I want to say 3. I think the TDM24 is too long to fit in a 1950, but I'm pretty sure (you'd have to check) that the 2950/70 has at least two full-length slots. HTH, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 F: 440.729.0884 I:http://www.controlworks.com Crestron Authorized Independent Programmer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kelvin Chan Sent: Wednesday, February 04, 2009 7:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] hardware that can accomondate 2 TDM24 Hi guys, I'm building a server that need to host 2 digium TDM24 cards. I know any 3U server with 2 PCI-E slots would do. Since I do prefer supermicro server, but getting one configured is pretty darn hard. Any suggestions here? Cheers, Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St. 604-585-2...@104 (main) | Surrey, BC 604-585-3056 (fax)| Canada, V3W 1R1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware that can accomondate 2 TDM24
Saw your post...let me know what suggestions arise (I do not watch the list that closely -- your was flagged because my monitoring software spotted your email address). g. -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware supporting groundstart signalling
Hello! Recently I posted a question about an installation I have that was experiencing glare problems. The solutions presented were to use inverse inbound and outbound line groups and to use groundstart signalling. As it turns out, the Sangoma A400D card that is in use does NOT support groundstart. I've confirmed this with a Sangoma engineer and their support staff. I've also read that Digium products do not support groundstart signalling. Since glare is a common problem with analog PBX systems, it would make sense that groundstart is a common signalling type. Why do the major manufacturers not support this? If you're using groundstart, what hardware are you using? Thank you! Tim Nelson Systems/Network Support Rockbochs Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware Platform
We are in the process of building out www.dialaway4free.com, a free world wide calling service. I am writing RFQ's for hardware, since we are going to use asterisk as our call processor. I was wondering what is the best server platform to use that will support digium cards and handle sip termination for both clients and service providers. Also should I go with the open source of asterisk as compared to Asterisk for Business. Please let me know. I want this system to be stable as we will do a lot of proprietary programming for it to switch to the advertising component so I want to know what people think to handle the a call volume of at least 100,000 calls an hour. Some of my choices: Dell Gateway Gigabyte Ausus Please advise what type of processors and how much memory and hard drives, there will be no voicemail initially maybe it will be offered at a later time. Thanks Visit www.dialaway4free.com and register for a free account and be ready for our September 1st 2008 launch. Worldwide calling to and from any country! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Platform
The real queestion is: What kind of provider is able to support such of a call volume? How do you plan to provide the service ? I mean besides generated AD's will you redirect your call to any sip providers ? Or we are dealing with mass install in most of the regions of the world (where possible) and redirect from dundi or enum, and then terminating direct into line (zap/isdn) -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Goran Donev Envoyé : jeudi 13 mars 2008 22:10 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Hardware Platform We are in the process of building out www.dialaway4free.com, a free world wide calling service. I am writing RFQ's for hardware, since we are going to use asterisk as our call processor. I was wondering what is the best server platform to use that will support digium cards and handle sip termination for both clients and service providers. Also should I go with the open source of asterisk as compared to Asterisk for Business. Please let me know. I want this system to be stable as we will do a lot of proprietary programming for it to switch to the advertising component so I want to know what people think to handle the a call volume of at least 100,000 calls an hour. Some of my choices: Dell Gateway Gigabyte Ausus Please advise what type of processors and how much memory and hard drives, there will be no voicemail initially maybe it will be offered at a later time. Thanks Visit www.dialaway4free.com and register for a free account and be ready for our September 1st 2008 launch. Worldwide calling to and from any country! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware needed
Dear List, I have to plan an instalation of an asterisk box for over 400 extensions (Sip and Iax2) and 4 PRI channels. I do not know which hardware (server) should I buy to support this amount of extensions. Someone made a similar instalation? which hardware (server) did you use? Which was the processor type and the amount of memory used by the server? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware needed
On Feb 13, 2008 10:15 AM, voip crazy [EMAIL PROTECTED] wrote: Someone made a similar instalation? which hardware (server) did you use? Which was the processor type and the amount of memory used by the server? You will probably get some useful info on the list but also check out voip-info.org: http://www.voip-info.org/wiki/view/Asterisk+dimensioning http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations cheers, stoffell ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements
At 01:58 10/14/2007, YT Lim wrote: I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? /Y.T. Case: 1 CodeGen 4U Server Case $80 http://tinyurl.com/bnobzhttp://tinyurl.com/bnobz http://tinyurl.com/95s2b http://www.newegg.com/Product/Product.aspx?Item=N82E16811182566 Or: 1 Eagle Tech ET-RMAL2025-SL Beige 2U Server Case 2 External 5.25 Drive Bays http://www.newegg.com/Product/Product.aspx?Item=N82E1687111 Power Supply: 1 Dual 450 W. power supply -- IStar https://www.ewiz.com/detail.php?name=PS-TC50R8A http://www.directron.com/tc400r8.html Or: 1 535W power supply -- Enermax https://www.mwave.com/mwave/viewspec.hmx?scriteria=BA23110 Motherboard, CPU 1GB of memory: http://www.mwave.com/mwave/skusearch.hmx?scriteria=MB-BA23835AMD ATHLON 64 X2 5000+ (ADO5000DDBOX) ENERGY EFFICIENT RETAIL BOXED W/512KB X 2 CACHE 65NM 65W (BRISBANE) BUNDLE W/ http://www.mwave.com/mwave/skusearch.hmx?scriteria=BA22827ASUS M2NPV-VMhttp://www.mwave.com/mwave/skusearch.hmx?scriteria=BA20346 CRUCIAL 1GB DDR2 533 http://www.mwave.com/mwave/skusearch.hmx?scriteria=BA20346(512MB x 2) http://www.mwave.com/mwave/skusearch.hmx?scriteria=TESTASSEMBLE/TEST BUNDLE $235.99 $235.99 SKU: http://www.mwave.com/mwave/skusearch.hmx?scriteria=MB-BA23835MB-BA23835 -http://www.mwave.com/mwave/skusearch.hmx?scriteria=BA22827 BA22827 -http://www.mwave.com/mwave/skusearch.hmx?scriteria=BA20346 BA20346 -http://www.mwave.com/mwave/skusearch.hmx?scriteria=BA20346 BA20346 - -http://www.mwave.com/mwave/skusearch.hmx?scriteria=TEST TEST 2 Hard Drives in RAID 1 config: SEAGATE 250GB ST3250410AS SATA2 16MB 7200RPM http://www.mwave.com/mwave/viewspec.hmx?scriteria=AA71142RSKU=AA71142 1 DVD ROM Drive: http://www.mwave.com/mwave/viewspec.hmx?scriteria=AA36690 1 Floppy Drive: http://www.mwave.com/mwave/viewspec.hmx?scriteria=AA00696 Interface card: 2 port, 64 bit, 3.3 volt http://www.google.com/search?q=Sangoma+2+port%2C+64+bit http://www.google.com/search?q=Digium+2+port%2C+64+bit ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements
On 10/15/07, Doug [EMAIL PROTECTED] wrote: Case: 1 CodeGen 4U Server Case $80 http://tinyurl.com/bnobz http://tinyurl.com/95s2b http://www.newegg.com/Product/Product.aspx?Item=N82E16811182566 Or: 1 Eagle Tech ET-RMAL2025-SL Beige 2U Server Case 2 External 5.25 Drive Bays http://www.newegg.com/Product/Product.aspx?Item=N82E1687111 Power Supply: 1 Dual 450 W. power supply -- IStar https://www.ewiz.com/detail.php?name=PS-TC50R8A http://www.directron.com/tc400r8.html Or: 1 535W power supply -- Enermax https://www.mwave.com/mwave/viewspec.hmx?scriteria=BA23110 Motherboard, CPU 1GB of memory: AMD ATHLON 64 X2 5000+ (ADO5000DDBOX) ENERGY EFFICIENT RETAIL BOXED W/512KB X 2 CACHE 65NM 65W (BRISBANE) BUNDLE W/ ASUS M2NPV-VM Don't get me wrong, the M2NPV are great boards we use them all the time for home appliances type devices they run 24/7 and process alot of media. And also for frontend because they have the HD video outputs. However I'd prefer to use a server mainboard for dedicated Asterisk systems. I've had great luck with the Tyan bareones systems. Or SuperMicro is great too. Its too bad you can't find any cheap 4U barebones... 1U $500.. 4U $1200 it makes no sense. Spend a few more dollars on a server or workstation mainboard I've found in 2 years reliability is greater. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements
At 16:13 10/15/2007, Andreas van dem Helge wrote: On 10/15/07, Doug [EMAIL PROTECTED] wrote: Case: 1 CodeGen 4U Server Case $80 http://tinyurl.com/bnobz http://tinyurl.com/95s2b http://www.newegg.com/Product/Product.aspx?Item=N82E16811182566 Or: 1 Eagle Tech ET-RMAL2025-SL Beige 2U Server Case 2 External 5.25 Drive Bays http://www.newegg.com/Product/Product.aspx?Item=N82E1687111 Power Supply: 1 Dual 450 W. power supply -- IStar https://www.ewiz.com/detail.php?name=PS-TC50R8A http://www.directron.com/tc400r8.html Or: 1 535W power supply -- Enermax https://www.mwave.com/mwave/viewspec.hmx?scriteria=BA23110 Motherboard, CPU 1GB of memory: AMD ATHLON 64 X2 5000+ (ADO5000DDBOX) ENERGY EFFICIENT RETAIL BOXED W/512KB X 2 CACHE 65NM 65W (BRISBANE) BUNDLE W/ ASUS M2NPV-VM Don't get me wrong, the M2NPV are great boards we use them all the time for home appliances type devices they run 24/7 and process alot of media. And also for frontend because they have the HD video outputs. However I'd prefer to use a server mainboard for dedicated Asterisk systems. I've had great luck with the Tyan bareones systems. Where do you buy them? Or SuperMicro is great too. That's very debatable. Purchase from who? Its too bad you can't find any cheap 4U barebones... 1U $500.. 4U $1200 it makes no sense. Spend a few more dollars on a server or workstation mainboard I've found in 2 years reliability is greater. Asus 3 year warranty: http://support.asus.com/service/service_right.aspx?SLanguage=en-usno=231 Supermicro 3 year warranty? http://www.google.com/search?q=supermicro+motherboard+%22year+warranty%22 (Also known for their horrible support) Tyan 3 year warranty: http://www.tyan.com/archive/products/html/warranty.html A server motherboard might be better--it will certainly cost more. But the manufacturer won't guarantee it for more than 3 years. 3 years is probably the useful life of an Asterisk server anyway. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware requirements
I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? /Y.T. Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. http://au.docs.yahoo.com/mail/unlimitedstorage.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements
On Sun, 14 Oct 2007, YT Lim wrote: I don't seem to be able to find the necessary hardware specs for an Asterisk server. Look more. There are 100's of pages on it. Start at http://www.voip-info.org/wiki/ What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? You would get away with a 1GHz intel (or intel like) processor for this system, so the answer is: Any modern server will do the job you need it to. Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? Can't help you there I'm afraid. Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements
We use dell 860 rackmount server - not too expensive, readily available and can handle well over 50 phones. PaulH On Sun, 2007-10-14 at 16:58 +1000, YT Lim wrote: I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? /Y.T. Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. http://au.docs.yahoo.com/mail/unlimitedstorage.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements
About memory, I think 512MB will be more than enougth. And hard drive requirements depends on the configuration of your voice boxes, but any modern server will be OK, I don't think that you need more than 20GB... On 10/14/07, Paul Hales [EMAIL PROTECTED] wrote: We use dell 860 rackmount server - not too expensive, readily available and can handle well over 50 phones. PaulH On Sun, 2007-10-14 at 16:58 +1000, YT Lim wrote: I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? /Y.T. Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. http://au.docs.yahoo.com/mail/unlimitedstorage.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements
20GB should be fine - unless you want to do a lot of recording. PaulH On Sun, 2007-10-14 at 21:07 -0600, Edgar Guadamuz wrote: About memory, I think 512MB will be more than enougth. And hard drive requirements depends on the configuration of your voice boxes, but any modern server will be OK, I don't think that you need more than 20GB... On 10/14/07, Paul Hales [EMAIL PROTECTED] wrote: We use dell 860 rackmount server - not too expensive, readily available and can handle well over 50 phones. PaulH On Sun, 2007-10-14 at 16:58 +1000, YT Lim wrote: I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is the necessary configuration on Asterisk? /Y.T. Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. http://au.docs.yahoo.com/mail/unlimitedstorage.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Platform Recommendations for Digium Card Compatability
On 8/10/07, Jason K. Carter [EMAIL PROTECTED] wrote: Could everyone that has a working production Asterisk server that uses a Digium telephony card as a BRI/PRI gateway let me know what motherboard/processor your server uses? Currently running a TE412P in a IBM x3650 Model 7979. I had some problems when I also had a TDM400B in the same system. I have also run this card successfully on a Intel SE7230NH-1 board (having the TDM400B installed as well was not a problem on this board) I had a reproduceable kernel panic under moderate load running this board on a HP DL380G5 with Zaptel 1.4. Zaptel 1.2 was just fine. All of my testing was done on CentOS 4.4 and 4.5. My zttest scores (on the IBM) are generally above 98%, but sometimes I see the tests start at 97.73% for about 20 seconds before it climbs. I often see spikes up to 100% rapidly followed by a drop back to 98.x%. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Platform Recommendations for Digium Card Compatability
James FitzGibbon wrote: On 8/10/07, Jason K. Carter [EMAIL PROTECTED] wrote: Could everyone that has a working production Asterisk server that uses a Digium telephony card as a BRI/PRI gateway let me know what motherboard/processor your server uses? Currently running a TE412P in a IBM x3650 Model 7979. I had some problems when I also had a TDM400B in the same system. I have also run this card successfully on a Intel SE7230NH-1 board (having the TDM400B installed as well was not a problem on this board) I had a reproduceable kernel panic under moderate load running this board on a HP DL380G5 with Zaptel 1.4. Zaptel 1.2 was just fine. Do you have any more information on this? (i.e. stacktrace and error associated with it)? Make sure you're testing with a current 1.2 or 1.4 version of the drivers. There have been a few bug fixes in the last few releases. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware Platform Recommendations for Digium Card Compatability
Hi there, Could everyone that has a working production Asterisk server that uses a Digium telephony card as a BRI/PRI gateway let me know what motherboard/processor your server uses? We use a Digium TE210P as our telephony card. We have tried a couple motherboards, and neither is giving us satisfactory zttest scores. The machine we have in production right now is a Biostar U8668-D (VIA 8237R chipset) with a P4 2.4 GHz, but that system gives regular scores barely above the 98% mark, with occasional dips below. We've tried one other board based on an Intel 865 chipset with a 2.8GHz Celeron, and that board gives regular 100% marks, with dips down to 95% every 5 seconds or so, which makes it pretty unusable. Any help in identifying a better motherboard for Digium cards would be appreciated. Thank you, Jason Carter Software Engineer DLS Internet Services [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware advice for 100 extensions, routing via ISDN
In the O'Reilly Asterisk book it suggests that it is important to allow BIOS specification of the PCI slot IRQs -- the Tyan won't let us do that I don't think. Is this an issue with the Sangoma card? Also comments about how suitable this machine is would also be gratefully received. Rory On 04/08/07, Rory Campbell-Lange ([EMAIL PROTECTED]) wrote: I would be grateful for some comments on our proposed machine specs for a new Asterisk installation at a client with an initial 70 extensions. The system should be able to handle 100 extensions. The system will have the following main features: - PSTN connection via ISDN 30, dealing with all incoming calls. Outgoing will be through ISDN initially - 70-100 Snom 300 handsets - 1-2 Snom 370 reception phones - voicemail voicemail to email - occasional conferencing requirements This is a normal office environment (architects) and we do not anticipate exceptionally heavy call volumes; on the other hand some conversations will last a very long time. I've had a look at http://voip-info.org/wiki/view/Asterisk+dimensioning We are presently intending to put in 2 number CHASSIS/CASE: 2U 2HotSwap Bay 510W PSU MOTHERBOARD: Tyan s5197G2NR CPU(s): Core2Due E6600 (2*2.4GHz) MEMORY: 4GB 667 ECC (2*2048) HDD: 2*150GB Raptor HDD CD/DVD: DVD/RW OTHER: Sangoma A101PCI Card We will be running on 64 bit Debian. The second machine is to be used in place of the first in case of failure. -- Rory Campbell-Lange Campbell-Lange Workshop Ltd. [EMAIL PROTECTED] www.campbell-lange.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware advice for 100 extensions, routing via ISDN
On Sun, 5 Aug 2007, Rory Campbell-Lange wrote: In the O'Reilly Asterisk book it suggests that it is important to allow BIOS specification of the PCI slot IRQs -- the Tyan won't let us do that I don't think. Is this an issue with the Sangoma card? Probably not. Once the system is built, have a a look at /proc/interrupts to see what's on what. Sometimes moving a card into a different PCI slot helps. Turn off *ALL* unused hardware in the BIOS. Eg. Serial ports, printer, on-board sound, 2nd Ethernet port (if possible), and so on. Also comments about how suitable this machine is would also be gratefully received. If you root around the inteweb, you'll find success stories of people running more than 100 SIP extensions on much lesser hardware - eg. on a 1Ghz Via type motherboard. One site by our very own Tzafrir Cohen has some excellent data on it - see http://www.xorcom.com/support/xorcom_ts_1/test_results_for_xorcom_ts_1 The killer is transcoding - and my guess is that you're not doing any (or at the very minimal just support for a very small number of home/remite users) - you are basically a classic PBX type scenario - so make sure all the phones are using ulaw (if you're in the US, alaw elsewhere) and off you go. I don't know *exactly* what happens to the digital data stream when it's bridged between a PRI channel and a SIP channel, but I'd hope if you made sure all your SIP phones used the same codec as the PRI then life will be simple (and transcoding from ulaw to alaw isn't at all cpu intensive anyway, so even if it's wrong, it's not a big issue I reckon) I'd pick a motherboard based more on what you're familiar with than pure spec. So as I have a lot of experience with Asus motherboards, I choose them, in particular, I have a few servers based on a barebones mini tower server from Asus, the TS100 and with a dual-core processor running LAMP type applications extremely well, so I'm looking to to roll out a few for some Asterisk projects soon. (The key with them is remote bios access via a serial line, which for remotely hosted boxes is a good feature to have IMO!) So basically any modern server type box will be fine, so go with what you're familiar with. I'd also suggest sticking to 32-bit Debian too - no real reason other than (again) familiarity - I have a lot of servers running 32-bit Debian and they just work. I'm not convinced there's much advantage right now in moving to 64-bit stuff (unless you're heavilly into scientific stuff and lots lots of memory, and I worked in a place recently where they needed just that - 64-bit Suse, 64GB of RAM, 8 CPUs... but it was a bit specialist!!!) But if you've got a lot of experience with 64-bit debian, then go for it... I'd strongly suggest getting the Asterisk sources and compiling them than using the supplied packages which are a bit out of date by now, and I'd also suggest compiling up a custom kernel too and removing all the hotplug nonsense - no modules in the kernel other than the zap, etc. ones. It can make booting quicker and you only have loaded exactly what gets used, but again, this is personal preferance - I've been doing it since the year dot, so I keep on doing it. Gordon Rory On 04/08/07, Rory Campbell-Lange ([EMAIL PROTECTED]) wrote: I would be grateful for some comments on our proposed machine specs for a new Asterisk installation at a client with an initial 70 extensions. The system should be able to handle 100 extensions. The system will have the following main features: - PSTN connection via ISDN 30, dealing with all incoming calls. Outgoing will be through ISDN initially - 70-100 Snom 300 handsets - 1-2 Snom 370 reception phones - voicemail voicemail to email - occasional conferencing requirements This is a normal office environment (architects) and we do not anticipate exceptionally heavy call volumes; on the other hand some conversations will last a very long time. I've had a look at http://voip-info.org/wiki/view/Asterisk+dimensioning We are presently intending to put in 2 number CHASSIS/CASE: 2U 2HotSwap Bay 510W PSU MOTHERBOARD: Tyan s5197G2NR CPU(s): Core2Due E6600 (2*2.4GHz) MEMORY: 4GB 667 ECC (2*2048) HDD: 2*150GB Raptor HDD CD/DVD: DVD/RW OTHER: Sangoma A101PCI Card We will be running on 64 bit Debian. The second machine is to be used in place of the first in case of failure. -- Rory Campbell-Lange Campbell-Lange Workshop Ltd. [EMAIL PROTECTED] www.campbell-lange.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or
Re: [asterisk-users] Hardware advice for 100 extensions, routing via ISDN
Hi Gordon. Very many thanks for your comments. On 05/08/07, Gordon Henderson ([EMAIL PROTECTED]) wrote: On Sun, 5 Aug 2007, Rory Campbell-Lange wrote: In the O'Reilly Asterisk book it suggests that it is important to allow BIOS specification of the PCI slot IRQs -- the Tyan won't let us do that I don't think. Is this an issue with the Sangoma card? Probably not. Once the system is built, have a a look at /proc/interrupts to see what's on what. Sometimes moving a card into a different PCI slot helps. Turn off *ALL* unused hardware in the BIOS. Eg. Serial ports, printer, on-board sound, 2nd Ethernet port (if possible), and so on. OK, thanks for the advice. The killer is transcoding - and my guess is that you're not doing any (or at the very minimal just support for a very small number of home/remite users) - you are basically a classic PBX type scenario - so make sure all the phones are using ulaw (if you're in the US, alaw elsewhere) and off you go. As we are in the UK, looks like we should use alaw. (Thanks also for the example.) ... So basically any modern server type box will be fine, so go with what you're familiar with. ... But if you've got a lot of experience with 64-bit debian, then go for it... ... This is great advice if it has worked for you. We're going to go for it (and yes, serial console support on a server is a must-have). We having been using 64 bit Debian for a couple of years now without any problems on about 24 servers. I'd strongly suggest getting the Asterisk sources and compiling them than using the supplied packages which are a bit out of date by now, and I'd also suggest compiling up a custom kernel too and removing all the hotplug nonsense - no modules in the kernel other than the zap, etc. ones. It can make booting quicker and you only have loaded exactly what gets used, but again, this is personal preferance - I've been doing it since the year dot, so I keep on doing it. I'll give that a go. This is the approach we use for our gateway machines. Thanks again, Rory -- Rory Campbell-Lange Director Campbell-Lange Workshop Ltd. [EMAIL PROTECTED] www.campbell-lange.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware advice for 100 extensions, routing via ISDN
I would be grateful for some comments on our proposed machine specs for a new Asterisk installation at a client with an initial 70 extensions. The system should be able to handle 100 extensions. The system will have the following main features: - PSTN connection via ISDN 30, dealing with all incoming calls. Outgoing will be through ISDN initially - 70-100 Snom 300 handsets - 1-2 Snom 370 reception phones - voicemail voicemail to email - occasional conferencing requirements This is a normal office environment (architects) and we do not anticipate exceptionally heavy call volumes; on the other hand some conversations will last a very long time. I've had a look at http://voip-info.org/wiki/view/Asterisk+dimensioning We are presently intending to put in 2 number CHASSIS/CASE: 2U 2HotSwap Bay 510W PSU MOTHERBOARD: Tyan s5197G2NR CPU(s): Core2Due E6600 (2*2.4GHz) MEMORY: 4GB 667 ECC (2*2048) HDD: 2*150GB Raptor HDD CD/DVD: DVD/RW OTHER: Sangoma A101PCI Card We will be running on 64 bit Debian. The second machine is to be used in place of the first in case of failure. Advice gratefully received. Rory -- Rory Campbell-Lange Campbell-Lange Workshop Ltd. [EMAIL PROTECTED] www.campbell-lange.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote: This SOHO PBX box won't interop with Asterisk because it doesn't speak any of the protocols that Asterisk does. This box I tend agree with your evaluation. Still, I was thinking that since all these el-cheapo SOHO PBX boxes support manual attendant call transfer, what's to prevent Asterisk from mimicking an attendant by sending proper DTMF signals and make this box transfer the call to the single analog phone in the business? That is, Asterisk will connect (via RJ-11) to the unit as the attendant's phone, and my real phone (only one in the system) will connect via a second RJ-11 (there could be 4 of them). Or is Asterisk not capable of sending DTMF signals over an RJ-11 connection? You can send arbitrary DTMF over any of Asterisk's channels from the dialplan. I just figured that this level of integration was a bit deeper than you were looking for as a first project. It would be an interesting experiment, to be sure. The biggest issue I'd think would be feedback - you can send the DTMF along the wire, but how do you know that the SOHO box interpreted it correctly? If the only feedback is designed for a human (i.e. auditory), then interpreting those cues with Asterisk would be non-trivial. Do I undestand correctly that with this solution, I will still be able to connect to my analog Verizon phone line with the SIP phone? That is, the outside world will see my phone as an ordinary phone, when in fact I am using a SIP phone? If so, that means that Asterisk does all the magic behind the scene, right? Yes, your Verizon POTS line would go into a FXO port in your server (which in Asterisk would be referenced as the channel Zap/1 - zaptel being Asterisk's TDM driver) and your SIP phone would connect via your standard office network and be referenced as SIP/whateverusernameyouwant. A very simplistic example of bridging a call would be: [from-verizon] exten = s,1,Dial(SIP/whateverusername) Assuming that you'd configured zaptel to route calls that come in on the FXO port to the Asterisk context named from-verizon, then any such calls would immediately cause Asterisk to ring your SIP phone, and if answered to bridge the two calls together. A more complex example that makes them press one to call you and otherwise lets them leave a message: [from-verizon] exten = s,1,Background(Press1ToTalkOr2ToLeaveAMessage) exten = s,n,WaitExten(10) ; timeout exten = t,1,Goto(vm,1) ; invalid exten = i,1,Goto(vm,1) ; press 1 exten = 1,1,Dial(SIP/101,20) exten = 1,n,Goto(vm,1) ; press 2 exten = 2,1,Goto(vm,1) ; all voicemail activity ends up here exten = vm,1,VoiceMail(u101) exten = vm,n,Hangup [from-officephone] exten = *98,1,VoiceMailMain extne = *98,n,Hangup Assuming you've now set up your SIP phone as extension 101, this would play a sound file saying press 1 to talk to 2 to leave a message. If they press 1, your SIP phone rings. If they press 2, they go to voicemail. If they wait 10 seconds without pressing anything, or press something other than 1 or 2, they also go to voicemail. If they press 1 to dial your phone and you don't pick up after 20 seconds, they go to voicemail. On your deskphone (could just as easily be a SIP softphone if you prefer), you can dial *98 to log in and pick up your new voicemail messages. Hope that demystifies some of what you're trying to do. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
James FitzGibbon wrote: On 8/1/07, *Linux Lover* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: This SOHO PBX box won't interop with Asterisk because it doesn't speak any of the protocols that Asterisk does. This box I tend agree with your evaluation. Still, I was thinking that since all these el-cheapo SOHO PBX boxes support manual attendant call transfer, what's to prevent Asterisk from mimicking an attendant by sending proper DTMF signals and make this box transfer the call to the single analog phone in the business? That is, Asterisk will connect (via RJ-11) to the unit as the attendant's phone, and my real phone (only one in the system) will connect via a second RJ-11 (there could be 4 of them). Or is Asterisk not capable of sending DTMF signals over an RJ-11 connection? You can send arbitrary DTMF over any of Asterisk's channels from the dialplan. I just figured that this level of integration was a bit deeper than you were looking for as a first project. It would be an interesting experiment, to be sure. The biggest issue I'd think would be feedback - you can send the DTMF along the wire, but how do you know that the SOHO box interpreted it correctly? If the only feedback is designed for a human ( i.e. auditory), then interpreting those cues with Asterisk would be non-trivial. Do I undestand correctly that with this solution, I will still be able to connect to my analog Verizon phone line with the SIP phone? That is, the outside world will see my phone as an ordinary phone, when in fact I am using a SIP phone? If so, that means that Asterisk does all the magic behind the scene, right? Yes, your Verizon POTS line would go into a FXO port in your server (which in Asterisk would be referenced as the channel Zap/1 - zaptel being Asterisk's TDM driver) and your SIP phone would connect via your standard office network and be referenced as SIP/whateverusernameyouwant. A very simplistic example of bridging a call would be: [from-verizon] exten = s,1,Dial(SIP/whateverusername) Assuming that you'd configured zaptel to route calls that come in on the FXO port to the Asterisk context named from-verizon, then any such calls would immediately cause Asterisk to ring your SIP phone, and if answered to bridge the two calls together. A more complex example that makes them press one to call you and otherwise lets them leave a message: [from-verizon] exten = s,1,Background(Press1ToTalkOr2ToLeaveAMessage) exten = s,n,WaitExten(10) ; timeout exten = t,1,Goto(vm,1) ; invalid exten = i,1,Goto(vm,1) ; press 1 exten = 1,1,Dial(SIP/101,20) exten = 1,n,Goto(vm,1) ; press 2 exten = 2,1,Goto(vm,1) ; all voicemail activity ends up here exten = vm,1,VoiceMail(u101) exten = vm,n,Hangup [from-officephone] exten = *98,1,VoiceMailMain extne = *98,n,Hangup Assuming you've now set up your SIP phone as extension 101, this would play a sound file saying press 1 to talk to 2 to leave a message. If they press 1, your SIP phone rings. If they press 2, they go to voicemail. If they wait 10 seconds without pressing anything, or press something other than 1 or 2, they also go to voicemail. If they press 1 to dial your phone and you don't pick up after 20 seconds, they go to voicemail. On your deskphone (could just as easily be a SIP softphone if you prefer), you can dial *98 to log in and pick up your new voicemail messages. Hope that demystifies some of what you're trying to do. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users the way to have * send dtmf is with the D option, w inserts a half second pause. As an example I have a remote location that needs special 911, so they have a landline that connects to a linksys SPA, it doesnt like being passed the destination number through sip, so O do it this way: exten = 911,1,Dial(SIP/08CCB243-911,,D(w911)) works awesome, it connects, plays back the DTMF, and then passes the audio stream to the caller. Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Wow! Thank you so much, James - you have certainly clarified lots of things in my mind. You are correct about me overlooking the feedback issue (with the el-cheapo device). I see that I have to learn. This world of VoIP is new and mind boggling - to me. Thanks, Lynn --- James FitzGibbon [EMAIL PROTECTED] wrote: On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote: This SOHO PBX box won't interop with Asterisk because it doesn't speak any of the protocols that Asterisk does. This box I tend agree with your evaluation. Still, I was thinking that since all these el-cheapo SOHO PBX boxes support manual attendant call transfer, what's to prevent Asterisk from mimicking an attendant by sending proper DTMF signals and make this box transfer the call to the single analog phone in the business? That is, Asterisk will connect (via RJ-11) to the unit as the attendant's phone, and my real phone (only one in the system) will connect via a second RJ-11 (there could be 4 of them). Or is Asterisk not capable of sending DTMF signals over an RJ-11 connection? You can send arbitrary DTMF over any of Asterisk's channels from the dialplan. I just figured that this level of integration was a bit deeper than you were looking for as a first project. It would be an interesting experiment, to be sure. The biggest issue I'd think would be feedback - you can send the DTMF along the wire, but how do you know that the SOHO box interpreted it correctly? If the only feedback is designed for a human (i.e. auditory), then interpreting those cues with Asterisk would be non-trivial. Do I undestand correctly that with this solution, I will still be able to connect to my analog Verizon phone line with the SIP phone? That is, the outside world will see my phone as an ordinary phone, when in fact I am using a SIP phone? If so, that means that Asterisk does all the magic behind the scene, right? Yes, your Verizon POTS line would go into a FXO port in your server (which in Asterisk would be referenced as the channel Zap/1 - zaptel being Asterisk's TDM driver) and your SIP phone would connect via your standard office network and be referenced as SIP/whateverusernameyouwant. A very simplistic example of bridging a call would be: [from-verizon] exten = s,1,Dial(SIP/whateverusername) Assuming that you'd configured zaptel to route calls that come in on the FXO port to the Asterisk context named from-verizon, then any such calls would immediately cause Asterisk to ring your SIP phone, and if answered to bridge the two calls together. A more complex example that makes them press one to call you and otherwise lets them leave a message: [from-verizon] exten = s,1,Background(Press1ToTalkOr2ToLeaveAMessage) exten = s,n,WaitExten(10) ; timeout exten = t,1,Goto(vm,1) ; invalid exten = i,1,Goto(vm,1) ; press 1 exten = 1,1,Dial(SIP/101,20) exten = 1,n,Goto(vm,1) ; press 2 exten = 2,1,Goto(vm,1) ; all voicemail activity ends up here exten = vm,1,VoiceMail(u101) exten = vm,n,Hangup [from-officephone] exten = *98,1,VoiceMailMain extne = *98,n,Hangup Assuming you've now set up your SIP phone as extension 101, this would play a sound file saying press 1 to talk to 2 to leave a message. If they press 1, your SIP phone rings. If they press 2, they go to voicemail. If they wait 10 seconds without pressing anything, or press something other than 1 or 2, they also go to voicemail. If they press 1 to dial your phone and you don't pick up after 20 seconds, they go to voicemail. On your deskphone (could just as easily be a SIP softphone if you prefer), you can dial *98 to log in and pick up your new voicemail messages. Hope that demystifies some of what you're trying to do. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware that can ring my phone?
Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Lynn, If I understand you question correctly, you would need: A computer (preferably a server) to run Asterisk An analog interface card such as the Digium TDM400P An analog phone line (POTS) An analog (real) phone Calls would come in on the POTS line, get answered by Asterisk. Callers would hear your voice menu, and input their choice. If they opted for a live person, asterisk would then send the call to your analog (real) phone. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 8/1/2007 8:48:47 AM Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Lynn, What you need is an ATA (analog telephone adapter). The ATA is a SIP or IAX extension on your Asterisk box, and your standard phone plugs into it. Asterisk sends the call to the SIP extension (the ATA), and the ATA rings your phone. On the flip side, your phone dials normally and the ATA digitizes the data and sends it via SIP to Asterisk for routing. Check out Digium's IAXy or the GrandStream Budgetone/HandyTone. AR On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote: Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Quoting Linux Lover [EMAIL PROTECTED]: any of the various module based cards with one fxo and one fxs port will do what you want. Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
On Wed, 2007-08-01 at 06:48 -0700, Linux Lover wrote: I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. Assuming the incoming call is coming in on an analog phone line, you need a card such as the Digium TDM11B... this provides one FXO port (for connecting to the incoming phone line) and one FXS port (for ringing an analog phone). It's not the only way to do it, but it's probably the easiest (and a great way to get started with Asterisk, I might add). -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Yes, you understood correctly. Thank you - and all others who replied so quickly - for your precise and guiding answers. The Digium TDM11B looks looks like the perfect match for me: http://www.telephonyware.com/telephonyware/tw00068.html But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? http://www.soho-pbx.com/sp-104.htm Do you think this could work for me or did I expose a gross misconception on my part? Thanks, Lynn --- john beaman [EMAIL PROTECTED] wrote: Lynn, If I understand you question correctly, you would need: A computer (preferably a server) to run Asterisk An analog interface card such as the Digium TDM400P An analog phone line (POTS) An analog (real) phone Calls would come in on the POTS line, get answered by Asterisk. Callers would hear your voice menu, and input their choice. If they opted for a live person, asterisk would then send the call to your analog (real) phone. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 8/1/2007 8:48:47 AM Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Hi Lynn, You can use a Linksys SPA-3102 for both FXO (POTS) and FXS (phone) connection instead of a Digium card. The price is around $90-100. Almost any old PC will do if it can run Linux. There are also other alternatives to a PC such as the Linksys WRT54GL. I use a Linksys NSLU2 (Slug) at home, they are about the same price as an SPA and they are really tiny! regards, Drew Linux Lover wrote: Yes, you understood correctly. Thank you - and all others who replied so quickly - for your precise and guiding answers. The Digium TDM11B looks looks like the perfect match for me: http://www.telephonyware.com/telephonyware/tw00068.html But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? http://www.soho-pbx.com/sp-104.htm Do you think this could work for me or did I expose a gross misconception on my part? Thanks, Lynn --- john beaman [EMAIL PROTECTED] wrote: Lynn, If I understand you question correctly, you would need: A computer (preferably a server) to run Asterisk An analog interface card such as the Digium TDM400P An analog phone line (POTS) An analog (real) phone Calls would come in on the POTS line, get answered by Asterisk. Callers would hear your voice menu, and input their choice. If they opted for a live person, asterisk would then send the call to your analog (real) phone. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 8/1/2007 8:48:47 AM Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Lynn, I am unfamiliar with soho-pbx, so I cannot comment on quality, service, configurability, etc. They are based out of Hong Kong, and their box is probably already running some flavor of Asterisk, so you would need nothing additional except for the phone line coming in and the telephone. I got quite a kick out of their description for the SP-104 box as referenced by your link: The photos below are model SP-104, a model that costs only tens of US dollars Not sure how much that comes to, but sounds pretty cheap... John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 8/1/2007 10:29:51 AM Yes, you understood correctly. Thank you - and all others who replied so quickly - for your precise and guiding answers. The Digium TDM11B looks looks like the perfect match for me: http://www.telephonyware.com/telephonyware/tw00068.html But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? http://www.soho-pbx.com/sp-104.htm Do you think this could work for me or did I expose a gross misconception on my part? Thanks, Lynn --- john beaman [EMAIL PROTECTED] wrote: Lynn, If I understand you question correctly, you would need: A computer (preferably a server) to run Asterisk An analog interface card such as the Digium TDM400P An analog phone line (POTS) An analog (real) phone Calls would come in on the POTS line, get answered by Asterisk. Callers would hear your voice menu, and input their choice. If they opted for a live person, asterisk would then send the call to your analog (real) phone. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 8/1/2007 8:48:47 AM Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in
Re: [asterisk-users] Hardware that can ring my phone?
This is what I have at home and it works okay. I also added an SPA-2002 (~$70) that adds another two FXS (phone) ports for a total of three. Godspeed, Phil Drew Gibson wrote: Hi Lynn, You can use a Linksys SPA-3102 for both FXO (POTS) and FXS (phone) connection instead of a Digium card. The price is around $90-100. Almost any old PC will do if it can run Linux. There are also other alternatives to a PC such as the Linksys WRT54GL. I use a Linksys NSLU2 (Slug) at home, they are about the same price as an SPA and they are really tiny! regards, Drew Linux Lover wrote: Yes, you understood correctly. Thank you - and all others who replied so quickly - for your precise and guiding answers. The Digium TDM11B looks looks like the perfect match for me: http://www.telephonyware.com/telephonyware/tw00068.html But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? http://www.soho-pbx.com/sp-104.htm Do you think this could work for me or did I expose a gross misconception on my part? Thanks, Lynn --- john beaman [EMAIL PROTECTED] wrote: Lynn, If I understand you question correctly, you would need: A computer (preferably a server) to run Asterisk An analog interface card such as the Digium TDM400P An analog phone line (POTS) An analog (real) phone Calls would come in on the POTS line, get answered by Asterisk. Callers would hear your voice menu, and input their choice. If they opted for a live person, asterisk would then send the call to your analog (real) phone. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 8/1/2007 8:48:47 AM Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote: But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? http://www.soho-pbx.com/sp-104.htm Do you think this could work for me or did I expose a gross misconception on my part? This SOHO PBX box won't interop with Asterisk because it doesn't speak any of the protocols that Asterisk does. This box appears to be a solid-state (and I'd assume very feature restricted) alternative to Asterisk. That it happens to have both FXO (to the Telco) and FXS (to the analog phone) ports doesn't mean that it is usable as an analog interface for Asterisk. Your best bet is to find your closest Asterisk user's group and see when they're next doing a build seminar. Most user groups do these a few times a year and you might be able to find someone who will do one on demand. You bring some cheapo PC you might have lying around and buy a $20 FXO card and build a simple answering machine using Asterisk. From there, it's easy to extend so that when the user chooses a particular option in your IVR, the call is bridged to a phone in your office. The original single-FXO-port card from Digium was the X100P. These aren't sold anymore (the TDM400B modular card replaced it), but they can be found on eBay for $10-$30. If you can get your hands on one, you might consider going with a cheap SIP phone instead of a analog phone for your business. There isn't (as far as I know) a readily available cheap single-FXS-port card. If you go with an analog phone behind Asterisk, you'll need an FXS port. If you go with a SIP phone, you just need to have a network connection from the phone to the server, which might be cheaper. A quick search on eBay shows a few Grandstream Budgetone 101 phones (certainly not the best available, but they'll do the job) in the sub-$50 range. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
A phone system for under $100 is asking a lot. It can be done, but what is your time worth. You might want to consider some other phone system if all you need is IVR and analog support or look at hosted solutions. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Linux Lover Sent: Wednesday, August 01, 2007 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hardware that can ring my phone? Yes, you understood correctly. Thank you - and all others who replied so quickly - for your precise and guiding answers. The Digium TDM11B looks looks like the perfect match for me: http://www.telephonyware.com/telephonyware/tw00068.html But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? http://www.soho-pbx.com/sp-104.htm Do you think this could work for me or did I expose a gross misconception on my part? Thanks, Lynn --- john beaman [EMAIL PROTECTED] wrote: Lynn, If I understand you question correctly, you would need: A computer (preferably a server) to run Asterisk An analog interface card such as the Digium TDM400P An analog phone line (POTS) An analog (real) phone Calls would come in on the POTS line, get answered by Asterisk. Callers would hear your voice menu, and input their choice. If they opted for a live person, asterisk would then send the call to your analog (real) phone. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 8/1/2007 8:48:47 AM Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn __ __ Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ __ Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Do you think you'll outgrow 1 phone line any time soon. If so You'll want something that you don't have to completely redo when you add the next line. That digium card you linked to has 2 more expansion slots open for more lines or phones. The soho pbx you linked to looks like you can have more phones, but only one line, so you'd have to get the 2 or more line model later if your business grew. searching. Oh, it looks like the a a SP-416 on Ebay for $99 + $108 shipping from Hong Kong I don't know what is inside of the box, but it looks like an interesting product. Linux Lover wrote: Yes, you understood correctly. Thank you - and all others who replied so quickly - for your precise and guiding answers. The Digium TDM11B looks looks like the perfect match for me: http://www.telephonyware.com/telephonyware/tw00068.html But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? http://www.soho-pbx.com/sp-104.htm Do you think this could work for me or did I expose a gross misconception on my part? Thanks, Lynn --- john beaman [EMAIL PROTECTED] wrote: Lynn, If I understand you question correctly, you would need: A computer (preferably a server) to run Asterisk An analog interface card such as the Digium TDM400P An analog phone line (POTS) An analog (real) phone Calls would come in on the POTS line, get answered by Asterisk. Callers would hear your voice menu, and input their choice. If they opted for a live person, asterisk would then send the call to your analog (real) phone. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 8/1/2007 8:48:47 AM Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Linux Lover wrote: But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? Buy this, or another proven SOHO solution and forget, for the moment, Asterisk IF you really are starting out fresh, begin the business and learn Asterisk in your ( ha! ) spare time, then when you are ready you will be able to migrate without having your customers suffer through your learning curve. You DO want your business to succeed, don't you? You DON'T want your customers to have a bad first impression of you because of some small problem with Asterisk. Look at other hardware Asterisk solutions as well. The X100 can be more trouble than it is worth, the TDM400 CAN have issues with some motherboards that you will not discover until the driver can't find the board, and the standard support answer is try another motherboard Sangoma makes some nice FXO/FXS PCI cards as well with a 5 year warranty. John Novack -- Dog is my co-pilot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
John, thank you very much. Indeed, this is the direction I was thinking of taking. I just needed a quick dirty solution for the short term - I didn't realize that Asterisk is so complex. In fact, I am not sure I completely understand it: Will using Asterisk force me to use an external VoIP service? Or can I remain completely POTS based? (At the volume of phone calls that I am making, I found out that using an ordinary phone line is way cheaper than any VoIP service available to me right now - definitely cheaper tha Vonage et al.) Thanks, Lynn --- John Novack [EMAIL PROTECTED] wrote: Linux Lover wrote: But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? Buy this, or another proven SOHO solution and forget, for the moment, Asterisk IF you really are starting out fresh, begin the business and learn Asterisk in your ( ha! ) spare time, then when you are ready you will be able to migrate without having your customers suffer through your learning curve. You DO want your business to succeed, don't you? You DON'T want your customers to have a bad first impression of you because of some small problem with Asterisk. Look at other hardware Asterisk solutions as well. The X100 can be more trouble than it is worth, the TDM400 CAN have issues with some motherboards that you will not discover until the driver can't find the board, and the standard support answer is try another motherboard Sangoma makes some nice FXO/FXS PCI cards as well with a 5 year warranty. John Novack -- Dog is my co-pilot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Quoting Linux Lover [EMAIL PROTECTED]: John, thank you very much. Indeed, this is the direction I was thinking of taking. I just needed a quick dirty solution for the short term - I didn't realize that Asterisk is so complex. In fact, I am not sure I completely understand it: Will using Asterisk force me to use an external VoIP service? Or can I remain completely POTS based? (At the volume of phone calls that I am making, I found out that using an ordinary phone line is way cheaper than any VoIP service available to me right now - definitely cheaper tha Vonage et al.) there are plenty of supercheap voip services (far cheaper than a business line), but asterisk allows you to mix and match whatever you want from analog lines to digital lines to voip services, with softphones, voip phones, traditionalphones etc., in whatever combination you want. you just configure the appropriate channels for whatever you connect to it. Thanks, Lynn --- John Novack [EMAIL PROTECTED] wrote: Linux Lover wrote: But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? Buy this, or another proven SOHO solution and forget, for the moment, Asterisk IF you really are starting out fresh, begin the business and learn Asterisk in your ( ha! ) spare time, then when you are ready you will be able to migrate without having your customers suffer through your learning curve. You DO want your business to succeed, don't you? You DON'T want your customers to have a bad first impression of you because of some small problem with Asterisk. Look at other hardware Asterisk solutions as well. The X100 can be more trouble than it is worth, the TDM400 CAN have issues with some motherboards that you will not discover until the driver can't find the board, and the standard support answer is try another motherboard Sangoma makes some nice FXO/FXS PCI cards as well with a 5 year warranty. John Novack -- Dog is my co-pilot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
James, thank you for your educating answer. --- James FitzGibbon [EMAIL PROTECTED] wrote: This SOHO PBX box won't interop with Asterisk because it doesn't speak any of the protocols that Asterisk does. This box appears to be a solid-state (and I'd assume very feature restricted) alternative to Asterisk. That it happens to have both FXO (to the Telco) and FXS (to the analog phone) ports doesn't mean that it is usable as an analog interface for Asterisk. I tend agree with your evaluation. Still, I was thinking that since all these el-cheapo SOHO PBX boxes support manual attendant call transfer, what's to prevent Asterisk from mimicking an attendant by sending proper DTMF signals and make this box transfer the call to the single analog phone in the business? That is, Asterisk will connect (via RJ-11) to the unit as the attendant's phone, and my real phone (only one in the system) will connect via a second RJ-11 (there could be 4 of them). Or is Asterisk not capable of sending DTMF signals over an RJ-11 connection? Not that I am rushing to buy this cheap box right now, but I am curious whether this is possible at all - perhaps to get a better feel of how flexible Asterisk is. The original single-FXO-port card from Digium was the X100P. These aren't sold anymore (the TDM400B modular card replaced it), but they can be found on eBay for $10-$30. If you can get your hands on one, you might consider going with a cheap SIP phone instead of a analog phone for your business. There isn't (as far as I know) a readily available cheap single-FXS-port card. If you go with an analog phone behind Asterisk, you'll need an FXS port. If you go with a SIP phone, you just need to have a network connection from the phone to the server, which might be cheaper. A quick search on eBay shows a few Grandstream Budgetone 101 phones (certainly not the best available, but they'll do the job) in the sub-$50 range. Do I undestand correctly that with this solution, I will still be able to connect to my analog Verizon phone line with the SIP phone? That is, the outside world will see my phone as an ordinary phone, when in fact I am using a SIP phone? If so, that means that Asterisk does all the magic behind the scene, right? Thanks, Lynn Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
you would still need an fxo port of some sort for asterisk for it to pretend to be a phone. Quoting Linux Lover [EMAIL PROTECTED]: James, thank you for your educating answer. --- James FitzGibbon [EMAIL PROTECTED] wrote: This SOHO PBX box won't interop with Asterisk because it doesn't speak any of the protocols that Asterisk does. This box appears to be a solid-state (and I'd assume very feature restricted) alternative to Asterisk. That it happens to have both FXO (to the Telco) and FXS (to the analog phone) ports doesn't mean that it is usable as an analog interface for Asterisk. I tend agree with your evaluation. Still, I was thinking that since all these el-cheapo SOHO PBX boxes support manual attendant call transfer, what's to prevent Asterisk from mimicking an attendant by sending proper DTMF signals and make this box transfer the call to the single analog phone in the business? That is, Asterisk will connect (via RJ-11) to the unit as the attendant's phone, and my real phone (only one in the system) will connect via a second RJ-11 (there could be 4 of them). Or is Asterisk not capable of sending DTMF signals over an RJ-11 connection? Not that I am rushing to buy this cheap box right now, but I am curious whether this is possible at all - perhaps to get a better feel of how flexible Asterisk is. The original single-FXO-port card from Digium was the X100P. These aren't sold anymore (the TDM400B modular card replaced it), but they can be found on eBay for $10-$30. If you can get your hands on one, you might consider going with a cheap SIP phone instead of a analog phone for your business. There isn't (as far as I know) a readily available cheap single-FXS-port card. If you go with an analog phone behind Asterisk, you'll need an FXS port. If you go with a SIP phone, you just need to have a network connection from the phone to the server, which might be cheaper. A quick search on eBay shows a few Grandstream Budgetone 101 phones (certainly not the best available, but they'll do the job) in the sub-$50 range. Do I undestand correctly that with this solution, I will still be able to connect to my analog Verizon phone line with the SIP phone? That is, the outside world will see my phone as an ordinary phone, when in fact I am using a SIP phone? If so, that means that Asterisk does all the magic behind the scene, right? Thanks, Lynn Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
You can use a Linksys SPA-3102 for both FXO (POTS) and FXS (phone) connection instead of a Digium card. The price is around $90-100. Almost any old PC will do if it can run Linux. There are also other alternatives to a PC such as the Linksys WRT54GL. The OpenWRT (on whatever supported router hardware) + SPA-3102 is a pretty decent combo. You can reinvite the traffic between the FXO and FXS (g711 only) and get good quality without even taxing the router. FYI, a WRT54G had no problem running asterisk 1.2.x with 4 concurrent channels (g711, no transcoding, just RTP proxying). I'd look into something like that. And you can expand it fairly easily by adding another SPA for a second line. Luki ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware spec comparison
On 5 Jun 2007, at 22:01, Adrian Marsh wrote: Yeah I've heard the same breaks in conversations myself. It simply goes silent for a few seconds - making both parties say the usual sorry.. Missed that can you say again?... Connection quality via remote SIP (outside our network via internet) can be terrible (using GSM), though obviously theres a whole bunch of other issues there, so I'm focusing just on the internal network and IAX. Our connectivity to our IAX/PSTN provider is u-law, so should be ok but isn't. What does -p do ? Adrian -Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: 05 June 2007 19:21 To: Adrian Marsh Subject: Re: [asterisk-users] Hardware spec comparison On Tue, Jun 05, 2007 at 06:51:40PM +0100, Adrian Marsh wrote: All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality, especially on IAX2 calls, silences etc. Can you verify those complaints? What codec do you use? I hope that ulaw/alaw and not some compressed codec. I'm wondering whether or not the machine needs a faster CPU, something like a Duo, as I can't find any faults anywhere else that might cause these blips. But the 2% CPU usage seems to suggest it shouldn't. try running asterisk with the option -p That might be a bit memory light I'd put 1Gb in that box. Take a look at vmstat and see if it is swapping. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware spec comparison
All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality, especially on IAX2 calls, silences etc. I'm wondering whether or not the machine needs a faster CPU, something like a Duo, as I can't find any faults anywhere else that might cause these blips. But the 2% CPU usage seems to suggest it shouldn't. Adrian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hardware spec comparison
It could also be that network congestion is causing the quality degradation. Do you have quality of service configured on the LAN? You mention that you are IP only; does that mean you are doing local traffic only or are you connecting to the public network via your internet connection. If you are connecting outside your site, your outside connection could be saturated or having latency problems, etc. Also, you could look at memory usage as Asterisk is a memory intensive program. So there's a few more things for you to think about. Bobby -Original Message- All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality, especially on IAX2 calls, silences etc. I'm wondering whether or not the machine needs a faster CPU, something like a Duo, as I can't find any faults anywhere else that might cause these blips. But the 2% CPU usage seems to suggest it shouldn't. Adrian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware spec comparison
On Tue, 5 Jun 2007, Adrian Marsh wrote: All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality, especially on IAX2 calls, silences etc. I'm wondering whether or not the machine needs a faster CPU, something like a Duo, as I can't find any faults anywhere else that might cause these blips. But the 2% CPU usage seems to suggest it shouldn't. One thing that's been mentioned in the past is the accuracy of the timing source. I'm presuming you have ztdummy loaded... No first-hand experiences of badness with just ztdummy and that sort of call load myself though... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hardware spec comparison
Yeah I've heard the same breaks in conversations myself. It simply goes silent for a few seconds - making both parties say the usual sorry.. Missed that can you say again?... Connection quality via remote SIP (outside our network via internet) can be terrible (using GSM), though obviously theres a whole bunch of other issues there, so I'm focusing just on the internal network and IAX. Our connectivity to our IAX/PSTN provider is u-law, so should be ok but isn't. What does -p do ? Adrian -Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: 05 June 2007 19:21 To: Adrian Marsh Subject: Re: [asterisk-users] Hardware spec comparison On Tue, Jun 05, 2007 at 06:51:40PM +0100, Adrian Marsh wrote: All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality, especially on IAX2 calls, silences etc. Can you verify those complaints? What codec do you use? I hope that ulaw/alaw and not some compressed codec. I'm wondering whether or not the machine needs a faster CPU, something like a Duo, as I can't find any faults anywhere else that might cause these blips. But the 2% CPU usage seems to suggest it shouldn't. try running asterisk with the option -p -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware Echo cancellor Digium Wildcard TE212P
Hi, I am currently using TE110P Digium card on a PRI card. Basically the echo is so much that one can disticntly identify that. I have tried all the combination if tuning configuration seen in forums etc. I am using MG2 cancellor algorithm also tuned the RX TX gains, still there is an echo. So I am thing to purchase an hardware based echo cancellor like Digium Wildcard TE212P. So in this regards I would like to get some view whether its worth to buy a hardwrae based echo cancellor. Will this resolve the issue, or will be just waste of money. I am using Asterisk 1.2.18 latest version of zaptel drivers. Hope if someone had the same issue, I what has done to resolve it would be much appreciable. -- Deepak - Yahoo! Answers - Got a question? Someone out there knows the answer. Tryit now.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Echo cancellor Digium Wildcard TE212P
Deepak Naidu wrote: Hi, I am currently using TE110P Digium card on a PRI card. Basically the echo is so much that one can disticntly identify that. I have tried all the combination if tuning configuration seen in forums etc. I am using MG2 cancellor algorithm also tuned the RX TX gains, still there is an echo. So I am thing to purchase an hardware based echo cancellor like Digium Wildcard TE212P. So in this regards I would like to get some view whether its worth to buy a hardwrae based echo cancellor. Will this resolve the issue, or will be just waste of money. I am using Asterisk 1.2.18 latest version of zaptel drivers. Hope if someone had the same issue, I what has done to resolve it would be much appreciable. In my experience, it is well worth the money. After installing several for customers, we never bought the non-HWEC cards again... Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Echo cancellor Digium Wildcard TE212P
So Steven, did the echo problem stopped once the Hardware echo cancellor card was installed out of the box, or you needed to do some configuration changes like Rx Tx etc. Thanks for sharing your experience. -- Deepak »Steven Ringwald« [EMAIL PROTECTED] wrote: Deepak Naidu wrote: Hi, I am currently using TE110P Digium card on a PRI card. Basically the echo is so much that one can disticntly identify that. I have tried all the combination if tuning configuration seen in forums etc. I am using MG2 cancellor algorithm also tuned the RX TX gains, still there is an echo. So I am thing to purchase an hardware based echo cancellor like Digium Wildcard TE212P. So in this regards I would like to get some view whether its worth to buy a hardwrae based echo cancellor. Will this resolve the issue, or will be just waste of money. I am using Asterisk 1.2.18 latest version of zaptel drivers. Hope if someone had the same issue, I what has done to resolve it would be much appreciable. In my experience, it is well worth the money. After installing several for customers, we never bought the non-HWEC cards again... Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
