Re: [Asterisk-Users] zaprtc load issue (different that other postings)
zaprtc does not work with smp systems, unfortunately. There is some discussion on the wiki about the bristuff zaprtc module working with multi cpu systems, however. Link: http://www.voip-info.org/wiki-Asterisk+timer Brian On Sat, 22 Jan 2005 22:37:42 -0800, Spencer Nassar [EMAIL PROTECTED] wrote: zaprtc 'load' is failing on my machine (the make was fine, same output as other posts to this list) [EMAIL PROTECTED] zaptelrtc]# make load sync modprobe zaptel insmod ./zaprtc.o ./zaprtc.o: init_module: Input/output error Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg I've seen other references to ensuring that rtc is unloaded first, but it's not loaded on my machine [EMAIL PROTECTED] zaptelrtc]# lsmod Module Size Used byNot tainted zaptel181856 0 soundcore 7012 0 (autoclean) e1000 75744 1 iptable_nat22744 0 (autoclean) (unused) iptable_mangle 2776 0 (autoclean) (unused) ipt_REJECT 4632 1 (autoclean) ipt_multiport 1176 2 (autoclean) ipt_state 1080 3 (autoclean) ip_conntrack 29704 2 (autoclean) [iptable_nat ipt_state] iptable_filter 2412 1 (autoclean) ip_tables 16544 8 [iptable_nat iptable_mangle ipt_REJECT ipt_multiport ipt_state iptable_filter] microcode 6848 0 (autoclean) keybdev 2976 0 (unused) mousedev5624 0 (unused) hid22276 0 (unused) input 6144 0 [keybdev mousedev hid] usb-ohci 23176 0 (unused) usbcore80928 1 [hid usb-ohci] ext3 89960 2 jbd55060 2 [ext3] mptscsih 41780 3 mptbase43936 3 [mptscsih] sd_mod 13360 6 scsi_mod 112680 2 [mptscsih sd_mod] I'm running Redhat ES3 on a dual xeon system Kernel is linux-2.4 - linux-2.4.21-15.EL Any ideas? Thanks! Spencer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best VPN server for * and woad warriors using windows?
Hi list! I'm sure the topic has been discussed but I could not find what I was looking for. What would be the best / easiest VPN software solution. I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. Ideally the vpn solution will not require any additional software on the client side but will use IPSEC. (Ofcourse call quality is important) There are numerous vpn server daemons around and I found many messages about some of them using tcp/udp etc and instead of trying them all out hopefully someone can recommend one? (I guess this would make a useful wiki page too). Thanks!! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VPN server for * and woad warriors using windows?
Hi, Il giorno dom, 23-01-2005 alle 10:33 +0100, Remco Barende ha scritto: What would be the best / easiest VPN software solution. I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. Ideally the vpn solution will not require any additional software on the client side but will use IPSEC. (Ofcourse call quality is important) best if ofcourse some ipsec-based solutions, but that leads to installing a client on winblow machines. You can use pptp, ok is not secure as ipsec but is built in in winblow 98,2k,xp... so on the client you must only create a new VPN connection (under connections manager) and you're done. On the linux side, go to http://poptop.sourceforge.net/dox/ to grab the server. I think that this is the easiest solutions for a decent encryption ad ease of use, when using m$ clients. (hoping you don't need to protect millions $$$ value data : ) of course ipsec is better, but needs more work to set it up, on client and on server side. just my 2 cents, Matteo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone but visible in logfile
Try the variable PRI_NETWORK_CID instead of CALLERIDNUM
Peer Oliver Schmidt wrote:
Jens, thanks for the feedback.
I've added a ZAPHFC card to my CAPI based system. Calls coming in via
ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
via CAPI do forward the caller id to the SIP phones.
I think you didn't set usecallerid=yes in your zapata.conf?
Added it, rebooted, no change. (Before, I just had pritrustusercid =
yes, only.)
Another way is to set the callerid in your extensions.conf via exten
= 807440,2,SetCIDNum(0${CALLERIDNUM}).
Changed it, now the funny part comes:
extensions.conf
exten = 807440,1,Answer
exten = 807440,2,SetCIDNum(0${CALLERIDNUM})
exten = 807440,3,Dial(SIP/26,20,t)
exten = 807440,3,VoiceMail2(su25)
exten = 807440,103,VoiceMail2(sb25)
exten = 807440,104,Hangup
but the log says:
-- Accepting call from '1729731418' to '807440' on channel 0/1, span 1
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing SetCIDNum(Zap/1-1, 0) in new stack
It does not add the callerid it has two lines above ???
I know there have been some changes to the CID structure sometime
within Asterisk. But, this is using the bristuff download and install
script.
The same problem happens using the debian packages (1.0.3) from
marlow.dk.
Any and all help is greatly appreciated.
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Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone but visible in logfile
Hello Stephan,
Another way is to set the callerid in your extensions.conf via exten
= 807440,2,SetCIDNum(0${CALLERIDNUM}).
Try the variable PRI_NETWORK_CID instead of CALLERIDNUM
This did the trick. I will go and update the Wiki,,,
Thanks and have a good weekend.
--
Best regards
Peer Oliver Schmidt
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RE: [Asterisk-Users] Anyone know where a good source of mailing liststats might be found?
Thanks to everyone who provided feedback. [EMAIL PROTECTED] wrote: Folks, I'm curious to know how the volume of Asterisk-Users rates as far mailing lists go. This list sees over 200 messages per day, which has GOT to put it in the top 5%, doesn't it? I'd love to know if anyone has knowledge of any organization that might maintain such stats. Regards, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VPN server for * and woad warriors using windows?
On Sun, 23 Jan 2005 10:33:14 +0100 (CET), Remco Barende [EMAIL PROTECTED] wrote: I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. OK, take a hard look at this before you get too far. Installing VPN software *on* the Asterisk box is not a good idea. Now, you haven't explained the volume of users on the box, or the availability needs of the box, but either way, this is bad practice. The term roadwarriors' makes me think this is for a business. There are numerous vpn server daemons around and I found many messages about some of them using tcp/udp etc and instead of trying them all out hopefully someone can recommend one? If you want IPSec, take a look at OpenWall. If you must run this on your asterisk box, so be it. Now, if I were you, I would take this opportunity to install a good Linux based firewall solution that sits in FRONT of the asterisk server. I can't stress this enough. Take a look at m0n0wall. It has vpn support (ipsec and pptp) built in, and it will run on nearly anything. Put this on a machine by itself. http://m0n0.ch/wall/ (I guess this would make a useful wiki page too). Thanks!! Remco Hope this helps! -Chuji ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?
On Sat, 2005-01-22 at 23:56 -0800, Kenneth Long wrote: seem like some kind of port issue... Probably. Both try to set up listeners on the IAX port (4569 for IAX2). Disable or reconfigure one of them to bind to a different port, whichever you want to answer on it. Also, don't forget to disable chan_alsa and chan_oss in modules.conf. When running another client you won't want the * console hogging your soundcard. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delay before dialing extension on Zap channel
Hi, After using Asterisk with a SIP hardphone for a couple of weeks I've just installed a TDM400P card. My hardphone - a 7940 - allows me to use a dialplan to decide when a particular extension is complete and automatically trigger dialing. This works well with my internal extensions, which are all of the form Z00. When trying to dial these extensions from a handset connected to a Zap channel there appears to be a delay of about 4 seconds between the time I dial Z00 and the time asterisk decides I've finished dialing and connects me. Is there any way to reduce this delay? I'd ideally like asterisk to dial the extension as soon as it matches a valid extension. -Ronan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?
Hi! I can run iaxcomm by itself...and I start up Asterisk on it own... But if I start Asterisk first, then launch iaxcomm I get this error: You really do not want to run Asterisk and X-Windows on the same box. Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to debug core-file
Hi
I'm running safe_asterisk, but get core-files in /tmp - how do I debug
them ?
I know gdb asterisk core.12370
and bt full
But it didn't show anything usefull for me.
Can anyone help me ?
(Running asterisk 1.0.2 with ast_data
/Hans-Henrik
-
Last from bt full:
priority=200, callerid=0x81b8e90 Dial, action=1134845864) at
pbx.c:1384
e = (struct ast_exten *) 0x81bdcf0
sw = (struct ast_switch *) 0x0
data = 0x0
newstack = 1
res = 1134845864
status = 4
incstack = {0x0 repeats 23 times, 0x4000 Address
0x4000 out of bounds, 0x4025ba20 , 0x0, 0x0, 0x8fd0ee8
1106487988.3352,
0x43a4775c 1106487988.3352, 0x43a45fa8 SIP/000b82027e34-0205,
0x46600d84 ä\016`Ft\214\a\b¨_¤C, 0x808f25f \213\233à\003,
0x8fd0ee8 1106487988.3352, 0x43a4775c 1106487988.3352, 0x1f
Address 0x1f out of bounds, 0x4025ba20 , 0x0, 0x0, 0x0, 0x400382ae
\201ÃÞC, 0x0,
0x4025ba20 , 0x46600d1c 000b82027e34, 0x0, 0x62303030 Address
0x62303030 out of bounds, 0x32303238 Address 0x32303238 out of bounds,
0x34336537 Address 0x34336537 out of bounds, 0x0 repeats 12
times, 0x808fcfa \213\233à\003, 0x8fd0ea8 ´ªóAdö\002, 0x0, 0x0, 0x0}
passdata = zap/g1/00551138856342|120|rtS(10883), '\0'
repeats 8155 times
stacklen = 0
tmp = \e[1;36;40mDial\e[0;37;40m\000m\000;40m\00040m, '\0'
repeats 44 times
tmp2 = \e[1;35;40mSIP/000b82027e34-0205\e[0;37;40m, '\0'
repeats 38 times
tmp3 =
\e[1;35;40mzap/g1/00551138856342|120|rtS(10883)\e[0;37;40m\000accountcode:102190|UserID:3456|src:33225075|srcip:217.157.177.77|ConnectPrice:30|PeakPrice:60|RateID:55|CustomerID:30001|DestNameInt:Brazil_São...
#7 0x08078c74 in ast_pbx_run (c=0x43a45fa8) at pbx.c:1879
digit = 0 '\0'
exten = '\0' repeats 255 times
pos = 0
waittime = 1180700108
res = 0
#8 0x080804e1 in pbx_thread (data=0xfffc) at pbx.c:2102
No locals.
#9 0x40033f60 in pthread_start_thread () from /lib/i686/libpthread.so.0
No symbol table info available.
#10 0x40207327 in clone () from /lib/i686/libc.so.6
No symbol table info available.
Last from bt:
(gdb) bt
#0 0x4019f1f9 in free () from /lib/i686/libc.so.6
#1 0x080568bd in ast_frfree (fr=0x4025afd8) at frame.c:222
#2 0x0805fd54 in ast_channel_bridge (c0=0x43a45fa8, c1=0x8873018,
config=0x465fc73c, fo=0x465fbe8c, rc=0x465fbe90) at channel.c:2761
#3 0x4139c878 in ast_bridge_call (chan=0x43a45fa8, peer=0x8873018,
config=0x465fc73c) at res_features.c:342
#4 0x43032239 in dial_exec (chan=0x43a45fa8, data=0x465fc73c) at
app_dial.c:1003
#5 0x08076b11 in pbx_exec (c=0x43a45fa8, app=0x81b8e90,
data=0x465fec6c, newstack=148422744) at pbx.c:471
#6 0x0808009d in pbx_extension_helper (c=0x43a45fa8, context=0x43a46100
dialout, exten=0x465fec6c zap/g1/00551138856342|120|rtS(10883),
priority=200, callerid=0x81b8e90 Dial, action=1134845864) at
pbx.c:1384
#7 0x08078c74 in ast_pbx_run (c=0x43a45fa8) at pbx.c:1879
#8 0x080804e1 in pbx_thread (data=0xfffc) at pbx.c:2102
#9 0x40033f60 in pthread_start_thread () from /lib/i686/libpthread.so.0
#10 0x40207327 in clone () from /lib/i686/libc.so.6
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Re: [Asterisk-Users] grandstream sip phone calling Zap/1 on TDM20B rings and answers but not hear voice
Could you give us the output of the console when you try the call ? That would help us to point you in the right direction. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?
You really do not want to run Asterisk and X-Windows on the same box. That I understand... but this is not a production machine. Loading is not an issue. I'm using icewm. are there any other issues, besides loading, to not run x-windows at the same time? __ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?
Kenneth Long wrote: You really do not want to run Asterisk and X-Windows on the same box. That I understand... but this is not a production machine. Loading is not an issue. I'm using icewm. are there any other issues, besides loading, to not run x-windows at the same time? Yes. Whenever you scroll any windows you get clicks on any calls that are in progress. Even on FVWM. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power Alarm Error - Help
I have been getting the following message in Asterisk and it shuts Asterisk down, needing a reboot. Power alarm on Module 2 I have (1) TDM400P with (2) FXS (2) FXO cards (1) X100P card Any ideas? Since nobody answered, I'll guess something :) Did you plug the power on the TDM400P ? since you have FXS ports, you need to plug it in ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VPN server for * and woad warriors using windows?
Remco Barende wrote: What would be the best / easiest VPN software solution. I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. Ideally the vpn solution will not require any additional software on the client side but will use IPSEC. Remco, I've had very good success with OpenVPN http://sourceforge.net/projects/openvpn Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Best VPN server for * and woad warriors using windows?
Doug Lytle [EMAIL PROTECTED] writes: I've had very good success with OpenVPN http://sourceforge.net/projects/openvpn Me too, and I'd certainly use it in the original poster's stead. However, he specifically said that he must have an IPSEC tool, and OpenVPN is not IPSEC. -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream sip phone calling Zap/1 on TDM20Brings and answers but not hear voice
Here is the console screen. Starting simple switch on Zap/1-1 Executing Dial(Zap/1-1, SIP/403) in new stack Called 403 SIP/403-9c60 is ringing SIP/403-9c60 answered Zap/1 Spawn extension (smvoice-incoming, 403, 1) exited nonzero on Zap/1-1 Hangup Zap/1 I have a grandstream 101 that is calling an extension on Zap/1 of a TDM20B. The grandstream 101 can call another grandstream 101 at a different extension- that works fine. The two phones on TDM 20B can call each other.- no problem.When I call the TDM20B Zap/1 from the grandstream phone it rings - I answer and I dont hear any voice. for the grandstream I have tried allow=all for the codes but made no difference. Any ideas on what I am missing? Thanks, Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialogic D/4PCI
Steve Underwood wrote: Henry Devito wrote: Is there ant chance of the Dialogic card model D/4PCI working with asterisk ? Word of caution: Even if you can buy the drivers and make this card work with *, it is not meant to plug directly into a CO -48vdc talk battery and 90-130vac ring voltage delivered by your phone company. These cards were designed to be used behind a pbx. Most PBX's only deliver -24vdc on a analog line now and between 72v and 90vac for ring voltage. Not to say the D/4PCI or D/4PCIU will not work on a standard line and not have any problem, just a thought just incase you do get this to work and something blows up or gets ruined. You'll know why. Have a great day. Rubbish. That Dialogic card is designed and approved for use with PSTN lines. The only Dialogic cards I know of which are purely PBX oriented are their MSI cards, which only *provide* 24V and a lowish ring voltage. That said, the D/4PCI is useless with *, even if you buy the drivers. It is not a full duplex cards. The * drivers are for the JCT cards. Regards, Steve don't forget the older AMX cards. G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power Alarm Error - Help
Michael K. Rodriguez User wrote: I had a similar problem with power. I connected Asterisk to a Belkin UPS 1200VA and the the server would boot up and asterisk would load but the T1s on the Quad T1 card failed to come up. I placed a loop on the card and still no change. Finally, I removed the UPS and the T1s came up. Do know if this will help you, but the T1 card seems to be delicate with power. Hmmm strange. Makes you wonder if this is the problem that is occuring with the HP machines... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?
Kenneth Long wrote: You really do not want to run Asterisk and X-Windows on the same box. That I understand... but this is not a production machine. Loading is not an issue. I'm using icewm. are there any other issues, besides loading, to not run x-windows at the same time? Actually the issue seems to be more of a graphics drivers lock interrupts for long amounts of time, causing problems with Asterisk. Not specific to X. A simple VESA Frame Buffer can cause the same problems. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bandwidth, again, can someone check my math?
Bump -- anyone? -Original Message- From: Jay Milk [mailto:[EMAIL PROTECTED] Sent: Friday, January 21, 2005 11:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Bandwidth, again, can someone check my math? I want to put a single voice-mail box on a remote server, where I have metered bandwidth. Before I do this, I want to make sure it's feasible. Could someone confirm the following math for me? G.711, at 64kpbs has a rated network load of 88kbps. So for each second of conversation, about 11KB are crossing the wires in each direction. That means for a minute of two-way conversation, 1.3MB of data are transferred? That means for each GB of bandwith, callers can leave almost 800 minutes worth of voice-messages? Of course, this gets much better if we can get incoming calls on GSM, arriving at something like 2,500 minutes/GB. Is that correct, or did I mess up a decimal point somewhere? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Power Alarm Error - Help
Yes, The card is working fine most of the time. It just gets this message on occasion and then Asterisk shuts down. I debating putting surge suppressors on the PSTN lines. Could this be caused but a voltage issue from the Telco? Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 23, 2005 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Power Alarm Error - Help I have been getting the following message in Asterisk and it shuts Asterisk down, needing a reboot. Power alarm on Module 2 I have (1) TDM400P with (2) FXS (2) FXO cards (1) X100P card Any ideas? Since nobody answered, I'll guess something :) Did you plug the power on the TDM400P ? since you have FXS ports, you need to plug it in ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bandwidth, again, can someone check my math?
On January 21, 2005 12:26 pm, Jay Milk wrote: G.711, at 64kpbs has a rated network load of 88kbps. So for each second of conversation, about 11KB are crossing the wires in each direction. 88kbps = 88*1024 bps / 8 bits/byte =11kB/sec, yes, in each direction. That means for a minute of two-way conversation, 1.3MB of data are transferred? Yes, if you take the transmit and receive streams separate. 660kB in each direction. That means for each GB of bandwith, callers can leave almost 800 minutes worth of voice-messages? Seems right to me. 1024*1024 kBytes / 1320kB/min = 794.4 minutes Of course, this gets much better if we can get incoming calls on GSM, arriving at something like 2,500 minutes/GB. And even better if you can get VAD support into * so that it isn't sending back 660kB of silence per minute. Is that correct, or did I mess up a decimal point somewhere? Seems right. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?
On Jan 23, 2005, at 9:11 AM, Bruno Hertz wrote: On Sat, 2005-01-22 at 23:56 -0800, Kenneth Long wrote: seem like some kind of port issue... Actually, the fatal issue is that asterisk's chan_oss or chan_alsa grabs the sound device, so iaxclient can't do so. Probably. Both try to set up listeners on the IAX port (4569 for IAX2). Disable or reconfigure one of them to bind to a different port, whichever you want to answer on it. In the normal case, asterisk will start first, and get the port, then iaxclient will grab a transient port. This all works out OK, since the port that iaxclient uses doesn't matter unless you want to receive calls on it without registration. Also, don't forget to disable chan_alsa and chan_oss in modules.conf. When running another client you won't want the * console hogging your soundcard. Right, that's the thing that will make it work. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Florz patch for zaphfc
Has anyone had any success using the Florz patch for zaphfc ? I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN lines however the users are complaining of crackles on the line which I am assuming is related to the IRQ issues raised by Florz. I have tried to use the patch but it errors trying to patch zaphfc.h Any help would be appreciated. Regards, Stuart -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?
Steve Kann wrote: Actually, the fatal issue is that asterisk's chan_oss or chan_alsa grabs the sound device, so iaxclient can't do so. I can't run it anymore (I used to could. . . ) even on a machine that *isn't* running Asterisk. I haven't changed anything else on my machine, so I think it's somehow mis-aiming itself wrt the audio devices on the machine. At my site things degraded gradually: last summer things worked OK, but it only used hoggy codecs. Once ilbc was added it ran as long as I stayed a mile away from that codec. The latest version doesn't work at all. When I try to call I get lots of errors on the console: ortAudio error at opening separate output stream: Host error. PaHost_OpenStream: could not open /dev/dsp for O_RDONLY PaHost_OpenStream: ERROR - result = -1 PortAudio error at opening separate input stream: Host error. PaHost_OpenStream: could not open /dev/dsp for O_RDONLY PaHost_OpenStream: ERROR - result = -1 PortAudio error at opening separate input stream: Host error. PaHost_OpenStream: could not open /dev/dsp for O_RDONLY PaHost_OpenStream: ERROR - result = -1 PortAudio error at opening separate input stream: Host error. And the bottom of the interface says, Can't start audio. The IAX part appears to be working OK, though, and the little microphone VU meter bounces along like it sees *some* kind of audio, but nothing comes out the speakers. FWIW. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any experience with Sangoma cards?
Hi, I am considering A101/102/104 cards for my asterisk installations. Has any of you used these or any Sangoma cards in such environment? Any thoughts? How do they stack up against Digium cards? Any input would be greatly appreciated. robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power Alarm Error - Help
Martin Keding wrote: Yes, The card is working fine most of the time. It just gets this message on occasion and then Asterisk shuts down. I debating putting surge suppressors on the PSTN lines. Could this be caused but a voltage issue from the Telco? I was told the other day on IRC that telephone line surge protectors would only protect against *huge* voltages. Maybe there is such a thing as a power conditioner for telephone lines? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anybody a patch for oss/alsa to not constantly hog the sound card?
The subject says it all. After digging through latency and other issues with all kinds of linux softphones, I've found that only * works alright for me as a VoIP client. Problem now is that, unlike other apps, chan_oss resp. chan_alsa grab the card once and won't release it until shutdown, while other clients are friendly enough to grab the card only on calls. So, before getting lost in a regular coding frenzy, there isn't by chance any of you who already patched either of those chans to behave a little more cooperative? Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Autio cut off at beginning of call
I posted this question a while back, and I'm posting again in hopes that someone has some ideas. Sorry if you've already seen this. When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom, VoicePulse Connect) I often find that after the call is answered the first few seconds of audio are cut off (i.e. I don't hear the called party). This usually results in the called party saying hello Hello??? until I hear them. Has anyone else experienced this problem and found a cause or fix? My internal calls are perfect. It's just Internet-terminated calls that have the problem. Someone wrote in response to the last post saying that the audio path probably wasn't set up yet. I think this is the symptom, but I'm wondering what's the cause, and if there's a fix. Surely I'm not the only one who's having a huge problem with this. Can anyone help? Thank you, Reid Forrest, CISSP Max-IS, Inc. [EMAIL PROTECTED] ofc: 407.786.9600 x1200 cell: 321.439.8903 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy's apparantly failing in the field
Michael Giagnocavo wrote: Yes, the IAXy has faults, but until other IAX2 devices ship, it's the only game in town. I know that the Farfon device will be out soon and we'll be able to judge its quality at that time. Or any PA168 phones, which are already out, and support IAX2, SIP, H323, MGCP and N2P. (I've got one on my desk here as do a few others, and it works great.) I want one of them! Which model is it? Did you have to do any software upgrade? How much does it cost? Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call return?
For me this worked straight out of the box with [EMAIL PROTECTED] 0.3 Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Polk Sent: Sunday, 23 January 2005 4:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] call return? Hi: Can any one point me in the rite direction on this? I am using asterisk at home for learning purposes. I am trying to get the triditional *69 working. Has there been any success in getting it to announce the number and get it to give you the option to call back? Chris - Original Message - From: Diego Ventrice [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, January 22, 2005 8:03 AM Subject: Re: [Asterisk-Users] softswitch dilemma Thanks for answering Chad, Actually, I just want to Switch traffic between wholesale providers (my customers) which actually terminate traffic (or not, some of them have just controllers-softswitches like the one Im willing to set up) collect CDRs and bill them =) I have no gateways of my own (of any kind) so Im not originating nor terminating calls, just switching traffic is my goal, all this people use h.323 of course. Any advice would be appreciated. Thanks for your help D. Date: Fri, 21 Jan 2005 22:23:58 -0600 (CST) From: Chad Whitten [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] softswitch dilemma To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 are you looking to do actual pstn to voip termination? if so, then you are gonna need ss7, cama and imt trunks - things which asterisk doesnt necessarily support. now if you just want to buy pri/t1 from the local telco and sell voip services off an asterisk server that gets back to the pstn over these pri/t1's, then yes, asterisk can do this. Diego Ventrice said: Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that. An asterisk-ser or an asterisk-vocal combination may be the answer ? Thanks in advance for any help. Diego -- Chad Whitten ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP Providers and Backbone Servers
Hello All, Well, my explorations in to the world of VoIP is proving fruitful and in the near future I am hoping to have my small VoIP online service up and running ready to help promote the industry and hopefully gain a few customers in the process. Additionally, I will soon have my IAX and SIP softphone ready that will handle video, audio, and text communications. I am looking for quality and fair priced service providers so that I can add some of thier servies to my VoIP service which will start with an Asterisk PBX and some reliable Billing software (still trying to decide, but Trebas or ASTPP looks like it will work for me to get started.) In particular I will be providing Phone-Phone, Phone-PC, PC-PC, and PC-Phone connections. I am looking for services like 1. PSTN Termination Services (Good International Rates) 2. 800 Tollfree access line services 3. local, national, and international analog access line services in addition to my Asterisk PBX, if they exist (for my Phone-* services) and other services that you think are useful. Any suggestions or comments are appreciated and if you know of a quality service that I am looking for then now is the time and I invite your responses to this email. Thanks to everyone on this for giving me such great help, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Florz patch for zaphfc
On Sun, 23 Jan 2005, Stuart Hirst wrote: Has anyone had any success using the Florz patch for zaphfc ? I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN lines however the users are complaining of crackles on the line which I am assuming is related to the IRQ issues raised by Florz. I have tried to use the patch but it errors trying to patch zaphfc.h Any help would be appreciated. Im running bristuff-0.2.0-rc2b with Florians patch. 4 Billion hfc cards in ptp mode. Works like a charm. Even spandsp for receiving faxes works. Pelase describe your problem in more detail. /Nils Nils Segerdahl --- Upsala Systemkonsult, UPSYS AB Telefon:(+46) (0)18 56 80 41 Glunten, 751 83 UppsalaMobil: (+46) (0)703 55 65 03 http://www.upsys.seFax: (+46) (0)18 56 80 49 --- Jan 24 Eskimo Pie patented by Christian Nelson, 1922 Jan 24 Gold discovered in California at Sutter's Mill, 1848 Jan 24 DG Nova introduced, 1969 --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UPS for Asterisk
I'd considering an UPS backup system for my Asterisk server. I understand this is a linux issue, not a * issue, except for the following... Is the harddisk activity on a standard asterisk install such that I don't really have to worry if the power cuts?? As I understand, if HD activity is minimal, the probability of HD failure is significantly reduced. P.S. Power regulation is not needed, only protection against instantaneous power loss. Mike Sander -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in South Ontario?
MCI does not provide voice trunks T1/PRI by itself. They resell it as a add-ons to their IP solutions. Sprint is expensive. Bell is quite expensive as well. Allstream quite better in price. ISPTel is the least expensive one but their customer support is not one of the best. The best way to find rates for such lines to go to CRTC site and check the tariffs for that. All the Best! Sergey. Andrew Kohlsmith wrote: First things first -- don't reply to a message about something COMPLETELY different, erase everything and start your new message. Just click on the "To" and start your new message. When you reply and erase everything you are unintentionally placing your message in the middle of an existing message thread. This causes your message to get "buried" and far fewer people actually see it. You don't see this because you are using a mail client that has no concept of message threads. http://www.mixdown.ca/~andrew/dump/threaded_email.png is what a mailing list looks like to most people, and you can see why replying to a message, erasing its contents and starting an entirely new email about a different topic is frowned upon (yours is the highlighted message). Having said that, to your answer: On January 21, 2005 12:20 am, Robert Augustyn wrote: I am looking for a good provider of T1/PRI in Windsor, Ontario. You have many options in large cities. Bell, Group Telecom(360 networks), ATT(Allstream), Telus, Sprint, MCI(UUnet)... There may also be a dozen more "little guys" in your area. Get a few quotes, I find Bell is actually half-assed competitive when they have to be. Things to consider in your quotes received: - inbound or two-way call completion - Number of DIDs per DID/PRI order - # of #s received for incoming calls (4, 7, or 10 usually) - If they restrict the PRI signaling in any way - telephone number "fallback" if the PRI is down (i.e. where do the calls go) - 911/e911 - capability to set callerID/ANI to any DID you are leasing - ability to port existing numbers to the PRI as DIDs - charges for changing anything above once set up -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Florz patch for zaphfc
Nils Segerdahl wrote: Im running bristuff-0.2.0-rc2b with Florians patch. 4 Billion hfc cards in ptp mode. Works like a charm. 4 billion hfc cards! Wow that must be some server :) Oh a brand name - I guess I missed the capital letter. hehe -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS for Asterisk
Why risk it? Just go snag a cheap UPS from your local store. Just get something with enough run time to shut the system down gracefully. On Mon, 24 Jan 2005 08:04:36 +1100, Mike Sander [EMAIL PROTECTED] wrote: I'd considering an UPS backup system for my Asterisk server. I understand this is a linux issue, not a * issue, except for the following... Is the harddisk activity on a standard asterisk install such that I don't really have to worry if the power cuts?? As I understand, if HD activity is minimal, the probability of HD failure is significantly reduced. P.S. Power regulation is not needed, only protection against instantaneous power loss. Mike Sander -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some issues with X-Lite and codecs.
Yes I did. The same. It looks like there is some packet loss on the way to my VoIP box. Is there any optimal settings for jitter buffer for * ? All the Best! Sergey. Andrew Yager wrote: Hi Sergey, Have you tried phoning from X-Lite to your PSTN line, or your PSTN line to X-Lite? How is the audio quality then? Does it vary depending on the codec you have used? Andrew On 23/01/2005, at 4:31 PM, Sergey Kuznetsov wrote: Hi there, I am experiencing some issue with X-Lite. When I am calling over the phone thru my PSTN-to-VoIP gateway internationally using G.729 the quality is just perfect. When I am using X-Lite to connect the same box, and then to call internationally - I am experiencing some issues. I have 5Mbit/800Kbit cable link with average 60 msecs to my VoIP box. The transfer rate is never falling below 500Kbytes/sec. Therefore I am not suspecting quite noticeable packet loss. I enabled G.711 ulaw, alaw and speex codecs on both sides. By playing with different codecs I am trying to avoid some clicking and sound distortion, which is I am experiencing right now. Speex sometimes is better than G.711, but still having the same glitching. My question is, is there any way to fix it by playing with some parameters on * side, or it's better to play with X-Lite parameters? All the Best! Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS for Asterisk
On January 23, 2005 04:04 pm, Mike Sander wrote: Is the harddisk activity on a standard asterisk install such that I don't really have to worry if the power cuts?? Not typically; there isn't much writing going on, this is true. Are you that cash strapped that a $75 UPS with a serial port is out of your budget? As I understand, if HD activity is minimal, the probability of HD failure is significantly reduced. HDDs don't fail because they lose power. You get data corruption when writing and losing power, and you get filesystem corruption if the filesystem/OS is postponing writes to increase write performance. That's not HDD failure. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] simulating multiple lines using ADSI
Does anyone have any experience with making an adsi phone appear to have more than one line. It seems like this would be a very simple and very useful thing to be able to do. Ideally, it would be nice if you could make the 6 soft buttons appear as lines 1-6 and if you press one of the soft buttons, it puts the current line on hold and gives you a new dialtone. Has anyone either done something like this or would happen to know how to do something like this? Thanks, Jon. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario?
Sergey, Thanks for the input. I looked at the crtc site did few searches but I guess I do not know what to look for because I did not find anything related to tariffs. On the same note I am not able to find a Isptel web site either I guess it is not my day today :) robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergey KuznetsovSent: Sunday, January 23, 2005 4:15 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario? MCI does not provide voice trunks T1/PRI by itself. They resell it as a add-ons to their IP solutions.Sprint is expensive. Bell is quite expensive as well. Allstream quite better in price. ISPTel is the least expensiveone but their customer support is not one of the best.The best way to find rates for such lines to go to CRTC site and check the tariffs for that.All the Best!Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autio cut off at beginning of call
On January 23, 2005 03:42 pm, Reid Forrest wrote: When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom, VoicePulse Connect) I often find that after the call is answered the first few seconds of audio are cut off (i.e. I don't hear the called party). This usually results in the called party saying hello Hello??? until I hear them. This is not normal; I do *not* have this issue with NuFone and I have placed a ton of calls through them daily for the past year. I don't recall having this problem with voicepulse connect when I used them, nor do I have the issue with iax.cc for inbound calls. Has anyone else experienced this problem and found a cause or fix? My internal calls are perfect. It's just Internet-terminated calls that have the problem. Someone wrote in response to the last post saying that the audio path probably wasn't set up yet. I think this is the symptom, but I'm wondering what's the cause, and if there's a fix. It very much sounds like it's something on your end... How about some specifics? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario?
Sorry, I completely forgot. You have to have an experience how to use the CRTC site =) If you will click to "Public Proceedings" at the top of the main page you will be redirected to the page witch will show you the most of the useful information. At that page in the "Telecommunications" Part of the table you will see link "Tariff" with is going to this page: http://www.crtc.gc.ca/8740/eng/tariff.htm At that pages you have to choose year and then the name of the company you are interesting about. There is the some info buried there, but it's quite easy to find it. I cannot find the website of ISPtel either. But I have the PRIs from them and it's 2 times cheaper then PRIs from Sprint. http://www.crtc.gc.ca/8740/frn/2002/a4.htm - Allstream (ATT) rates. Probably there is some new rates. Have to go thru all recent years. All the Best! Sergey. Robert Augustyn wrote: Sergey, Thanks for the input. I looked at the crtc site did few searches but I guess I do not know what to look for because I did not find anything related to tariffs. On the same note I am not able to find a Isptel web site either I guess it is not my day today :) robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey Kuznetsov Sent: Sunday, January 23, 2005 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario? MCI does not provide voice trunks T1/PRI by itself. They resell it as a add-ons to their IP solutions. Sprint is expensive. Bell is quite expensive as well. Allstream quite better in price. ISPTel is the least expensive one but their customer support is not one of the best. The best way to find rates for such lines to go to CRTC site and check the tariffs for that. All the Best! Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Timing Slips
Ken Godee wrote: Does anyone know how to monitor * to see if they are receiving timing slips on a span connected to a T100P card? I am seeing b-channel restarts quite often and also getting No D-channels available warnings from time to time. Yesterday I had all the b-channels crash during a MeetMe Conference. Not good! This PRI is connected to an Avaya Definity PBX that is onsite and located in the same room as *. * is set to clock off the Definity. I am seeing no problems on the PRI from the Definity side. Just thought I'd run this by you. We've been running connected to our Definity G3si R6 via TN767 -- TE410P and have had no problems. I guess I'll eat crow alittle bit I guess your email made me focus a little more into it. I'm also having the same problems as you are, D-Channel bouncing. D-channel down and right back up and then b-channels restart, while restarting they DO drop any active channels. :( Experenced first hand on friday while remote monitoring and on a call. D-channel down No D-channel found, using channel 48 anyway. D-channel up restarting channel etc. As another poster suggested, I tried changing timing to internal clocking, vs. Definty, no help thou. I've done a ton of searching and have not found much more I can try. What protocol are you using on the Definity side? As I understand it a = ni1 / b = national If you come across anything that helps, please let me know. I'll also let you know if I find anything. I also see no problems on the Definity side. No errors when loop up circuit either. ztmonitor runs 100%-99%. No missing interrupts, etc. Load/no load doesn't seem to make a difference. Running astersk v1.0.3 ken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autio cut off at beginning of call
My wife brought to my attention just yesterday that this is happening on all my inbound PSTN calls. I am using a ZAP interface, not IAX. Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music On-Hold problem
My problem is: No matter what machine I install and configure Asterisk on it I just can't get the music on-hold to work. Is anyone of you out there have such problem? If so what have you done to fix the problem? I've tried so far three other computers and none of them I was able to get music on-hold to work. CAN PLEASE ANYONE HELP. attachment: winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-2000
600 is for the US only. FXS impedence for UK 370+620||310nF Europe CTR21 270+750||150nF Chris - Original Message - From: Remco Barende [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 22, 2005 10:12 AM Subject: Re: [Asterisk-Users] SPA-2000 On Sat, 22 Jan 2005, Duane wrote: Remco Barende wrote: On Fri, 21 Jan 2005, Henry Devito wrote: Hi, I have not implemented any of the spa-2000's yet. Do they work ok with asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is it two fxs ports with the same extension? They work pretty well, but I'm not impressed with the sound quality. Sound is quite soft and I have to adjust the input and output gains to something like +3 or +5 for in+out and then an annoying hiss is audible. I have a sipura 2000 and haven't had to alter gain at all, and no hiss, then again are you using ulaw or using g729? I use G711u. When I do not adjust the output gain the volume in+out is just too soft. Or would I need to change another setting? Under Regional I can also set port impedance. No idea though if the default value of 600 is ok for Europe? As for the original question, the 2 ports on the 2000 and the 3000 are both seperate SIP identities and you have to configure them as 2 seperate lines... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Notify and PHP AGI
Hello * Users. I need to be able to generate a Sip Notify message using PHP AGI but have no idea how I can do that. What I need to send is the balance of the prepaid card and display it on the soft phones display. Does anyone know how to do this? Thanks in advance. KF ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival
Is it possible to get the Festival command to read the text from a system file rather than having it input as a text string? I suppose I could put the text string into an Asterisk variable and reference that in the Festival command, but then, how do I get the contents of the file into the Asterisk variable? Is this a case of having to use AGI, or is there a simpler way? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Best VPN server for * and woad warriors using windows?
Tom Ivar Helbekkmo wrote: Me too, and I'd certainly use it in the original poster's stead. However, he specifically said that he must have an IPSEC tool, and OpenVPN is not IPSEC. -tih We are currently using OpenVPN too with good success. I'm not sure why you would require IPSEC. I thought that is what we wanted when we looked at VPN solutions, but OpenVPN is so easy compared the the IPSEC stuff out there. Calvin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can anyone recoment T1/PRI providerin SouthOntario?
Thanks You sure have to have experience ...:) Do you know how I can contact ISPtel? Sprint quoted me a realy high number. btw: what do you get with your PRI service? robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergey KuznetsovSent: Sunday, January 23, 2005 5:54 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Can anyone recoment T1/PRI providerin SouthOntario? Sorry, I completely forgot. You have to have an experience how to use the CRTC site =)If you will click to "Public Proceedings" at the top of the main page you will be redirected tothe page witch will show you the most of the useful information.At that page in the "Telecommunications" Part of the table you will see link "Tariff" with isgoing to this page: http://www.crtc.gc.ca/8740/eng/tariff.htmAt that pages you have to choose year and then the name of the company you are interesting about.There is the some info buried there, but it's quite easy to find it.I cannot find the website of ISPtel either. But I have the PRIs from them and it's 2 times cheaper then PRIs fromSprint.http://www.crtc.gc.ca/8740/frn/2002/a4.htm - Allstream (ATT) rates.Probably there is some new rates. Have to go thru all recent years.All the Best!Sergey.Robert Augustyn wrote: Sergey, Thanks for the input. I looked at the crtc site did few searches but I guess I do not know what to look for because I did not find anything related to tariffs. On the same note I am not able to find a Isptel web site either I guess it is not my day today :) robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey KuznetsovSent: Sunday, January 23, 2005 4:15 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario?MCI does not provide voice trunks T1/PRI by itself. They resell it as a add-ons to their IP solutions.Sprint is expensive. Bell is quite expensive as well. Allstream quite better in price. ISPTel is the least expensiveone but their customer support is not one of the best.The best way to find rates for such lines to go to CRTC site and check the tariffs for that.All the Best!Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On-Hold problem
What is the CLI output you are getting? Do you have a timer source installed? On Sun, 23 Jan 2005 18:10:27 -0500, Computer Onsite Support [EMAIL PROTECTED] wrote: My problem is: No matter what machine I install and configure Asterisk on it I just can't get the music on-hold to work. Is anyone of you out there have such problem? If so what have you done to fix the problem? I've tried so far three other computers and none of them I was able to get music on-hold to work. CAN PLEASE ANYONE HELP. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP Providers and Backbone Servers
I don't want to be a kill-joy, but after reading your various messages over the last few days, I think you're in over your head on this one. I suggest you first get your own * system up and running. Then, re-examine your goals. So far, you don't seem to be adding anything new to the VOIP community, so I'm at a loss at how you expect to make money. If you're reselling services, then you can't compete with the current players in the market, as obviously their prices would be better than yours. If you're trying to bundle resold services, the big ones (like Vonage) have an insurmountable advantage in infrastructure, volume and installed base. And lastly, if you expect to sell service along with your own softphone client (which you still have to complete), there are free solutions with established services out there (iconnecthere, etc). So, besides (weak) competition, what are you going to add? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, January 23, 2005 2:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoIP Providers and Backbone Servers Hello All, Well, my explorations in to the world of VoIP is proving fruitful and in the near future I am hoping to have my small VoIP online service up and running ready to help promote the industry and hopefully gain a few customers in the process. Additionally, I will soon have my IAX and SIP softphone ready that will handle video, audio, and text communications. I am looking for quality and fair priced service providers so that I can add some of thier servies to my VoIP service which will start with an Asterisk PBX and some reliable Billing software (still trying to decide, but Trebas or ASTPP looks like it will work for me to get started.) In particular I will be providing Phone-Phone, Phone-PC, PC-PC, and PC-Phone connections. I am looking for services like 1. PSTN Termination Services (Good International Rates) 2. 800 Tollfree access line services 3. local, national, and international analog access line services in addition to my Asterisk PBX, if they exist (for my Phone-* services) and other services that you think are useful. Any suggestions or comments are appreciated and if you know of a quality service that I am looking for then now is the time and I invite your responses to this email. Thanks to everyone on this for giving me such great help, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Definity PBX with a T100P TN767E
Ken Godee wrote: I'm currently playing with a Digium T100P card and 2 Grandstream phones, things are working well. I wanted to move on to linking our Definity G3R Rev 8.2 to the T100P. Everything that I've read so far shows that you need a TN464 to accomplish this. We have a TN767E available. Inbound/outbound, DID from G3 inbound, ext./ext., etc. Ken, Hope your holiday went well. We have been able to get our Definity G3R working with Asterisk via a T100P card and a TN767E card, works very well! But, I'm a little stuck on how to get the DID info from the G3 and ext/ext info to the G3. Incoming shows the trunk info setup by our phone admin. Happen to have a link that you could point me to on this setup? Thanks again! Doug Lytle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy's apparantly failing in the field
I have one of these phones. I bought it off of eBay. Not sure where to get them direct. You will need to load the proper image, in that I believe it ships with SIP by default. Each protocol has its own image. Erik On Mon, 24 Jan 2005 00:54:48 +0400, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Michael Giagnocavo wrote: Yes, the IAXy has faults, but until other IAX2 devices ship, it's the only game in town. I know that the Farfon device will be out soon and we'll be able to judge its quality at that time. Or any PA168 phones, which are already out, and support IAX2, SIP, H323, MGCP and N2P. (I've got one on my desk here as do a few others, and it works great.) I want one of them! Which model is it? Did you have to do any software upgrade? How much does it cost? Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Data calls with Asterisk
Hi All: I am new to Asterisk so if my question sounds too newbeeish then pleasebear with me. I have about 10 remote locations which are collecting some data. Iwould like to upload that data every night. All remote locations have56K modem. I was wondering can Asterisk be used to receive this data? Basically I will have an asterisk with 1 FXO card and have it receivedata calls. Can asterisk receive data calls? Thanks in advance for your responsesRegards,Karim MardhaniZeeCore Consulting ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP Providers and Backbone Servers
I really appreciate your comments regarding the challenges that face a new VoIP service. While it is true that there is always still much to learn in both the VoIP arena and with a true business model that will try to bring something new and exciting to the existing community, I still strongly contend that as this is still a young industry that building conglomerations of existing VoIP services along with strategic partnering will foster industry innovation and lay a strong foundation for the VoIP industry to eventually overtake the analog corporate strong-holds that currently exist. Additionally, these small beginings enable people like myself to learn the industry quickly and get involved. It also allow us to learn about the Astrisk PBX system as well as the multitude of hardware and software that comprise this exciting field. In as much a funding is concerned, we intend to come online just as in the manner as other VoIP services and to move quickly to branch out by partnering to cover a broader scoping topology. Could be challenging at first, but your response implies that an Internet store could not come online just because there are other currently existing online stores in that market. It's a long hard road, but I do believe that in order to become big you must plan from the onset to be big and think in those terms. Everyday should be the question in your mind of How can I expand the business today?. Always this and always moving forward; never back. With regards to the softphone, once finished, it would be given away for free and probably open source as well. We are here to promote the industry, make it grow, and build a solid foundation for the future of telecommunications. At least that is what I see in this. Hope that it is the same for you as well. Just my personal input to the email that you sent to the list and I am sure that there will be some disagreement with its content and to the philosophy indicated but this is an exciting time for getting started and to embark upon the multitude of innovation that awaits us all. I truly thank all of the members of the list for giving me an opportunity to learn from you and to, hopefully, eventually be able to give back to the list and help others in the near future. There is so much that the Astrisk PBX and supporting hardware can do to open doors on the virtal highway and I am happy to begin travelling down that road. Thanks again for your response and have a great day, Lonnie Cumberland I don't want to be a kill-joy, but after reading your various messages over the last few days, I think you're in over your head on this one. I suggest you first get your own * system up and running. Then, re-examine your goals. So far, you don't seem to be adding anything new to the VOIP community, so I'm at a loss at how you expect to make money. If you're reselling services, then you can't compete with the current players in the market, as obviously their prices would be better than yours. If you're trying to bundle resold services, the big ones (like Vonage) have an insurmountable advantage in infrastructure, volume and installed base. And lastly, if you expect to sell service along with your own softphone client (which you still have to complete), there are free solutions with established services out there (iconnecthere, etc). So, besides (weak) competition, what are you going to add? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, January 23, 2005 2:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoIP Providers and Backbone Servers Hello All, Well, my explorations in to the world of VoIP is proving fruitful and in the near future I am hoping to have my small VoIP online service up and running ready to help promote the industry and hopefully gain a few customers in the process. Additionally, I will soon have my IAX and SIP softphone ready that will handle video, audio, and text communications. I am looking for quality and fair priced service providers so that I can add some of thier servies to my VoIP service which will start with an Asterisk PBX and some reliable Billing software (still trying to decide, but Trebas or ASTPP looks like it will work for me to get started.) In particular I will be providing Phone-Phone, Phone-PC, PC-PC, and PC-Phone connections. I am looking for services like 1. PSTN Termination Services (Good International Rates) 2. 800 Tollfree access line services 3. local, national, and international analog access line services in addition to my Asterisk PBX, if they exist (for my Phone-* services) and other services that you think are useful. Any suggestions or comments are appreciated and if you know of a quality service that I am looking for then now is the time and I invite your responses to this email. Thanks to everyone on this for giving me such great help, Lonnie
[Asterisk-Users] No music with Blind transfer on GS ATA + Sipura-841
Hi there, I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream ATA's. The problem is that with both of these devices the Unattended call transfer process seems to be just like Attended but instead you hang up as soon as you have dialled the number of the party your are transferring to. The call transfer all works fine BUT as you complete your side of the transfer and the destination extension is ringing - ie. Caller calls you, you transfer call to another extension and hangup before they answer and another extension is left ringing waiting for someone to pickup, the inital caller has only silence and no MOH (or ringing). I cannot tell wether this is a function of Asterisk (which seems to end MOH during bridging of the calls) or the SPA-841 / GS's but the problem is that doing this type of unattended transfer results in what the initial caller hears as a dead line and they are prone to hangup should the called party not answer quickly. I guess we could just do all transfers as attended and wait for the called party to answer so as to avoid this assumed dead line problem but I wonder if anyone has also experienced this and or found a solution. My Asterisk log is as follows: asterisk*CLI -- Executing Macro(SIP/201-6447, oneline|Sip/200) in new stack -- Executing SetMusicOnHold(SIP/201-6447, random) in new stack -- Executing Dial(SIP/201-6447, Sip/200|30|tr) in new stack -- Called 200 -- SIP/200-08b4 is ringing -- SIP/200-08b4 answered SIP/201-6447 -- Attempting native bridge of SIP/201-6447 and SIP/200-08b4 -- Started music on hold, class 'random', on SIP/201-6447 -- Executing Macro(SIP/200-eedc, oneline|Sip/202) in new stack -- Executing SetMusicOnHold(SIP/200-eedc, random) in new stack -- Executing Dial(SIP/200-eedc, Sip/202|30|tr) in new stack -- Called 202 -- SIP/202-f8cf is ringing -- Stopped music on hold on SIP/201-6447 == Spawn extension (macro-oneline, s, 2) exited non-zero on 'SIP/200-eedcZOMBIE' in macro 'oneline' == Spawn extension (from-internal, 200, 1) exited non-zero on 'SIP/200-eedcZOMBIE' -- SIP/202-f8cf answered SIP/201-6447 -- Attempting native bridge of SIP/201-6447 and SIP/202-f8cf Thanks, Justin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anyone recoment T1/PRI providerin SouthOntario?
I got my PRIs from ISPtel as an add-on to my colo with MCI and thru MCI. I'll try to find ISPtel web-site (if it's exists) thru MCI's customer service. Actually Allstream's PRI will cost you around 700-750 CAD per month. It's not that bad. I got just few PRIs with set of DIDs I need. This is enough for me. I can set any ANI/C*ID form my range on my PRIs. My incoming DNIS is 10-digit length. I didn't try if I can port existing DIDs from another ILECs/CLECs. All the Best! Sergey. Robert Augustyn wrote: Thanks You sure have to have experience ...:) Do you know how I can contact ISPtel? Sprint quoted me a realy high number. btw: what do you get with your PRI service? robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey Kuznetsov Sent: Sunday, January 23, 2005 5:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI providerin SouthOntario? Sorry, I completely forgot. You have to have an experience how to use the CRTC site =) If you will click to "Public Proceedings" at the top of the main page you will be redirected to the page witch will show you the most of the useful information. At that page in the "Telecommunications" Part of the table you will see link "Tariff" with is going to this page: http://www.crtc.gc.ca/8740/eng/tariff.htm At that pages you have to choose year and then the name of the company you are interesting about. There is the some info buried there, but it's quite easy to find it. I cannot find the website of ISPtel either. But I have the PRIs from them and it's 2 times cheaper then PRIs from Sprint. http://www.crtc.gc.ca/8740/frn/2002/a4.htm - Allstream (ATT) rates. Probably there is some new rates. Have to go thru all recent years. All the Best! Sergey. Robert Augustyn wrote: Sergey, Thanks for the input. I looked at the crtc site did few searches but I guess I do not know what to look for because I did not find anything related to tariffs. On the same note I am not able to find a Isptel web site either I guess it is not my day today :) robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey Kuznetsov Sent: Sunday, January 23, 2005 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario? MCI does not provide voice trunks T1/PRI by itself. They resell it as a add-ons to their IP solutions. Sprint is expensive. Bell is quite expensive as well. Allstream quite better in price. ISPTel is the least expensive one but their customer support is not one of the best. The best way to find rates for such lines to go to CRTC site and check the tariffs for that. All the Best! Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Upgrade to the newest cvs now asterisk will not start
Hello group I just update to the newest CVS now I'm not able to get asterisk to start. No error during the make or make install I did a make clean before the make;make install Any help would be great Here is the output asterisk -vgcd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf == Binding realtime_ext to mysql/realtime/extensions_table == Binding voicemail to mysql/realtime/voicemail_users == Binding sipfriends to mysql/realtime/sip_buddies Asterisk CVS-HEAD-01/23/05-19:38:48, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == Manager registered action ListCommands Asterisk Management interface listening on port 5038 == RTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [ImportVar] == Registered application 'ImportVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: [chan_modem.so] = (Generic Voice Modem Driver) [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' Junk at the beginning 49443302 Warning, flexibel rate not heavily tested! Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! [res_adsi.so] = (ADSI Resource) [res_features.so] = (Call Parking Resource) == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_crypto.so] = (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup' [res_indications.so] = (Indications Configuration) -- Registered indication country 'cl' -- Registered indication country 'tw' -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Registered indication country 'fi' -- Registered indication country 'no' -- Registered indication country 'br' -- Registered indication country 'za' -- Registered indication country 'it' -- Registered indication country 'us-o' -- Registered indication country 'gr' -- Registered indication country 'ru' -- Registered indication country 'nz' -- Setting default indication country to 'us' == Registered application 'Playtones' == Registered application 'StopPlaytones' [res_monitor.so] = (Call Monitoring Resource) == Registered application 'Monitor' == Registered application 'StopMonitor' == Registered application 'ChangeMonitor' == Manager registered action Monitor == Manager registered action StopMonitor == Manager registered
Re: [Asterisk-Users] IAXy's apparantly failing in the field
Try here for the iax2 phone http://www.ngtel.de/products.php#1 - Original Message - From: Erik Espinoza [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 23, 2005 6:40 PM Subject: Re: [Asterisk-Users] IAXy's apparantly failing in the field I have one of these phones. I bought it off of eBay. Not sure where to get them direct. You will need to load the proper image, in that I believe it ships with SIP by default. Each protocol has its own image. Erik On Mon, 24 Jan 2005 00:54:48 +0400, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Michael Giagnocavo wrote: Yes, the IAXy has faults, but until other IAX2 devices ship, it's the only game in town. I know that the Farfon device will be out soon and we'll be able to judge its quality at that time. Or any PA168 phones, which are already out, and support IAX2, SIP, H323, MGCP and N2P. (I've got one on my desk here as do a few others, and it works great.) I want one of them! Which model is it? Did you have to do any software upgrade? How much does it cost? Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Providers and Backbone Servers
Monday, January 24, 2005, 12:20:46 AM, Jay Milk wrote: I don't want to be a kill-joy, but after reading your various messages over the last few days, I think you're in over your head on this one. I suggest you first get your own * system up and running. Then, re-examine your goals. So far, you don't seem to be adding anything new to the VOIP community, so I'm at a loss at how you expect to make money. If you're reselling services, then you can't compete with the current players in the market, as obviously their prices would be better than yours. If you're trying to bundle resold services, the big ones (like Vonage) have an insurmountable advantage in infrastructure, volume and installed base. And lastly, if you expect to sell service along with your own softphone client (which you still have to complete), there are free solutions with established services out there (iconnecthere, etc). So, besides (weak) competition, what are you going to add? Competition is what drives prices down, VoIP is a new challange to traditional telecoms many are starting to wakeup to the fact that soon you will no longer have an area code, country code, but rather a global number that is yours and will follow you where you go globally, a much bigger version of the GSM roaming we had in europe for years without all the extra silly costs. once you have IP you have a phone, you can collect messages, phone people and do your work on the move. With the explosion of wireless hotspots this makes a VoIP on the move a reality, some telecos are now realising that if they don't join the 'revolution' they'll be out of profitable call business and will endup installing circuits for other voip providers to use. BT for example here in the UK are planning to convert their entire network to VoIP based network. IP to everyhome and that sort of thing because they want a slice of the action. My guess is telecos will move away from charging for calls and start charging for VoIP traffic but it is a rapid development and my guess is some telecos will be caught out and probably be out of business in the next 5-10 years. Competition however small encourages companies to improve their services, compete on pricing and look for ways to attract new customers, many companies I know now use skype to contact staff working from home, because it is convenient and easy to use. Like I said, compeition isn't a bad thing, some people work on * for the sake of technology development and some work on it in the hope of making a living out of it. Just my £0.02 :) -- Best regards, Subhi S Hashwamailto:[EMAIL PROTECTED] When everything is heading your way, you're in the wrong lane. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy's apparantly failing in the field
BTW they also an iax2 ATA Try here for the iax2 phone http://www.ngtel.de/products.php#1 - Original Message - From: Erik Espinoza [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 23, 2005 6:40 PM Subject: Re: [Asterisk-Users] IAXy's apparantly failing in the field I have one of these phones. I bought it off of eBay. Not sure where to get them direct. You will need to load the proper image, in that I believe it ships with SIP by default. Each protocol has its own image. Erik On Mon, 24 Jan 2005 00:54:48 +0400, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Michael Giagnocavo wrote: Yes, the IAXy has faults, but until other IAX2 devices ship, it's the only game in town. I know that the Farfon device will be out soon and we'll be able to judge its quality at that time. Or any PA168 phones, which are already out, and support IAX2, SIP, H323, MGCP and N2P. (I've got one on my desk here as do a few others, and it works great.) I want one of them! Which model is it? Did you have to do any software upgrade? How much does it cost? Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP software for MAC OS older than X?
Hello, is there anybody reading this who has experience with VoIP (IAX or not) on Macintosh computers? If so, have you ever seen or heard of (even an experimental, i.e., never marketed) VoIP application for any of the older Mac OSs, such as 9, 8, or 7? I can't quite believe that VoIP is such a recent idea that it was invented only *after* Mac OS X had become firmly established, but so far my searches have turned out nothing. However, not all good stuff and good ideas are on the web,so a community of knowledgable people often has information that a web search cannot produce. Appreciate any leads and comments... Thanks: H.D. -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy's apparantly failing in the field
Henry Devito wrote: BTW they also an iax2 ATA Try here for the iax2 phone http://www.ngtel.de/products.php#1 Do you have a contact email for these guys? I couldn't see anything listed on their site anywhere. Seems the site is in current development. Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can anyone recomentT1/PRI providerin SouthOntario?
Thanks for your help Sergey. robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergey KuznetsovSent: Sunday, January 23, 2005 8:00 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Can anyone recomentT1/PRI providerin SouthOntario? I got my PRIs from ISPtel as an add-on to my colo with MCI and thru MCI. I'll try to find ISPtel web-site (if it's exists) thruMCI's customer service. Actually Allstream's PRI will cost you around 700-750 CAD per month. It's not that bad.I got just few PRIs with set of DIDs I need. This is enough for me. I can set any ANI/C*ID form my range on my PRIs.My incoming DNIS is 10-digit length.I didn't try if I can port existing DIDs from another ILECs/CLECs.All the Best!Sergey.Robert Augustyn wrote: Thanks You sure have to have experience ...:) Do you know how I can contact ISPtel? Sprint quoted me a realy high number. btw: what do you get with your PRI service? robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey KuznetsovSent: Sunday, January 23, 2005 5:54 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Can anyone recoment T1/PRI providerin SouthOntario?Sorry, I completely forgot. You have to have an experience how to use the CRTC site =)If you will click to "Public Proceedings" at the top of the main page you will be redirected tothe page witch will show you the most of the useful information.At that page in the "Telecommunications" Part of the table you will see link "Tariff" with isgoing to this page: http://www.crtc.gc.ca/8740/eng/tariff.htmAt that pages you have to choose year and then the name of the company you are interesting about.There is the some info buried there, but it's quite easy to find it.I cannot find the website of ISPtel either. But I have the PRIs from them and it's 2 times cheaper then PRIs fromSprint.http://www.crtc.gc.ca/8740/frn/2002/a4.htm - Allstream (ATT) rates.Probably there is some new rates. Have to go thru all recent years.All the Best!Sergey.Robert Augustyn wrote: Sergey, Thanks for the input. I looked at the crtc site did few searches but I guess I do not know what to look for because I did not find anything related to tariffs. On the same note I am not able to find a Isptel web site either I guess it is not my day today :) robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey KuznetsovSent: Sunday, January 23, 2005 4:15 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario?MCI does not provide voice trunks T1/PRI by itself. They resell it as a add-ons to their IP solutions.Sprint is expensive. Bell is quite expensive as well. Allstream quite better in price. ISPTel is the least expensiveone but their customer support is not one of the best.The best way to find rates for such lines to go to CRTC site and check the tariffs for that.All the Best!Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival
Howard Lowndes wrote: Is it possible to get the Festival command to read the text from a system file rather than having it input as a text string? Is this a case of having to use AGI, or is there a simpler way? Most people would use AGI for that (combined with the text2wave or whatever program). In fact there may even be an example on the wiki. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP USB Phone?
There are a number of Skype USB phones available. Are there any when connected to a Windows PC can access Asterisk? Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anyone recomentT1/PRI providerin SouthOntario?
You are very welcome! All the Best! Sergey. Robert Augustyn wrote: Thanks for your help Sergey. robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey Kuznetsov Sent: Sunday, January 23, 2005 8:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can anyone recomentT1/PRI providerin SouthOntario? I got my PRIs from ISPtel as an add-on to my colo with MCI and thru MCI. I'll try to find ISPtel web-site (if it's exists) thru MCI's customer service. Actually Allstream's PRI will cost you around 700-750 CAD per month. It's not that bad. I got just few PRIs with set of DIDs I need. This is enough for me. I can set any ANI/C*ID form my range on my PRIs. My incoming DNIS is 10-digit length. I didn't try if I can port existing DIDs from another ILECs/CLECs. All the Best! Sergey. Robert Augustyn wrote: Thanks You sure have to have experience ...:) Do you know how I can contact ISPtel? Sprint quoted me a realy high number. btw: what do you get with your PRI service? robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey Kuznetsov Sent: Sunday, January 23, 2005 5:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI providerin SouthOntario? Sorry, I completely forgot. You have to have an experience how to use the CRTC site =) If you will click to "Public Proceedings" at the top of the main page you will be redirected to the page witch will show you the most of the useful information. At that page in the "Telecommunications" Part of the table you will see link "Tariff" with is going to this page: http://www.crtc.gc.ca/8740/eng/tariff.htm At that pages you have to choose year and then the name of the company you are interesting about. There is the some info buried there, but it's quite easy to find it. I cannot find the website of ISPtel either. But I have the PRIs from them and it's 2 times cheaper then PRIs from Sprint. http://www.crtc.gc.ca/8740/frn/2002/a4.htm - Allstream (ATT) rates. Probably there is some new rates. Have to go thru all recent years. All the Best! Sergey. Robert Augustyn wrote: Sergey, Thanks for the input. I looked at the crtc site did few searches but I guess I do not know what to look for because I did not find anything related to tariffs. On the same note I am not able to find a Isptel web site either I guess it is not my day today :) robert From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sergey Kuznetsov Sent: Sunday, January 23, 2005 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario? MCI does not provide voice trunks T1/PRI by itself. They resell it as a add-ons to their IP solutions. Sprint is expensive. Bell is quite expensive as well. Allstream quite better in price. ISPTel is the least expensive one but their customer support is not one of the best. The best way to find rates for such lines to go to CRTC site and check the tariffs for that. All the Best! Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP USB Phone?
Adi, Yes there are... You can probably use the exact same skype USB phone with X-Lite or one of the many other windows SIP softphones. It is not a matter of being compatable with Asterisk so much as being compatable with your Asterisk softphone.. In the X-Lite menu, system settings - USB Settings. If you can spare the dollars, a hardware phone is almost always better though.. Cheers Shane Adi Linden Sent: Monday, 24 January 2005 1:16 PM There are a number of Skype USB phones available. Are there any when connected to a Windows PC can access Asterisk? Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP USB Phone?
There are heaps but why not use a headset If you insist on usb handset then there are 3 listed here. http://www.telecoms.co.uk/catalog/default.php?cPath=583_829_830 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adi Linden Sent: Sunday, January 23, 2005 9:16 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP USB Phone? There are a number of Skype USB phones available. Are there any when connected to a Windows PC can access Asterisk? Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP USB Phone?
Hi,Adi, We provide the USB phone you wanted, it can access Asterisk natively. It can support Skype,X-Lite,X-PRO,eyeBeam,StanaPhone,SJphone,Net2Phone,Firefly and MSN too. To get more information about that, contact with me offline or goto our website please. Regards. David at iaxtalk.com http://www.iaxtalk.com - Original Message - From: Adi Linden [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, January 24, 2005 10:16 AM Subject: [Asterisk-Users] SIP USB Phone? There are a number of Skype USB phones available. Are there any when connected to a Windows PC can access Asterisk? Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy's apparantly failing in the field
BTW they also an iax2 ATA Try here for the iax2 phone http://www.ngtel.de/products.php#1 Do you have a contact email for these guys? I couldn't see anything listed on their site anywhere. Seems the site is in current development. Matt Hi Matt, I was just getting ready to try to order a IP phone and ATA in the morning. This is the contact info I have. a.. email: [EMAIL PROTECTED] a.. Phone: +49 69 949 44 185 a.. Fax: +49 69 949 44 118 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On-Hold problem
i have had some problems with music on hold. some of the handsets havnt been able to put people on hold... When using a grandstream 101 i push hold it puts the other end on hold but doesnt play the music. although when i do it with a x-lite client it does put the other end on hold and stats the music. i havnt been able to get the grandsteam clients to play the hold music yet though :/ Computer Onsite Support [EMAIL PROTECTED] wrote : My problem is: No matter what machine I install and configure Asterisk on it I just can't get the music on-hold to work. Is anyone of you out there have such problem? If so what have you done to fix the problem? I've tried so far three other computers and none of them I was able to get music on-hold to work. CAN PLEASE ANYONE HELP. ___ NOCC, http://nocc.sourceforge.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS for Asterisk
Andrew Kohlsmith wrote: On January 23, 2005 04:04 pm, Mike Sander wrote: Is the harddisk activity on a standard asterisk install such that I don't really have to worry if the power cuts?? Not typically; there isn't much writing going on, this is true. Are you that cash strapped that a $75 UPS with a serial port is out of your budget? No kidding... the cost of a server than won't come up again is much more substantial than the countermeasure... the $75 (you can get a 350 Va for $45 even!) and a slightly less energy efficient system. If you can afford to spend more, a decent active UPS would keep your power conditioned as well... As I understand, if HD activity is minimal, the probability of HD failure is significantly reduced. HDDs don't fail because they lose power. Unless the heads crash, which can happen if power fails. I know HDD manufacturers have done head unloading and such recently, but the risk is still higher if power is suddenly lost during a write. Nick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.5
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello everyone, As you know, we released Asterisk 1.0.4 earlier this week. However, there was a callerid bug in chan_zap that has caused us to go ahead and make another release. Asterisk 1.0.5 is available at all of the usual locations. I'm sorry for any inconvenience this may cause. Russell Bryant -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.0 (GNU/Linux) iD8DBQFB9GyrrwroOS5t/FoRAqwTAJ96XvSW7QzctTkV+MBh+nLkfe8RgQCeO8Ep 68u3BuZgT9jgANDceGT1u1k= =aghw -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival
On Mon, 24 Jan 2005 14:57:06 +1300, Matt Riddell wrote: Howard Lowndes wrote: Is it possible to get the Festival command to read the text from a system file rather than having it input as a text string? Is this a case of having to use AGI, or is there a simpler way? Most people would use AGI for that (combined with the text2wave or whatever program). In fact there may even be an example on the wiki. I might also add that if you look in the wiki for cepstral as well some good examples. And cepstral voices sound much nicer than festival :-) . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] here's my IAX callthrough app and some questions about problems I have.
Did you ever get DTMF to work reliably with LiveVoip. I am having the
exact same problems.
On Mon, 17 Jan 2005 20:22:30 -0500, Jess Coburn [EMAIL PROTECTED] wrote:
Hello all,
What my app does is accepts a call in on a Dial-In Number (DID) via
IAX, and then prompts the caller for the top secret password (123) and
then authenticates the user and prompts them to dial in the number
they'd like to call. Once they press pound after dialing in the number
it will read it back to them, if they press pound it will attempt to
connect via the second IAX provider, if they press star it will allow
them enter in the number over again.
Now here's the problems and questions:
1. DTMF detection seems flawed, sometimes it's dead on but alot of
times it will see a single keypress as multiple keypresses. So I may
press 561 but it will see 51 and all three keypresses are about
the same length. Is this unique to my case or do you others see this
too. I suspect it's due to either background noise or maybe
packetloss? Any ideas on how to clean this up?
2. The only way I can get the app to fire off is if I put the
extension mapping in as _NXXNXX,1,CMD I'd like to use s,1,CMD but
I don't know what I'm missing here or doing wrong.
Below are a copies of my extensions.conf file and my iax.conf file.
Regards,
Jess
extensions.conf
file-
[general]
static=yes
writeprotect=no
[globals]
${OUTGOING-NUM}=
[arbitrary-in] ; -- Should match the context you have
; under [incoming] in iax.conf
exten = _NXXNXX,1,Answer
exten = _NXXNXX,2,Background(vm-password)
exten = _NXXNXX,3,Authenticate(123)
exten = _NXXNXX,4,Playback(beep)
exten = _NXXNXX,5,SetVar(NR=)
exten = _NXXNXX,6,Goto(testdtmf|s|1)
;
; This context is used by the sample [arbitrary-name]
; context above to read back each digit you press.
;
[testdtmf]
exten = s,1,SetVar(NR=)
exten = s,2,Background(pls-entr-num-uwish2-call)
exten = s,3,Background(and-prs-pound-whn-finished)
exten = s,4,Background(beep)
exten = s,5,WaitExten(10)
exten = _x,1,SetVar(NR=${NR}${EXTEN})
exten = _x,2,NoOp(${NR})
exten = _x,3,Goto(testdtmf|s|5)
exten = _#,1,Goto(verifynumber|s|1)
exten = i,1,Goto(testdtmf|s|1)
exten = t,1,Hangup
[verifynumber]
exten = s,1,Background(you-dialed)
exten = s,2,SayDigits(${NR})
exten = s,3,Background(if-correct-press)
exten = s,4,Background(pound)
exten = s,5,Background(otherwise-press)
exten = s,6,Background(star)
exten = _#,1,Background(pls-wait-connect-call)
exten = _#,2,Dial(IAX2/[EMAIL PROTECTED]/${NR},30)
exten = _#,3,Background(something-terribly-wrong);
exten = _#,4,Background(goodbye)
exten = _#,5,Hangup
exten = _*,1,Goto(testdtmf|s|1)
iax.conf file
--
; iax.conf
[general]
${INCOMING-USR}=SECRET-USERNAME
${INCOMING-PWD}=SECRET-PWD
${LIVEVOIP-SVR}=217.160.244.186
bandwidth=high
disallow=lpc10
jitterbuffer=yes
dropcount=2
maxjitterbuffer=500
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1
register = ${INCOMING-USR}:[EMAIL PROTECTED]
tos=lowdelay
[incoming]
; this is the incoming IAX provider
type=user
secret=ITS-SECRET
deny=0.0.0.0/0.0.0.0
permit=217.160.244.186/255.255.255.0
context=arbitrary-in
[outgoing]
;this is the outgoing IAX provider
type=peer
host= 216.118.117.46
secret= ITS-SECRET
auth=md5
notransfer=yes
context=default
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Re: [Asterisk-Users] UPS for Asterisk
On Jan 23, 2005, at 7:30 PM, Nick Bachmann wrote: As I understand, if HD activity is minimal, the probability of HD failure is significantly reduced. HDDs don't fail because they lose power. Unless the heads crash, which can happen if power fails. I know HDD manufacturers have done head unloading and such recently, but the risk is still higher if power is suddenly lost during a write. And, in fact, some drives *do* have problems with sudden outages. Some recent IBM drives will interpret sectors that were only partially written when the power failed as bad blocks and refuse to read or write to them when the power comes back on. I wouldn't be surprised if other drives have similar problems. FWIW, we have a drive in a test system at work that started developing problems immediately after a power outage a couple months ago. It might be just a coincidence, but the timing was right--the power went out in the afternoon, and the evening SMART media check found bad sectors. Scott ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS for Asterisk
And, in fact, some drives *do* have problems with sudden outages. Some Relative to the cost of a cheap UPS, downtime is much much much more expensive. You can power pretty much any single server you want for ~$150 CDN, and shut it down cleanly when the power goes out. Compare $150 with the cost of rebuilding the machine and it's money well spent. That doesn't even consider the screaming customers. Every machine I have in the field with a hard disk has a UPS - sometimes only a 350VA UPS, but a UPS none the less. The machines that boot from CF cards are a different story... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music On-Hold problem
Title: RE: [Asterisk-Users] Music On-Hold problem It should work right off the install.. Make sure you have MPG123 installed and running. _ From: Computer Onsite Support [mailto:[EMAIL PROTECTED]] Sent: Sunday, January 23, 2005 3:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Music On-Hold problem My problem is: No matter what machine I install and configure Asterisk on it I just can't get the music on-hold to work. Is anyone of you out there have such problem? If so what have you done to fix the problem? I've tried so far three other computers and none of them I was able to get music on-hold to work. CAN PLEASE ANYONE HELP. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS for Asterisk
On January 23, 2005 10:30 pm, Nick Bachmann wrote: HDDs don't fail because they lose power. Unless the heads crash, which can happen if power fails. I know HDD manufacturers have done head unloading and such recently, but the risk is still higher if power is suddenly lost during a write. Why would the heads come in contact with the platters on a powerfail? The arms are very rigid -- the heads only float a few thousandths of an inch over the platters -- something that I don't believe has anything to do with the platters spinning (that may *help* but I don't think the heads will contact the platters if they're not spinning) and besides -- any drive manufactured in the last 5 years will autopark on power fail... There's an awful lot of energy stored up in the spindle motor that is used to slam the heads into the parking zone... -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaprtc from bristuff? not there?
I'm trying to take advice to use zaprtc from bristuff (from both a posting here and references on voip-info) because I have a 2.4 kernel SMP machine. I've downloaded and installed bristuff-0.2.0-RC3a and now have the modules zaphfc and zaptel loaded. Running meetme says the extension is invalid (I've double checked meetme.conf and extensions.conf). There is no zaprtc version in that bristuff package (I thought there would be based on other posts). Trying to load zaprtc 0.0.1 gives the same error as the standard asterisk distribution. Do I download something more recent than zaprtc 0.0.1? Can anyone tell me were (no luck googling)? Other insights? Thanks much! Spencer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip - h323 translation stability capacity limit
Hi! All I would appreciate if someone could advice me on how stable is sip-h323 h323-sip translation as well as how many calls can it handle when doing such translation.( assuming single 2.8Ghz intel processor 1GB RAM) Regards, John -- ___ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Peculiar one way convesation fault with Asterisk.
Dear Fellow Ast-Masters Has any one here experienced the following issue (or similar). Setup Description: - Asterix Server - 192.168.1.10 - Analog Phone connected through FXS to Asterisk Box - (ext) 2000 - SoundPoint IP 300 - 192.168.1.2 - (ext) 4004 Problem: I have a very basic sip.conf, extension.conf and zapata.conf. Now when I try using 4004 my call doesn't go through. But if I dial 4004 from 2000, it works like a charm. And, soon after I dial 4004 from 2000, if I try using 4004, it works fine too. Now if I leave it idle for a while and then try using 4004 to dial any number it doesn't work. The LCD on my soundpoint IP shows status as 'connecting' and then after a while gives up. Whats even more peculiar is, if I issue 'reload' command in my asterisk server console it starts working fine. Problem solving attempts: To find out more I started sniffing packets in asterisk server. In ethereal this is exactly what I see, Source Dest Info 192.168.1.2 192.168.1.10 SIP Request (blah blah) 192.168.1.10192.168.1.2 ICMP: Destination Unreachable, type-3, code-10 (host administratively prohibited) 192.168.1.2 192.168.1.10 SIP Request (blah blah) 192.168.1.10192.168.1.2 ICMP: Destination Unreachable, type-3, code-10 (host administratively prohibited) repeat After some reading I found out that this type of ICMP messages are normally sent out by the router when a host is blocked. The document I read also adds, its mostly used in US government implementations(?). I haven't seen this type of ICMP message previously and not completely sure if I am interpreting it correctly. Are these messages even getting up to Asterisk server? If not, why not? if yes, then why is asterisk not accepting these messages? Any leads or help greatly appreciated. -r PS: I have googled around, but to no avail. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS for Asterisk
On 24/01/2005, at 3:26 PM, Andrew Kohlsmith wrote: On January 23, 2005 10:30 pm, Nick Bachmann wrote: HDDs don't fail because they lose power. Unless the heads crash, which can happen if power fails. I know HDD manufacturers have done head unloading and such recently, but the risk is still higher if power is suddenly lost during a write. Why would the heads come in contact with the platters on a powerfail? The arms are very rigid -- the heads only float a few thousandths of an inch over the platters -- something that I don't believe has anything to do with the platters spinning (that may *help* but I don't think the heads will contact the platters if they're not spinning) and besides -- any drive manufactured in the last 5 years will autopark on power fail... There's an awful lot of energy stored up in the spindle motor that is used to slam the heads into the parking zone... Yet it is still a problem, and still happens. In fact, I've had three machines in the last three months. It was enough to convince the cusotmers that a UPS was indeed what they needed to protect their investment. A UPS is a good investment. It protects hardware against anything going wrong, and allows for those rare, but painful blackouts that sometimes occur. Andrew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] finding current codec?
See www.websitemanagers.com.au/asterisk/ It will allow anyone to contribute their tools/etc... Of course, some things are more suitable to the wiki (eg, dialplan snippets/info/etc). Regards, Adam On Sat, 2005-01-22 at 13:12 +0200, Mike Tkachuk wrote: Hello, I dunno if it's really needed, we should ask Mark. Anyway I created site http://b2bua.berlios.de where I will post all my asterisk patches and applications. On Fri, 21 Jan 2005 11:57:52 -, Muhammad Nasim [EMAIL PROTECTED] wrote: Hi Mike This is a damn useful app. Do you know if its been put in cvs yet? Kind Regards Mo Muhammad Nasim Telappliant Ltd Tel: 020 7043 3492 Int: +44 20 7043 3492 Main: 0845 004 4040 Fax: 0845 004 4041 www.telappliant.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music On-Hold problem
What handset? Some such as the Planet dont work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, January 24, 2005 1:10 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Music On-Hold problem My problem is: No matter what machine I install and configure Asterisk on it I just can't get the music on-hold to work. Is anyone of you out there have such problem? If so what have you done to fix the problem? I've tried so far three other computers and none of them I was able to get music on-hold to work. CAN PLEASE ANYONE HELP. File: ATT00718.txt attachment: winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip - h323 translation stability capacity limit
HI John, It also depends which H323 channel you will use for this translation. I can recommend you to use the chan_oh323 from inAccess Networks - according to our experience it's much stable and bug free channel. Our Asterisk based translation system is running much stable with chan_oh323 .. The sip-h323 h323-sip translation now is working pretty well :) Best regards, Lubo - AppRadius Project: Full RADIUS AAA support for Asterisk PBX http://appradius.minitelecom.org/ - [EMAIL PROTECTED] com wrote: Hi! All I would appreciate if someone could advice me on how stable is sip-h323 h323-sip translation as well as how many calls can it handle when doing such translation.( assuming single 2.8Ghz intel processor 1GB RAM) Regards, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for a prepaid calling card platform
I'm looking for a prepaid calling card platform that: * easily scales to multiple servers with a common database for: redundancy, capacity, and performance Looking to start with capacity to handle100 simultaneous calls andbe able to easilyscale to 1000+ simultaneous calls. * in addition to the normal anti-fraud measures, supports an API for easily adding new anti-fraud tests along the lines of the following: For each newcall being attempted the system wouldinvoke an external authentication program and pass:reseller ID, card ID, time left on card, called # andcalling #; and the history/status for thelast several calls including for each call the called #, calling #, call duration, call timestamp andcall status (in-progress, completed, etc). Progrm would return: call OK, deny call with recording #x, invalidate card with recording #y. * ability to limit calls to a maxium duration and/or to require periodic IVR user response to continue a long call. * contolled, managed andprovisionedwith a web interface * support multiple resellers, each with password protected web access for managing their customers. * ability for customers to call an 800# tohear a recording giving themthe user a local non-800number they need to call to use the card. * credit card recharge support While willing to do minor customizations, would like to find something that is mostlyinstall and go. Open source would be nice, but willing to pay for a well done package. Suggestions welcome. Thanks. Jim James H. Thompson[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IAX Inbound Sound Quality
On Sun, 23 Jan 2005 01:51:56 -0500, Andrew Kohlsmith I have *no* issues on inbound quality with sixTel. They *had* a problem where the first second of audio was cut off upon connect (Wait() did not help) but that seems to have been fixed. I see this problem intermittently, typically during a long call, after 45 min. or so. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS for Asterisk
On Sun, 23 Jan 2005, Andrew Kohlsmith wrote: Why would the heads come in contact with the platters on a powerfail? The arms are very rigid -- the heads only float a few thousandths of an inch over the platters -- something that I don't believe has anything to do with the platters spinning (that may *help* but I don't think the heads will contact the platters if they're not spinning) and besides -- any drive manufactured in the last 5 years will autopark on power fail... There's an awful lot of energy stored up in the spindle motor that is used to slam the heads into the parking zone... Actually, the only thing that keeps the heads off the platter is the fact that they are spinning. The movement of the platters cause an airstream which the heads float on. This airstream is what keeps the heads at just the right distance. The arms are not very rigid at all in the axis direction of the disks. This has been the standard design in hard disks since a very long time. The comment about autopark is correct. Actually, with the voice coils used on modern disks the energy needed to retract the heads is already stored in the return spring. The platter energy is sometimes used to complete any sector write that is in progress. Some hard disks did not do this and those generated bad sectors every time they were powered down in mid-write. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
