Re: [Asterisk-Users] TDM400 lost after reboot
do a google search for tdm400p hardware problems (fix) This is a problem with the tdm card and driver If you are using the older zaptel software the file referenced in the doc is wcfxs.c if you are using the cvs version the wcfxs file needs to be replaced with wctdm.c also the line number 2127 is changes in the wctdm file so do a search for pci_device_id or go to approx line 2130 HTH Greg At 09:37 AM 1/16/05, you wrote: Hi My card is working, but when I reboot the machine, most of the times it is not working, I get ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6) To make it work again I have to shut down, remove the card, reboot so kudzu will remove the config. shut down again, put the card back in, reboot, now kudzu see it, I choose Ignore and then it's working again (until the next reboot). I'm on WBEL 3.0 and the card is not sharing is IRQ. Is anybody else having this problem ? When kudzu see it (as a Jens Schoenfeld Intel 537), what should I choose ? Is there something I can do to prevent this from happening ? Thanks ___ 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 Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ 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] Qs about FXO/FXS cards
At 03:25 PM 1/3/05 -0500, you wrote: Unfortunately that makes Asterisk installs for small businesses more expensive than necessary. At US$500 for a T100P and US$300ish for a channel bank (FXS only, FXO is significantly more expensive!) plus your time and system for an Asterisk install it raises the bar for the small business to adopt Asterisk. The TDM400P would fit a very nice little niche if it worked reliably. Let's face it -- most businesses are looking at VOIP to reduce their telephone bills and if the time it takes for the install to pay for itself is raised significantly (like an added $1000 price tag for reliable equipment)... well... the writing on the wall is pretty clear. -A. With the above said, now you have just entered the realm of the talkswitch. I was speaking with a cable installer friend of mine, who told me he installs the talkswitch at all the jenny craig franchises (the franchise, I assume pays for the devices as he does not resell them). His words talkswitch is great, just plug it in and it works, pbx and voip, you can't beat it He also wires them to their pa system. List price is 1800 bucks for the top of the line unit. 8fxo, 16fxs, ethernet When you try to sell the asterisk system, you have to compete with that and frankly, all the people want is to make phone calls. Mention voice over ip and eyebrows raise, I've heard of that, but in reality nobody cares how their phone calls are made, just that it goes through. If you can't save them a bunch of money, there is little or no reason to diverge to a more costly system, that will save in the long run, regardless of the additional feature set, which if they can't touch it, feel it taste it, smell it, smoke it, makes no difference at all. In reality, I want to off as many boxes as I can, maybe tie in some service contracts, and be done with it. From our experience, most small office/businesses have a bunch of phone lines, 3-8, and are leery of spending a chunk of change on a new system and possibly phones as well. 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] Qs about FXO/FXS cards
At 11:34 AM 1/4/05, you wrote: On Tue, 4 Jan 2005 10:08:27 -0500, Daryl G. Jurbala wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Monday, January 03, 2005 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Qs about FXO/FXS cards [...] For business use, I would suggest you first find a BRI card you can use here in the states. Hint, bug Kapejod into making that 4 port card US ready. Then move any business user over [...] That might work out where you do your deployments. In Verizon territory, you can get analog business lines with unlimited long distance and no metered minutes for about $37 a month. A BRI costs you about double that for the loop, with metered minutes and bring your own LD. Past the technology aspects, BRI just doesn't work here. And I'm going to guess that pricing structure is similar in other areas as well. Daryl Are you talking about residential lines with those rates? Business rates for POTS lines are more than that here in Houston. Michael business rates here in the North East (us) are 49/mo cheepy T1's start at about 250/mo plus minutes, one case I can think of is 2cents per minute 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] Qs about FXO/FXS cards
At 12:29 PM 1/4/05, you wrote: Greg - Cirelle Enterprises wrote: When you try to sell the asterisk system, you have to compete with that and frankly, all the people want is to make phone calls. Mention voice over ip and eyebrows raise, I've heard of that, but in reality nobody cares how their phone calls are made, just that it goes through. If you can't save them a bunch of money, there is little or no reason to diverge to a more costly system, that will save in the long run, regardless of the additional feature set, which if they can't touch it, feel it taste it, smell it, smoke it, makes no difference at all. Yeppers. Then in six months or a year, or whatever timeframe (these things are expected to provide many years of use by those who consider their price expensive) they will ask you, How do I use one of these newfangled ITSPs with our system? Why do I have to still have to use an answering machine? Why am I paying $7/line for CallerID? Can I set this thing up to automatically forward to my cell phone when I'm not in the office? Can we have an autoattendant like everyone else does now? Why do I still have to pay for conferencing? Can I set things up so that all the sales phones ring at the same time until someone picks one up? Etc. etc. I fielded questions like these from a businessman the other day who loudly bemoaned having invested some fairly hefty cash (in small business terms) on a Nortel key system, THREE YEARS AGO. Imagine what he would have sounded like had he cut that check last month instead. Maybe these talkswitches are smart and can do a lot of those things. I don't know anything about them. I do think the play that VoIP is getting, pretty much all over the place, in the mainstream and business media, will soon result in the average businessperson knowing much more about it than is presently known. At that point some of the economy mentioned in your email is going to seem misguided, IMO. B. They come with all the above ___ 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] Qs about FXO/FXS cards
At 12:50 PM 1/4/05, you wrote: Sipura SPA 3000... forget the channel bank and PRI card. Buy a PRI card and ebay the SPAs when you arte ready to move from POTS to PRI, or better yet, forget both and find an ITSP that can offer QoS (private line!!!) and interface with * Talkswitch? Get on the VoIP bus or get run over buy it, your choice. comes with it... ___ 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] PSTN to VoIP
At 02:17 PM 1/3/05, you wrote: I m about to purchase an adaptor for a POTS data modem and was looking at the Sipura line of adaptors (SPA-1000, SPA-1001, SPA-2000, SPA-3000). Do these work well? Anyone have a suggestion on which model of the Sipura I should get? Does one work better with * than the others? Are there other adaptors that work better that I should get? the spa 1k and 2k serve to connect an analog handset to ethernet Not sure about pots. The 3K series may have an fxo side which connects to the phone line, but not sure about that ___ 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] Is asterisk that unstable ????
from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21 or 10 7 * * * root /usr/sbin/asterisk -r -x restart gracefully /dev/null 21 Does this software have substantial problems that one would have to do this??? Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ 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] Is asterisk that unstable ????
At 09:19 AM 12/30/04, you wrote: Greg - Cirelle Enterprises wrote: from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx restart now /dev/null 21 or 10 7 * * * root /usr/sbin/asterisk -r -x restart gracefully /dev/null 21 Does this software have substantial problems that one would have to do this??? I'm runing Asterisk for a year now as the IPBX of our little consulting firm. It stopped working only 4 times: two of these where power failures and the other two turned out to be Telco company problems (dead line). We have 2 PSTN lines (using Digium X101P cards), 5 intrernal VoIP extentions (Grandstream budgettone - one of which is located on another continent, using a Wifi connection to a near by village that hosts an ADSL router... don't ask) and 2 VoIP termination/origination lines. Of course, your mileage may very, but at least here there is no nightly restart script. Hope that helps you in any way. Gilad Are you running a stable (v 1.0 - 1.0.3) or cvs Greg ___ 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] Is asterisk that unstable ????
At 11:00 AM 12/30/04, you wrote: I wouldn't say it's unstable... these boxes all run res_perl and reload 100's of times a day. It all depends on if you know what the hell you're doing. bkw why are they reloading 100's of times a day?? greg ___ 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] MYSQL_FRIENDS
At 01:37 PM 12/27/04, you wrote: Hello *'s, Hi, I've just tried to enable MYSQL Friends in CVS HEAD. But i cannot find this option.On wiki i found this. To enable this, you need to edit the Makefile in the channels directory of your source tree and enable MYSQL_FRIENDS. This enables database definition of both IAX2 and SIP friends. Make sure you have the MySQL development kit (libraries) installed before compilation.But where is MYSQL_FRIENDS option.I can't find it.I used Latest CVS. you need to download asterisk-addons ___ 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 dies no calls in or out
does anybody know what these log messages mean? what ever it is, asterisk needed a restart to become active again. (server was not rebooted and remained live to ssh and other network functions.) no outgoing calls can be made. The system was just sitting idle over night and trying to make a call about mid morning failed. == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Binding sip.conf to mysql/asterisk/sip_config == Binding voicemail.conf to mysql/asterisk/voicemail_config == Binding sipfriends to mysql/asterisk/sip_buddies == Binding voicemail to mysql/asterisk/voicemail_users Asterisk CVS-HEAD-12/13/04-07:50:27, Copyright (C) 1999-2004 Digium. W 1 tdm400p with 4fxo ports (all lights remain lit) zaptel was not restarted Dec 27 10:42:40 DEBUG[31402]: *18007354887 is not a local user Dec 27 10:48:23 DEBUG[31409]: Device 'Zap/25' changed to state '2' Dec 27 10:48:24 DEBUG[31410]: Prodding channel 'Zap/25-1' Dec 27 10:48:24 DEBUG[31410]: Scheduling timer at 160 sample intervals Dec 27 10:48:24 DEBUG[31410]: Generator got voice, switching to phase locked mode Dec 27 10:48:24 DEBUG[31410]: Scheduling timer at 0 sample intervals Dec 27 10:48:25 DEBUG[31410]: Auto-deactivating generator Dec 27 10:48:25 DEBUG[31410]: Scheduling timer at 0 sample intervals Dec 27 10:48:25 DEBUG[31410]: Prodding channel 'Zap/25-1' Dec 27 10:48:25 DEBUG[31410]: Scheduling timer at 160 sample intervals Dec 27 10:48:25 DEBUG[31410]: Generator got voice, switching to phase locked mode Dec 27 10:48:25 DEBUG[31410]: Scheduling timer at 0 sample intervals Dec 27 10:48:25 DEBUG[31410]: Auto-deactivating generator Dec 27 10:48:25 DEBUG[31410]: Scheduling timer at 0 sample intervals Dec 27 10:48:25 DEBUG[31410]: Prodding channel 'Zap/25-1' Dec 27 10:48:25 DEBUG[31410]: Scheduling timer at 160 sample intervals Dec 27 10:48:25 DEBUG[31410]: Generator got voice, switching to phase locked mode Dec 27 10:48:25 DEBUG[31410]: Scheduling timer at 0 sample intervals Dec 27 10:48:25 DEBUG[31410]: Auto-deactivating generator Dec 27 10:48:25 DEBUG[31410]: Scheduling timer at 0 sample intervals Dec 27 10:48:25 DEBUG[31410]: Prodding channel 'Zap/25-1' Dec 27 10:48:25 DEBUG[31410]: Scheduling timer at 160 sample intervals Dec 27 10:48:25 DEBUG[31410]: Generator got voice, switching to phase locked mode Dec 27 10:48:25 DEBUG[31410]: Scheduling timer at 0 sample intervals Dec 27 10:48:26 DEBUG[31410]: Auto-deactivating generator Dec 27 10:48:26 DEBUG[31410]: Scheduling timer at 0 sample intervals Anyone seen this type behaviour before or have a clue to what this means? Regards Greg Cirino ___ 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: Asterisk dying...
At 05:49 PM 12/27/04, you wrote: I have a similar problem with my *.Works fine but after some number of hours, nothing works with no apparent reason. Restarting * fixes everything. I hope someone comes up with some suggestions! Norm Z I just downloaded a new cvs to see if that helps, Regards Greg Cirino ___ 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]
At 06:12 PM 12/27/04, you wrote: James Moran wrote: I just updated my asterisk box and now it's giving me this error I looked it up on the internet found no solutions any other information that you need please ask. [EMAIL PROTECTED] root]# modprobe wcfxo /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: insmod /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o failed /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: insmod wcfxo failed What type of card is it? It is an X100P? If not, you should be modprobing wcfxs or wctdm for recent versions (with a TDM400P card). Also, what does dmesg say? -- Cheers, Matt Riddell did you rebuild the zaptel driver as well? If you did, did you also copy the /usr/src/zaptel/zaptel.init to /etc/init.d/zaptel ??? I had that error because my zaptel was older and used the wcfxs but needed to load wctdm because of a file name change. If you tried all of that and there is still issues, sorry I couldn't be of more help Greg ___ 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] ALERT_INFO issue CVS-HEAD-12/24/04
At 06:24 AM 12/25/04, you wrote: On Sat, 25 Dec 2004, John Bittner wrote: Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. This is a what I have in my dialplan. exten = 207,1,SetVar(ALERT_INFO=Ring Answer) exten = 207,2,Dial(SIP/207) exten = 207,3,Hangup This has been covered onm asterisk-users already. The syntax for passign ALERT_INFO has changed. Set the variable _ALERT_INFO instead of ALERT_INFO. The new, outgoing, channel will inherit ALERT_INFO then. Peter isn't it wonderful how that made it into the change log... or is my esp wearing out :P ___ 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] Registration failure with debug
can anybody identify why the CLI is issuing a failure message while debug shows everything is fine this makes no sense to me. also, why is the username being updated? this has got to be wrong from CLI -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]: chan_sip.c:7742 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 -- Saved useragent Grandstream BT100 1.0.5.20 for peer 52221 -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 from debug /var/log/messages Dec 24 12:16:34 DEBUG[15776]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '52221' Dec 24 12:16:34 DEBUG[15776]: MySQL RealTime: Everything is fine. Dec 24 12:16:34 VERBOSE[15776]: -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '52221' Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Everything is fine. Dec 24 12:16:35 VERBOSE[15776]: -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '52221' Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Everything is fine. Dec 24 12:16:35 VERBOSE[15776]: -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 Dec 24 12:16:35 VERBOSE[15776]: -- Saved useragent Grandstream BT100 1.0.5.20 for peer 52221 Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Update SQL: UPDATE sip_buddies SET ipaddr = '192.168.70.26', port = '5060', regseconds = '1103912195', username = '52221' WHERE name = '52221' Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Everything is fine. Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Updated 1 rows on table: sip_buddies Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '52221' Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Everything is fine. Dec 24 12:16:35 VERBOSE[15776]: -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 Dec 24 12:16:35 DEBUG[16042]: Device 'SIP/52221' changed to state '0' Dec 24 12:16:50 DEBUG[15776]: Auto destroying call '[EMAIL PROTECTED]' Regards Greg Cirino ___ 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] Grandstream 1.0.5.20 firmware?
Got that from grandstream, and testing it for a couple of things Greg At 12:47 PM 12/24/04, you wrote: Greg - Cirelle Enterprises wrote: -- Saved useragent Grandstream BT100 1.0.5.20 for peer 52221 Greg, Completely unrelated to your current query. Your logs show that your BT100 is running 1.0.5.20 firmware. Is this correct? The last I knew, they were at 1.0.5.18 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 Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ 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 seeding vs registration
At 07:00 PM 12/22/04, you wrote: What registration failure is that? from the asterisk messages log: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25' The only way to tell is a complete SIP trace of what's going on. That may be, but the point is when the registration failure like above occurs, the phone is useless, the calls directed to that phone go to voice mail The registration timeout on the phone and in Asterisk should be the same, unless the server goes down and reboots. The server usually has no way to tell a phone to re-register (no real need to do so) On the next phone registration they will be in sync again. We tried that but still had the registration failures. What has stopped the registration failures is stripping out a bunch of unused (in our case) modules to try to isolate the issue. so far No Registration Failures have been detected. the following is what our current modules.conf file looks like: modules.conf [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so noload = chan_modem.so noload = chan_modem_aopen.so noload = chan_modem_bestdata.so noload = chan_modem_i4l.so noload = chan_mgcp.so noload = chan_skinny.so ; require for voicemail load = res_adsi.so load = res_musiconhold.so noload = app_festival.so noload = app_url.so noload = app_image.so noload = app_disa.so noload = app_qcall.so noload = app_adsiprog.so noload = app_ices.so noload = codec_lpc10.so noload = codec_g729.so noload = codec_g726.so noload = codec_alaw.so noload = format_vox.so noload = format_h263.so noload = format_jpeg.so noload = cdr_csv.so noload = cdr_manager.so noload = app_zapras.so noload = app_flash.so noload = app_zapbarge.so noload = app_zapscan.so noload = app_talkdetect.so noload = app_alarmreceiver.so noload = chan_alsa.so noload = chan_oss.so noload = res_config_odbc.so noload = res_odbc.so noload = cdr_odbc.so noload = cdr_pgsql.so noload = app_realtime.so [global] chan_modem.so=no Eventually, we will retry the app_realtime again, but so far that has been a failure. The more pressing issue is the registration failure issue Greg ___ 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] Realtime sipbuddies table structure why?????
Is there some reason the sipbuddies table structure was designed with sip config values as column names? Doesn't look very flexible It really should take the form of ast_config so when a new sip feature is implemented, you don't have to re-write the entire data structure too. Regards Greg Cirino ___ 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] Registration Failure Directly related to realtime
Apparently, the realtime system in asterisk is faulty. Implementing realtime, begins a host of seeding messages along with registration messages visible at the CLI prompt. This is not the case with .conf file configuration Unfortunately, it is not clear where the bug originates but is shows it's head while calling the register_verify function, (which there are 2 one in chan_sip.c and one in chan_iax2.c) from the error message, I would guess it is coming from the chan_sip file, but with 2 functions of the same mane in one program, who knows. Dec 23 09:24:57 NOTICE[12406]: chan_sip.c:7742 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' As I have noted in previous posts, in our case, when the extension fails registration all calls to the extension are sent directly to voice mail. Seeding makes no difference here. Also, Registration Failures are sporadic. The frequency they appear, makes no real sense. One thing is certain. if you do not use (load) app_realtime the errors do not occur. shortly after you load app_realtime the errors begin. too bad this application was written around config files instead of a database to begin with. So 70's Regards Greg Cirino ___ 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] Realtime sipbuddies table structure why?????
At 09:53 AM 12/23/04, you wrote: It was written the way it is because that is how RealTime works. =P If you don't like the schema design, talk to Mark so he can rewrite RealTime for you. Read up some more on how RealTime works then you will understand why all the tables are designed the way they are. Read docs/README.extconfig -Matthew Read it, makes no difference, it's broken :) Also, it doesn't say why the table structure is the way it is. just poor data modeling. greg ___ 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] Realtime sipbuddies table structure why?????
At 10:32 AM 12/23/04, you wrote: just poor data modeling How so? How would you change it? Are you aware that they have written code into app_voicemail.c that allows you to store the actual soundfiles for voicemail in the database itself? You want to talk about poor database design...sheesh.. because it works, doesn't make it right Read it, makes no difference, it's broken :) Whats broken? from tons of previous posts.. Dec 23 09:24:57 NOTICE[12406]: chan_sip.c:7742 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' happens with app_realtime, doesn't happen without realtime there are previous posts identifying issues with this. Also, it doesn't say why the table structure is the way it is. It most certainly does. Seems you didn't read the REAME after all. Quoted from README.extconfig: quote It is designed to provide a flexible, seamless integration between Asterisk's internal configuration structure and external SQL other other databases the columns in your tables should line up with the fields you would specify in the given entity declaration. For example, an entity that looks like: [foo] host=dynamic secret=bar context=default context=local could be stored in a table like this: +--++---+--+--+-+---+ | name | host | secret| context | ipaddr | port| regseconds| +--++---+--+--+-+---+ | foo | dynamic| bar | default;local| 127.0.0.1| 4569| 1096954152| +--++---+--+--+-+---+ /quote Seems pretty simple and easy to use. This way if new config options are ever added, all you have to do to support them is to add a new column. And if all you are storing in most columns is 1 byte, it can't take up that much space. -Matthew I think you are still missing the point. for example, host, secret, context, ipaddr, port, regseconds should not be column names, they should be data points with an associated column to contain the related value(s) From what I can see, i can't add a column say musicclass = something, or pedantic = yes like I could in the conf file and have it mean anything. regards greg - Original Message - From: Greg - Cirelle Enterprises [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 23, 2004 9:00 AM Subject: Re: [Asterisk-Users] Realtime sipbuddies table structure why? At 09:53 AM 12/23/04, you wrote: It was written the way it is because that is how RealTime works. =P If you don't like the schema design, talk to Mark so he can rewrite RealTime for you. Read up some more on how RealTime works then you will understand why all the tables are designed the way they are. Read docs/README.extconfig -Matthew Read it, makes no difference, it's broken :) Also, it doesn't say why the table structure is the way it is. just poor data modeling. greg ___ 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 Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ 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] Fw: [digium.com #12961] T100P as bandwidth
At 10:37 AM 12/23/04, you wrote: Even though they make the cards and advertise that they support data modes, digium won't support data mode on the $500 card they sold to me, so I must turn to the list. Has anyone configured a T100P to use a T1 strictly as bandwidth? Is there a HOWTO somewhere? Wiki has nothing I could find. I've got plently of public IPs I can assign to it but don't know how. Thanks, Matthew I've gone round and round with them on this. From what I understand, the card will only work with a pri type t1 not a data line unless you have a device that will emulate that configuration. (same with the sangoma card by the way) only sangoma will tell you that. If you are thinking of a return on this card, They will tell you the loopback works so the card is ok, no return. ___ 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] Fw: [digium.com #12961] T100P as bandwidth
At 10:43 AM 12/23/04, you wrote: On December 23, 2004 10:37 am, Matthew Boehm wrote: Even though they make the cards and advertise that they support data modes, digium won't support data mode on the $500 card they sold to me, so I must turn to the list. If Digium won't support it return the card and get a Sangoma A101u, it's approximately the same price and they've been doing HDLC/data T1s for damn near a decade. I am a fan of Digium but I am quickly growing tired of the lack of responsiveness from them on their hardware. Software wise I have *zero* complaints but the story is quite different when it comes to hardware support. :-( -A. I spoke with a fellow, (can't remember his name, but had a british accent, there are only about 10 folks working there) at sangoma, and he specifically said the sangoma card will only work with a pri t1 (24channel isdn) not with a data line. regards greg ___ 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] Fw: [digium.com #12961] T100P as bandwidth
At 11:17 AM 12/23/04, you wrote: On December 23, 2004 10:59 am, Greg - Cirelle Enterprises wrote: I spoke with a fellow, (can't remember his name, but had a british accent, there are only about 10 folks working there) at sangoma, and he specifically said the sangoma card will only work with a pri t1 (24channel isdn) not with a data line. That's David -- I have used Sangoma T1 cards for strictly data (no PRI, this was Frame Relay) for years -- perhaps not the A101u then but they do have T1 cards that do data. Yes David. The card I was discussing with him was the A100 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] Fw: [digium.com #12961] T100P as bandwidth
At 11:52 AM 12/23/04, you wrote: On December 23, 2004 11:14 am, TC wrote: but thats the bitch Mark has put years of blood sweat into it, now as asterisk start to become much bigger than the single developer/co how do you divest that control in a fair/equitable manner I agree with you on all points -- If Digium needs to make money on hardware then they obviously need to get some decent hardware/driver design done -- What we have works for the most part but these ongoing problems and the almost total lack of dialogue is a big issue which *does* hurt future sales. I like Digium. I want to buy Digium. But I won't if the support or quality isn't there. It's really that simple. Digium's in a hard spot -- spend more money to fix the issues and eat into current revenues, or deal with lower future revenues. I don't envy them. -A. If it aint broke, don't fix it... in this case, it's broke. ___ 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] TDM400 success?
At 12:14 PM 12/23/04, you wrote: On Thu, Dec 23, 2004 at 09:58:19AM -0700, Damon Estep wrote: Has anyone had success with the TDM400 in production? I have multiple boxes where these cards lock up and the only thing that will fix them is to unload *, modprobe -r wctdm, modprobe wctdm, load asterisk. Does not matter if it is a FXS/FXO module. I have a recently installed TDM400P with one FXO in slot 4 which hasn't locked up yet, but it's only been a day or so at this point and usage is light so far... -Dorn We have one TDM400 with 4FXO Daughter Boards, that have been running for a couple of months. The main problem we had was the issue where the card would not come alive on reboot. Fixing the driver code to have the card recognize a key helped fix it, but the new zaptel drivers don't do the trick which for now there is no upgrading zaptel. Broken stuff??? We Report, You Decide ___ 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] Realtime sipbuddies table structure why?????
At 01:21 PM 12/23/04, you wrote: Greg - Cirelle Enterprises wrote: Read it, makes no difference, it's broken :) Also, it doesn't say why the table structure is the way it is. just poor data modeling. God, I'm sure everyone on the list must be thinking, Oh, why oh why didn't *Greg* write Asterisk instead of Mark; he seems so very much smarter. . . B. Don't claim to be smarter, just pointing out the obvious greg ___ 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 seeding vs registration
At 03:43 PM 12/23/04, you wrote: Oh, I see. This is the realtime connected problem. Can't say too much constructive about that without info, I'm not a fan of it. We need a debug trace of the registration process (SIP trace and * messages) to debug why it failed, not just a one-line message, and anything after that is useless, as you point out. However, I don't think it has anything do to with loading (or not) all your modules, unless you're running out of memory. The module elimination was to try and rule out memory issues as the machine is limited to 512MB RAM. When utilizing the app_realtime: The CLI interface is consistently issuing these messages. it has been a slow day with no real phone activity. -- SIP Seeding '40853' at [EMAIL PROTECTED]:5060 for 3600 -- Saved useragent Sipura/SPA2000-2.0.10(e) for peer 40853 -- SIP Seeding '40853' at [EMAIL PROTECTED]:5060 for 3600 -- SIP Seeding '40854' at [EMAIL PROTECTED]:5061 for 3600 -- SIP Seeding '40854' at [EMAIL PROTECTED]:5061 for 3600 -- Saved useragent Sipura/SPA2000-2.0.10(e) for peer 40854 -- SIP Seeding '40854' at [EMAIL PROTECTED]:5061 for 3600 -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 Dec 23 16:22:44 NOTICE[12551]: chan_sip.c:7742 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 -- Saved useragent Grandstream BT100 1.0.5.20 for peer 52221 -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 -- SIP Seeding '1002' at [EMAIL PROTECTED]:5060 for 1800 -- SIP Seeding '1002' at [EMAIL PROTECTED]:5060 for 1800 -- Saved useragent X-Lite release 1103m for peer 1002 -- SIP Seeding '1002' at [EMAIL PROTECTED]:5060 for 1800 -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 3600 -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 3600 Dec 23 16:48:47 NOTICE[12551]: chan_sip.c:7742 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25' -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 3600 -- Saved useragent Grandstream BT100 1.0.5.20 for peer 40852 -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 3600 -- SIP Seeding '1002' at [EMAIL PROTECTED]:5060 for 1800 -- SIP Seeding '1002' at [EMAIL PROTECTED]:5060 for 1800 -- Saved useragent X-Lite release 1103m for peer 1002 The /var/log/asterisk/messages file gives Dec 23 12:24:00 NOTICE[12551]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' Dec 23 12:50:05 NOTICE[12551]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25' Dec 23 13:23:41 NOTICE[12551]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' Dec 23 14:23:22 NOTICE[12551]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' Dec 23 14:49:26 NOTICE[12551]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25' Dec 23 15:23:03 NOTICE[12551]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' Dec 23 16:22:44 NOTICE[12551]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' Dec 23 16:48:47 NOTICE[12551]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25' Dec 23 17:22:25 NOTICE[12551]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' D Restoring the system to using *.conf files eliminates all of this output and calls going directly to voicemail Unfortunately, I don't have the exact channel cannot be created or ? messages as there were non today and are usually seen in the CLI. Unless I'm mistaken, these general messages indicate a registration failure, do they not? When a call comes in and goes directly to voicemail while the extension is sitting idle waiting for a call, not busy or off the hook, I think is an issue. 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] Budgetone is not registering
At 07:30 AM 12/22/04, you wrote: I tried type=friend and it is registering now... I'm happy with it this time, but why can't I have the phone as user only (only to make calls) and not as peer (to receive calls)?? Thanks, RODOLFO Rodolfo Grave wrote: Hi again. I cant get my Budgetone registered in Asterisk, and I cant find what's wrong... uff. This is my config: This fragment is from my sip.conf: [12345] type=user user=12345 username=12345 secret=12345 authuser=12345 qualify=1000 nat=no host=dynamic dtmfmode=rfc2833 reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw context=sip_default And this is from my Budgetone configuration: SIP Server: 192.168.1.175 -- asterisk is in my LAN SIP User ID: 12345 Authenticate ID: 12345 Authenticate Password: 12345 And this is the message repeated in the asterisk console: Dec 21 21:36:15 NOTICE[3024]: chan_sip.c:7531 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '192.168.1.176' Can you please help? What is that user=phone about? Thanks a lot in advance, RODOLFO ___ You will probably still get the registration errors. I have backed down to a vanilla installation of ast and a very basic sip.conf. phone is all defaults except for ip addresses although, my dtmfmode is info and i only allow ulaw, alaw I began getting them this morning. This is not restricted to bugetone phones. it also happens on sipura 2k units, though less frequently. greg Regards Greg Cirino ___ 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] What is sip-friends.sql??????
At 09:44 AM 12/22/04, you wrote: Its a way of storing ur sip stuff in a database rather than using the flat files. Sip friends - extensions.conf stuff. Sip_buddies - sip.conf stuff ___ this is the database to flat file storage I take it. Regards Greg Cirino ___ 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 seeding vs registration
At 12:43 AM 12/22/04, you wrote: Seeding occurs if there is still a persistent record (in astdb) of a preceding location registration of a peer after a restart of asterisk or the sip channel. If Asterisk goes down and the peer has a long registration refresh time, the phone maybe inaccessible for a while (until its own refresh timer expires) if there is no record of its IP address after the restart. The persistent record and seeding (of the IP address) solves this. so this might be the problem with the registration failures? Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' Asterisk registration timeout is shorter than the phone registration timeout??? I'm not sure my statement makes any kind of sense, if it does, than there is a serious issue with the asterisk device communication system. Regards Greg ___ ___ 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] Another Asterisk Certification
certify this ___ 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] Status of asterisk.xvoip.com?
At 12:21 PM 12/22/04, you wrote: Did anyone here use the * forums over at asterisk.xvoip.com? I've been unable to connect for a few days now and was wondering if anyone knew if they're down for good. It'd be a shame if they are since * newbs like me need every resource we can find. Joel Moore no dns for that domain Regards Greg Cirino ___ 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] Phone Registration Failure Test
If anyone is experiencing this type of registration error: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25' try adding the following line to your modules.conf file noload = app_adsiprog.so This error is clearly asterisk trying to register with the phone and not the other way around. the app_adsiprog.so is only used to download adsi scripts to a phone. If you don't use adsi scripts, (if you don't know what they are, you probably don't) disable this module and see if your error messages stop. greg ___ 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] register_verify defined in 2 files?
I know I'm getting tired of looking at code, but why is the function register_verify defined in 2 different files? chan_iax2.c line 3860 static int register_verify(int callno, struct sockaddr_in *sin, struct iax_ies *ies) chan_sip.c line 4869 /*--- register_verify: Verify registration of user */ static int register_verify(struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req, char *uri, int ignore) Regards Greg ___ 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 somebody email me the Sipura SPA-2000 andSPA-3000 documentation?
which docs are you talking about? At 06:15 PM 12/22/04, you wrote: Yeah, I d like to get those docs too. -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan Sent: Wednesday, December 22, 2004 4:03 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Can somebody email me the Sipura SPA-2000 andSPA-3000 documentation? I heard Sipura had really awesome documentation on the SPA-2000 and SPA-3000, but you have to email them for it. When I did, they said I had to get it from a reseller. It s been a while since I bought my units, I don t even remember where or who they were bought from. Can somebody email me the documentation for these devices? I m quite interested in knowing what every one of their 200+ options does. ___ 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 Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ 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] Quick questions ( maybe a little confidence building too )
At 04:17 PM 12/20/04, you wrote: On December 20, 2004 04:02 pm, Greg - Cirelle Enterprises wrote: Could I ask how you've connected the t1s? I'm going to be getting a non-pri t1 ( 9 channels of voice, the rest off ). I assume I'll just get an rj45(ish) plug to plug into the back of the card and I'll specify my settings in /etc/zaptel.conf. Does this sound about right? this does not work with non pri t1's according to digium Care to elaborate? I have done this with Asterisk (fractional T1s for voice) I tried reading the thread but it's hurting my head to try and make sense of the quoting, so I am guessing that you're raising concerns about fractional T1s and the T100P? -A. Is this an hdlc implementation? ___ 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] What is sip-friends.sql??????
maybe a dumb question but what do we have here??? sip-friends.sql # # Table structure for table `sipfriends` # CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `username` varchar(40) default '', `ipaddr` varchar(20) NOT NULL default '', `port` int(6) NOT NULL default '0', `regseconds` int(11) NOT NULL default '0', PRIMARY KEY (`name`) ) TYPE=MyISAM; Realtime SIP # # Table structure for table `sip_buddies` # CREATE TABLE `sip_buddies` ( `uniqueid` int(11) NOT NULL auto_increment, `name` varchar(30) NOT NULL default '', `accountcode` varchar(30) default NULL, `amaflags` char(1) default NULL, `callgroup` varchar(30) default NULL, `callerid` varchar(50) default NULL, `canreinvite` char(1) default NULL, `context` varchar(30) default NULL, `defaultip` varchar(15) default NULL, `dtmfmode` varchar(7) default NULL, `fromuser` varchar(50) default NULL, `fromdomain` varchar(31) default NULL, `host` varchar(31) NOT NULL default '', `incominglimit` char(2) default NULL, `outgoinglimit` char(2) default NULL, `insecure` char(1) default NULL, `language` char(2) default NULL, `mailbox` varchar(50) default NULL, `md5secret` varchar(32) default NULL, `nat` varchar(5) default NULL, `permit` varchar(95) default NULL, `deny` varchar(95) default NULL, `pickupgroup` varchar(10) default NULL, `port` varchar(5) NOT NULL default '', `qualify` varchar(4) default NULL, `restrictcid` char(1) default NULL, `rtptimeout` char(3) default NULL, `rtpholdtimeout` char(3) default NULL, `secret` varchar(30) default NULL, `type` varchar(6) NOT NULL default '', `username` varchar(30) NOT NULL default '', `allow` varchar(100) default NULL, `disallow` varchar(100) default NULL, `regseconds` int(11) NOT NULL default '0', `ipaddr` varchar(15) NOT NULL default '', PRIMARY KEY (`uniqueid`), UNIQUE KEY `name` (`name`), KEY `name_2` (`name`) ) TYPE=MyISAM; Regards Greg Cirino ___ 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] Bug, Feature, or Limitation?
At 11:37 AM 12/21/04, you wrote: Howdy-- I'm playing with different IAX softphones. I've got DIAX and IAXPHONE on a windows (XP) machine on my network, and I'm running asterisk on a newly loaded Fedora Core 3 machine. I set up a separate IAX account for each phone. I was EXPECTING them to each register seperately with asterisk But I am swiftly finding out, that ONLY one registers. The first one to start running, gets to register with Asterisk, and the other is left out in the cold. Am I seeing things right? Is this the way it should be, or is something wrong? Many thanks! -- Steve Murphy [EMAIL PROTECTED] we have a similar issue with SIP phones and ATA's but differs slightly in that the devices all fail to register after a dozen hours or so I originally thought it was the mysql realtime config but have found it to be an issue with or without sql involved we have an FC1 system with Asterisk CVS-HEAD-12/13/04-07:50:27, Greg ___ 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 seeding vs registration
does anybody have an idea what the difference and significance of sip seeding and registration is. g Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ 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] Poor Grammar or is this a bug
from the asterisk messages log: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25' the only place I can see extension 40852 linked to the ip is in the phone's configuration. sip.conf [40852] ;for a grandstream bt100 musicclass=homeline pedantic=yes accountcode = 40852 amaflags = billing ;callgroup = callerid = 40852 40852 canreinvite = no context = 40852 ;defaultip = dtmfmode = info ;fromuser = ;fromdomain = host = dynamic ;incominglimit = ;outgoinglimit = ;insecure = ;language = mailbox = [EMAIL PROTECTED] ;md5secret = nat = no ;permit = ;deny = ;pickupgroup = port = 5060 qualify = no ;restrictcid = ;rtptimeout = ;rtpholdtimeout = secret = xxx type = friend username = 40852 Regards Greg Cirino ___ ___ 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] Poor Grammar or is this a bug
At 02:46 PM 12/21/04, you wrote: Greg - Cirelle Enterprises wrote: from the asterisk messages log: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.25' the only place I can see extension 40852 linked to the ip is in the phone's configuration. pedantic=yes Take out pedantic=yes and see if it makes any difference. ___ it makes no difference whether it is there or not. 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] Budgetone is not registering
At 03:37 PM 12/21/04, you wrote: Hi again. I cant get my Budgetone registered in Asterisk, and I cant find what's wrong... uff. This is my config: This fragment is from my sip.conf: [12345] type=user user=12345 username=12345 secret=12345 authuser=12345 qualify=1000 nat=no host=dynamic dtmfmode=rfc2833 reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw context=sip_default And this is from my Budgetone configuration: SIP Server: 192.168.1.175 -- asterisk is in my LAN SIP User ID: 12345 Authenticate ID: 12345 Authenticate Password: 12345 And this is the message repeated in the asterisk console: Dec 21 21:36:15 NOTICE[3024]: chan_sip.c:7531 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '192.168.1.176' Can you please help? What is that user=phone about? Thanks a lot in advance, RODOLFO this is a major bug on my system looks like asterisk wants to register nobody seems to know what is up This does it regularly on the grandstream phones and occassionally on the 2 sipura 2k units we have this is an asterisk issue, from what I can tell. no messages anywhere indicating failure, except if this continues for many times, asterisk dies with no warning or message just poof! Greg ___ 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 Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ 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 prevent res_odbc from loading
I am trying to resolve a problem where grandstream phones (only) fail to register after a period of time. I have a mysql realtime setup that appears to work, but fails for no reason. before i classify the realtime system as unusable, I am trying to isolate the problem. One thing I have noticed is the odbc driver continues to load even though I have removed (from what I can tell) references to the res_odbc system. i have deleted the res_odbc.conf file is there anything else I do want to keep the mysql functionality, just get rid of the odbc stuff. Greg Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ 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] Realtime voicemail failure
having set up mysql per instructions for the voicemail system in realtime, we have noticed, email notification has stopped on receipt of voicemail. this works fine on conf file setup, not under realtime. Regards Greg Cirino ___ ___ 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] grandstream MWI?
At 09:46 AM 12/20/04, you wrote: Actually, I got the display flashing when I have a new message. But it is possible to get the Grandstream's Message button working? My goal is to pickup earphone and press Message button to retrieve my messages. Thanks. update your firmware past 1.0.5.16 and put your voicemail address in the bt configuration in the voicmail item. ___ 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 Fails To Start on Reboot Mysql
Apparently asterisk cannot reboot gracefully (unattended) when using realtime MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. WARNING[3763]: MySQL RealTime: Couldn't establish connection. Check debug. Since asterisk starts before mysql it will never start any workaround for this? g Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ 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] Asterisk Fails To Start on Reboot Mysql
At 01:06 PM 12/20/04, you wrote: Even if MySQL RealTime fails to connect to the database server on Asterisk startup, Asterisk will continue to load. I just tested successfully tested this. MySQL RealTime will try to re-connect upon any further RealTime code execution. -Matthew my installation of FC1 i could not connect with asterisk -rvv this persisted until i did an /etc/init.d/asterisk restart then it would fail, then start, and the phones would work as well as connecting with asterisk -r the mysql was starting S78 and asterisk was starting S60 so i changed asterisk to S79 and all is well. The reason I am going through all this is to find out why registration of our 2 grandstream phones fail after a bunch of hours, when using the realtime setup. not a problem when using the .conf files. greg ___ 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] Realtime voicemail failure
At 10:57 AM 12/20/04, you wrote: Have you looked at the debug logs to see if there is any SQL errors? -Matthew I realized the problem, was using a semi colon to separate addresses to send the VM email notification from one extension to 2 addresses. I solved that issue by using a forward address which contains several email addresses, keeping the single address in the email field for notification ex. [EMAIL PROTECTED] points to 2 real email addresses Greg ___ 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] Quick questions ( maybe a little confidence building too )
At 03:50 PM 12/20/04, you wrote: Matt Riddell wrote: Sean Kennedy wrote: Second thing is this: My office is scouting out VoIP solutions, and I have suggested an asterisk solution. We will be getting our voice lines over 9 channels of a t1. I feel comfortable enough with asterisk to set this up, but I am concerned about the Wildcard T100P's sound quality. I First off the T100P is a T1 card, not a single FXO card like the X100P. got the cheapy X100P clone, and it does some weird things ( example: Remote caller hangs up, and the card doesn't detect it. Random hang ups, ect...While I haven't gotten obsesive with the config files, I have Card doesn't detect hangup: make sure you have busycount=8 and busydetect=yes Random Hangups: make sure you don't have callprogress=yes or busycount=lower than 8 done some work trying to eliminate the problems. No luck yet ), and while I'm guessing that's probably due to this being a clone and maybe my config file, I am hoping that people with the T100Ps will give me a first hand accounting on these cards before I commit to this project. I use the T100P card here for connection to a channel bank full of FXS. The sound quality is perfect. I have used multiple cards (X100P, TDM400P etc) for FXO termination and they have all worked perfectly for me (I don't even get any echo). Let us know if those things above don't fix your problems. Also, you didn't say what country you are in, there are some additional changes depending on what country (outside of the USA) you are in. Could I ask how you've connected the t1s? I'm going to be getting a non-pri t1 ( 9 channels of voice, the rest off ). I assume I'll just get an rj45(ish) plug to plug into the back of the card and I'll specify my settings in /etc/zaptel.conf. Does this sound about right? Thanks again! Sean this does not work with non pri t1's according to digium ___ 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] Grandstream and CallerID
At 04:42 PM 12/20/04, you wrote:
I went into the phone and made sure that 'User ID is phone number' was set
to 'No' and made sure that Fromuser=ext# was not present in the sip.conf
file. When a call comes in, the log shows the incoming number but the phone
still reads the extension number. I also have the sip.conf dtmfmode=inband
and the SIP phone set to in-audio. Maybe this is a problem with the
firmware version (1.0.5.16) or my phone is broken.
-Dave
did you try this in your extensions.conf
...
exten = context,6,NoOp(${CALLERID})
exten = context,7,Dial(SIP/${Ext}SIP/${Ext2},15,Ttr)
...
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Re: [Asterisk-Users] [Asterisk-Dev] RE: [Asterisk-biz] Asterisk training andcertification :: AstriconTraining
Get out your wallets boys if you want to get sucked in by this line Microsoft did it and sucked in a bunch. that certification and a 3.50 will get you a coffee at starbucks. IT jobs are in the dumper anyway, so again why? do you honestly think you are going to be asked for your asterisk certification when you go for a job Maybe if you want to work at digium. I think they (digium) should spend more time on making cards that work, read truth in advertising, not stretch the truth in advertising, and support the things they advertise, let's see I think not supporting hdlc after they say the card supports is comes to mind. this one's free ___ 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] what the heck? codec_gsm.c:135 gsmtolin_framein: Huh?
I park a call and instead of the parked extension being returned, I get silence and the log shows a bunch of the following messages WARNING[26220]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from (null) (320)? what does this mean? BTW these messages are intermittant. sometimes it works fine other times i get the above message Regards Greg Cirino ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] erroneous errors - registration fails for grandstream phones
At 09:38 AM 12/17/04, you wrote: Hi, Look in your sip.conf host=192.168.20.2 and your phone is set to use 192.168.20.25 try to change host directive in sip.conf to host=192.168.20.25 Diego Aguirre Host is set to dynamic host=dynamic type=friend I think this is an issue in the way chan_sip handles things with the grandstream phones. this does not occur with the sipura 2k units or xten softphones on the same network. I have set the dtmfmode to info (instead of inband) to see if there is a difference. This might also be specific to the way realtime handles this. I have not seen this occur with the sip.conf file (no database). Then again, I may not have noticed it happening. it appears to be an inconsistent bug in the the way registration is handled with the GS phones. This settles down with a restart of asterisk, but just happens again later. Operation doesn't seem to be affected, but I can't swear to that either. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bugtracker Karma Hall Of Fame
At 09:01 PM 12/16/04, you wrote: Paul Crick wrote: But seriously, if you think you're owed karma for something and haven't received it, flag it to a bug marshall. I'm not one, I just did the web stuff. funny thing that karma stuff, you are never owed any, you just keep doing good stuff to prevent any bad stuff from hunting you down Regards Greg Cirino ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] erroneous errors - registration fails for grandstream phones
Has anybody seen this behaviour? sip conf is stored in mysql database in 2 tables ast_config for static (general) key/values sip_buddies for sip extension detail. database on the same machine as asterisk Grandstream phones (I happen to have 2) register with asterisk via sip with accounts and passwords successfully for a variable period of time. Then after a while, i get errors which appear to be erroneous since the phones/extensions apparently are working. example of 1 phone, but it happens with both: *** from asterisk CLI -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 2400 Dec 17 08:01:59 NOTICE[22259]: chan_sip.c:7742 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.20.25' -- Saved useragent Grandstream BT100 1.0.5.20 for peer 40852 -- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 2400 The date obviously changes *** from /var/log/asterisk/messages Dec 17 08:01:59 NOTICE[22259]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.20.25' The phones appear to work no traffic on the server 3Ghz P4 512MB RAM 75GB Free Disk Space Regards Greg Cirino ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Boss wants background music!!!!
At 04:48 AM 12/16/04, you wrote: Dear Members, I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. If you don't mind my asking, what application would require this feature? Regards Greg Cirino ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] My Boss wants background music!!!!
At 07:49 AM 12/16/04, you wrote: Thank you, My boss believes that people are more happy when soft music playing in the background. The volume has to be low or even of when the phone rings. If this is coupled to the *, then the volume can automatically switch of or switch low. Therefore the volume can be set as high as the user wants because when the phone rings it switches of. With the existing very rudimentary PA system the volume has to be low all the time because of not being coupled. Thanks for your response, now I have a better understanding. I like the 2 line telephone system idea, but it will probably be much more expensive. I was planning the Grandstream 102 phones. What phones would you propose me to use? Willy The Grandstream phones are priced well. we currently use the bt100's. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg - Cirelle Enterprises Sent: Thursday, December 16, 2004 1:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] My Boss wants background music At 04:48 AM 12/16/04, you wrote: Dear Members, I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. If you don't mind my asking, what application would require this feature? Regards Greg Cirino ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] list broken again?
Sure, why not? You know, like how your PHB emails you to let you know the mail server is down. -- Tracy Reedhttp://copilotcom.com PHB ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: FREE BSD
At 01:53 PM 12/15/04, you wrote: anynody knows if I Can install and run Asterisk under Free BSD? /usr/ports/net/asterisk randy several months ago, the port for asterisk was not working because of a security failure in h323. don't know if it has changed or not or if the security issue in h323 has been resolved. Regards Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and RealTime
At 11:02 AM 12/15/04, you wrote: Your sip_buddies table should have 2 columns, allow and disallow. You should be able to: INSERT INTO sip_buddies (allow,disallow) VALUES (g729;g726;gsm,g711); to give the equiv of: allow=g729 allow=g726 allow=gsm disallow=g711 -Matthew I have the sip in 2 tables, the general section is loaded in the ast_config table while each sip extension is defined in sip_buddies. The allow and disallow statements are in the ast_config table and not in the sip_buddies table. Is this wrong? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] list broken again?
It's been hours since I've seen a post from this list Must be broken again. Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip_buddies mysql table
Some of the others you mentioned, name etc, can be increased. But most of those options that call for 'Yes', 'No' or NULL can all be 1 char wide. -Matthew Thanks Matthew, greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] list broken again?
At 06:14 PM 12/14/04, you wrote: On December 14, 2004 04:11 pm, Greg - Cirelle Enterprises wrote: It's been hours since I've seen a post from this list Must be broken again. So you'll email a broken list to send a message...? :-) -A. (yes I realize I'm replying to it) : just making sure it wasn't my crappy machine disclaimer: Please disregard the previous list broken again message regards Greg ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS zaptel missing files
it appears the cvs for zaptel as of 12/13/04 am is missing at least 1 file -- wcfxs.c greg Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS zaptel missing files
At 08:19 AM 12/13/04, you wrote: On Mon, 2004-12-13 at 08:08 -0500, Greg - Cirelle Enterprises wrote: it appears the cvs for zaptel as of 12/13/04 am is missing at least 1 file -- wcfxs.c How about wctdm.c ? -- Dave Cotton [EMAIL PROTECTED] Not sure what that is supposed to do but it sure don't do the trick out of the box. To get the tdm and t100 cards to light up i have to revert to ver 101, 102, or 103 of zaptel greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL
At 09:32 AM 12/13/04, you wrote: Same here. I've deleted and re-installed asterisk a few times and the RealTime voicemail never works. The best I've gotten is the MySQL query to execute with the wrong context. When I use cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds to download the latest version the res_mysql.conf.sample isn't even there. I made it from scratch but it still doesn't work. If that file isn't there what else is missing? Bill I just found out (on my system), the res_mysql.conf has the local mysql socket setting looking for mysql.sock in /tmp/mysql.sock I did a locate mysql.sock which found the actual location and I put that location in res_mysql.conf in the dbsock parameter and it began working. Hope this helps you Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recommended IP phones and VoIP providers?
At 04:59 PM 12/13/04, you wrote: Can anyone give me some recommendations for IP phones that work well with Asterisk? I'm hoping for something not much more then $100 bux or so. grandstream bt100 will work 100 Also does vonage service work directly through Asterisk or would I have to use their hardware? Or are there any other suggestions for a VoIP provider? vonage requires you to have their device and you need to have an fxo of some sort to work with them, (from what their tech support told me). They are not a pure voip provider - ethernet only requirement. livevoip.com is one and there are others mentioned on voip-info.org Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip_buddies mysql table
Not being an asterisk expert, but having been around the block once or twice when it comes to data and the like, I have made some observations based on the examples given on voip-info.org Sip configs. it appears there is an adjustment to be made in the sip_buddies example table: name Although set to 30 characters, I don't see where it is limited in the text file. In theory, this field could be 250 chars long for those who like to get descriptive in there naming convention. ( I guess this is personal taste). insecure The column insecure is 1 character long. according to a search on the key sip insecure, the 3 values that are allowed are Null (no value) yes and very so it appears that column should be at least 4 chars long . Example (from voip-info) insecure=very insecure=yes ; To match a peer based by IP address only and not peer insecure=very ; To allow registered hosts to call without re-authenticating Typically used to allow incoming calls (e.g. from FWD) while having a type=friend entry defined with username and password. amaflags The column amaflags currently 1 character long should be at least 13 chars long amaflag- Categorization for CDR records. Choices are default, omit, billing, documentation and of course Null though not stated. canreinvite this looks to be 3 chars not 1 (Null, no, yes) context See name above incominglimit/outgoinglimit these have been depreciated and probably should be removed unless there is come Realtime coding that requires these fields to be present. restrictcid currently 1 char long, should be 3 chars for values (Null, no, yes) pickupgroup Since callgroup was set to 30 I just set this value to 30 chars since the example shows the same number of characters. CREATE TABLE `sip_buddies` ( `uniqueid` int(11) NOT NULL auto_increment, `name` varchar(250) NOT NULL default '', `accountcode` varchar(30) default NULL, `amaflags` char(1) default NULL, `callgroup` varchar(30) default NULL, `callerid` varchar(50) default NULL, `canreinvite` char(3) default NULL, `context` varchar(250) default NULL, `defaultip` varchar(15) default NULL, `dtmfmode` varchar(7) default NULL, `fromuser` varchar(50) default NULL, `fromdomain` varchar(31) default NULL, `host` varchar(31) NOT NULL default '', `incominglimit` char(2) default NULL, `outgoinglimit` char(2) default NULL, `insecure` varchar(4) default NULL, `language` char(2) default NULL, `mailbox` varchar(50) default NULL, `md5secret` varchar(32) default NULL, `nat` varchar(5) default NULL, `permit` varchar(95) default NULL, `deny` varchar(95) default NULL, `pickupgroup` varchar(30) default NULL, `port` varchar(5) NOT NULL default '', `qualify` varchar(4) default NULL, `restrictcid` char(3) default NULL, `rtptimeout` char(3) default NULL, `rtpholdtimeout` char(3) default NULL, `secret` varchar(30) default NULL, `type` varchar(6) NOT NULL default '', `username` varchar(30) NOT NULL default '', `allow` varchar(100) default NULL, `disallow` varchar(100) default NULL, `regseconds` int(11) NOT NULL default '0', `ipaddr` varchar(15) NOT NULL default '', PRIMARY KEY (`uniqueid`), UNIQUE KEY `name` (`name`), KEY `name_2` (`name`) ) TYPE=MyISAM AUTO_INCREMENT=1 ; Regards Greg Cirino ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL
At 09:59 AM 12/13/04, you wrote: Get newest CVS. Its in there. Trust me. Oh..be sure your getting asterisk-addons. -Matthew Got it thanks Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL
Just so I understand the data structure and what goes in Static configuration is where you can store regular *.conf files into the database. These configurations are read at Asterisk startup/reload. Some modules may also re-read this info upon their own reload (Ex. sip reload). The table structure ast_config in Realtime Static Holds all the contents of ALL CONF FILES including zapata and cdr_mysql, and res_mysql with the exception of: Non Global SIP elements, Non Global IAX elements, Non Global EXTENSIONS elements with limitations of i and s extensions voicemail And all conf files need to be removed from the /etc/asterisk directory? greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BT100 how to pickup a parked call
Does anyone know why the bt100 cannot park and pickup a parked call? attendant announces the call is parked at extension 701 but the call cannot be retrieved by any other phone. also, the bt100 constantly rings when the phone is hung up after parking. anyone experienced this? using the basic features.conf [general] parkext = 700 ; What ext. to dial to park parkpos = 701-709 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 60 ; Number of seconds a call can be parked for pickupexten = *8; Configure the pickup extension. Default is *8 Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BT100 how to pickup a parked call
At 08:57 AM 12/10/04, you wrote:
On Friday 10 December 2004 13:23, Greg - Cirelle Enterprises wrote:
Does anyone know why the bt100 cannot park and pickup
a parked call?
attendant announces the call is parked at extension 701
but the call cannot be retrieved by any other phone.
also, the bt100 constantly rings when the phone is
hung up after parking.
anyone experienced this?
using the basic features.conf
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-709 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
parkingtime = 60 ; Number of seconds a call can be parked
for
pickupexten = *8; Configure the pickup extension. Default
is *8
Have you got 'include = parkedcalls' in the bt100's context ?
Jon
Yes... see below:
extensions.conf
[40852]
exten = 40852,1,SetVar(Ext=40852)
exten = 40852,2,SetVar(Ext3=40853)
exten = 40852,3,SetVar(Ext4=40854)
exten = 40852,4,SetVar(Ext2=40855)
exten = 40852,5,SetVar(ALERT_INFO=Bellcore-r2)
exten = 40852,6,SetMusicOnHold,homeline
exten = 40852,7,NoOp
exten = 40852,8,Dial(SIP/${Ext}SIP/${Ext2}SIP/${Ext3}SIP/${Ext4},15,tTr)
exten = 40852,9,goto(s-${DIALSTATUS},1 )
; go here for no anwer
exten = s-NOANSWER,1,Goto(s-${DIALSTATUS},2)
exten = s-NOANSWER,2,Background(silence/1)
exten = s-NOANSWER,3,Background(chickensmonkeys)
exten = s-NOANSWER,4,Goto(s-${DIALSTATUS},5)
exten = s-NOANSWER,5,Voicemail(${Ext})
exten = s-NOANSWER,6,Goto(${${Ext}.9)
;go here for a busy line
exten = s-BUSY,1,Goto(s-BUSY,2)
exten = s-BUSY,2,Background(silence/1)
exten = s-BUSY,3,Background(chickensmonkeys)
exten = s-BUSY,4,Goto(s-BUSY,5)
exten = s-BUSY,5,Voicemail(${Ext})
exten = s-BUSY,6,Goto(${Ext},9)
;done
exten = 40852,9,Hangup
;this line has access to the following
;outbound access
include = local-trunks
include = toll-free
include = toll-access
include = parkedcalls
include = default
sip.conf
[40852]
type=friend
context=40852
username=40852
fromuser=40852
callerid=BT100 40852
secret=secret
host=dynamic
musicclass=line1
[EMAIL PROTECTED]; set to extension so vmail indicator works
canreinvite=no
nat=no
qualify=no
dtmfmode=inband; inband for grandstream and sipura, otherwise problems
arise with IVRs requiring DMTF tones,
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RE: [Asterisk-Users] BT-100 Transfer!!
At 07:05 AM 12/10/04, you wrote: greg wrote: Transfer feature is not enabled in 1.0.5.18 for asterisk, it is for Nortel Diego Aguirre wrote: I use atendent transfer in Asterisk!!! ok, the # extension key combo too bad the 3 buttons cannot be programmed to emulate the functions to make it work. i.e. the transfer button to send the key code of the # key, etc... same with conference and flash. Greg At 09:02 AM 12/10/04, you wrote: Its the version of * not the phone downgrade and it will work! I'm using * v 1.0.2 I need do downgrade this?? greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BT-100 Transfer!!
At 09:24 AM 12/10/04, you wrote: No, i don't use the # key... 100 cal to 200 (BT-100), 200 press flash then 200 call to 300.. 200 talk to 300 and press transfer key (or hangup), now 100 talk to 300. the same is useful for Handytone ATA 286... sorry my english, this is not my language... Diego Aguirre Operações Internet - ramal 2563 Embratel - RJ Thank you, your english is fine, If I tried to speak spanish, it would probably look like i was talking about automobiles and notepads ... not asterisk. regards Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BT-100 Transfer!!
At 07:05 AM 12/10/04, you wrote: greg wrote: Transfer feature is not enabled in 1.0.5.18 for asterisk, it is for Nortel Diego Aguirre wrote: I use atendent transfer in Asterisk!!! ok, the # extension key combo too bad the 3 buttons cannot be programmed to emulate the functions to make it work. i.e. the transfer button to send the key code of the # key, etc... same with conference and flash. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BT-100 Transfer!!
At 09:09 AM 12/10/04, you wrote: On Friday 10 December 2004 13:53, Greg - Cirelle Enterprises wrote: At 07:05 AM 12/10/04, you wrote: greg wrote: Transfer feature is not enabled in 1.0.5.18 for asterisk, it is for Nortel Diego Aguirre wrote: I use atendent transfer in Asterisk!!! ok, the # extension key combo too bad the 3 buttons cannot be programmed to emulate the functions to make it work. i.e. the transfer button to send the key code of the # key, etc... same with conference and flash. It does work. Make sure on configuration page, you set send flash event to be NO, then pressing flash will not send any dtmf signal, but try to open another session. A talk to B, B press flash and hear dial tone, B dial C an talk to C. B press transfer to let A talk to C. then B hangs up. B That worked! How did you determine that sequence? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BT100 cannot park a call properly???
bt100 1.0.5.18 (same results with previous versions)
an outside call comes in via fxo to an extension hosted
by a sipura 2k. the call is parked and is able to be
picked up (un parked) on a second extension hosted on
a completely different xipura 2k. the call is then reparked
and an attempt is made to answer with a bt100. the call is
retrieved (un parked) but any further attempts to park
from the bt100 fail and just hang up the caller.
an outside call comes in to an extension using a bt100.
an attempt is made to park the call, the attendant says
701. hanging up the phone disconnects the caller.
the dial command is as such:
exten = 40852,8,Dial(SIP/${Ext}SIP/${Ext2}SIP/${Ext3}SIP/${Ext4},15,tTr)
${Ext} is the bt100
the bt100 is set to INFO and the sip.conf is set to inband (was set to info
with the same results)
The parking has to be done by flashing the hook, the flash conference
and transfer buttons do not work as I would think they would.
anybody have a similar issue.
extensions.conf
[40852]
exten = 40852,1,SetVar(Ext=40852)
exten = 40852,2,SetVar(Ext3=40853)
exten = 40852,3,SetVar(Ext4=40854)
exten = 40852,4,SetVar(Ext2=40855)
exten = 40852,5,SetVar(ALERT_INFO=Bellcore-r2)
exten = 40852,6,SetMusicOnHold,homeline
exten = 40852,7,NoOp
exten = 40852,8,Dial(SIP/${Ext}SIP/${Ext2}SIP/${Ext3}SIP/${Ext4},15,tTr)
exten = 40852,9,goto(s-${DIALSTATUS},1 )
; go here for no anwer
exten = s-NOANSWER,1,Goto(s-${DIALSTATUS},2)
exten = s-NOANSWER,2,Background(silence/1)
exten = s-NOANSWER,3,Background(chickensmonkeys)
exten = s-NOANSWER,4,Goto(s-${DIALSTATUS},5)
exten = s-NOANSWER,5,Voicemail(${Ext})
exten = s-NOANSWER,6,Goto(${${Ext}.9)
;go here for a busy line
exten = s-BUSY,1,Goto(s-BUSY,2)
exten = s-BUSY,2,Background(silence/1)
exten = s-BUSY,3,Background(chickensmonkeys)
exten = s-BUSY,4,Goto(s-BUSY,5)
exten = s-BUSY,5,Voicemail(${Ext})
exten = s-BUSY,6,Goto(${Ext},9)
;done
exten = 40852,9,Hangup
;this line has access to the following
;outbound access
include = local-trunks
include = toll-free
include = toll-access
include = parkedcalls
include = default
sip.conf
[40852]
type=friend
context=40852
username=40852
fromuser=40852
callerid=BT100 40852
secret=secret
host=dynamic
musicclass=line1
[EMAIL PROTECTED]; set to extension so vmail indicator works
canreinvite=no
nat=no
qualify=no
dtmfmode=inband; inband for grandstream and sipura, otherwise problems
arise with IVRs requiring DMTF tones, such as voicemail
features.conf
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
parkingtime = 60 ; Number of seconds a call can be parked for
; (default is 45 seconds)
pickupexten = *8; Configure the pickup extension. Default
is *8
Regards
Greg Cirino
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Re: [Asterisk-Users] BT-100 Transfer!!
At 06:29 PM 12/9/04, you wrote: www.grandstream.com/BETATEST - Original Message - From: Mark Willis [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Thursday, December 09, 2004 11:36 PM Subject: RE: [Asterisk-Users] BT-100 Transfer!! I never could get attended transfer to work with the BT-100 on 1.0.5.16. Where did you get 1.0.5.18? It's not anywhere obvious on Grandstream's web site. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Thursday, December 09, 2004 02:56 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BT-100 Transfer!! You need firmware 1.0.5.16 (Broken message button for voicemail) or 1.0.5.18 (Still in Beta, phone display '403' error about once per hour for 10 seconds or so. In order to use attended transfer you place the caller on hold by pressing the flash button and then dial the third person. Once you hang up the caller is transferred to the third person. Craig Transfer feature is not enabled in 1.0.5.18 for asterisk, it is for Nortel greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] save dialplan missing in 1.0.2??
I seem to be missin the save dialplan command in asterisk 1.0.2, I have been searching for info but all I get is how to use it. Anybody have any info on this? Regards Greg Cirino ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog FXO Woes Continue
analog phone. (Example: some central offices don't like dtmf tones within xxx milliseconds after going off-hook. You'll get wrong numbers, etc. Insert the 'w' option in your Dial statement to delay those dtmf tones a little bit.) To be a little sneaky, We had one line, it happened to be a business line, that required putting a w before the number in the dial statement in order for it to work. Fixed our problem, Regards Greg Cirino ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] just testing please ignore
just testing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone 100 Caller ID
At 06:24 PM 12/4/04, you wrote:
Greg - Cirelle Enterprises wrote:
Hi,
Is there an * configuration that will allow the BT100 to
display the numeric callerid instead of the broken
text?
exten = extension,priority,SetCIDNum(${EXTEN})
Doug
Thanks Doug, will try that
Greg
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[Asterisk-Users] Budgetone 100 Caller ID
Hi, Is there an * configuration that will allow the BT100 to display the numeric callerid instead of the broken text? Regards Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] budge tone 100 caller id
Does anybody have the settings to allow the BT 100 to properly display the caller id text or replace the text with the numeric value? Regards Greg Cirino ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Experiences with Termination Providers?
At 02:07 AM 11/28/04, you wrote: I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two of them to be offer a quality service with most of the features I want but horrible customer service/support and response times to my questions etc. The other two seem to respond quickly and have great customer service but have awful connections to the web and basically unusable services. Can someone recommend a termination partner for our VOIP Venture that can provide reliable services, good features/DID's and GOOD customer service? Have you had experience with livevoip? I saw their rates 1 - 10 minutes etc but their TOS says only residential... very little market there But I tried emailing them with regards to reselling their service, so far no response. Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
