[asterisk-users] Recommend some good Click 2 Dial Application
Hello All, Can anyone please recommend me some good Click 2 Dial application ? We need to dial using Microsoft Outlook Business Contact Manager. Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to load module chan_zap.so
I am having trouble with chan_zap.so not loading. When I load it from
modules.conf, Asterisk bails out without any error message. When I
load it from the console, it just says Unable to load module
chan_zap.so no matter what verbose level I am using.
dmesg says:
Zaptel Version: 1.4.4
Zaptel Echo Canceller: MG2
Freshmaker version: 73
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXO (FCC mode)
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
Registered tone zone 1 (Australia)
lsmod says:
wctdm 30912 0
wcfxo 9344 0
zaptel180388 2 wctdm,wcfxo
ztcfg -vv says:
Zaptel Version: 1.4.4
Echo Canceller: MG2
Configuration
==
Channel map:
Channel 01: FXO Loopstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
4 channels to configure.
cat /proc/zaptel/* says:
Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1
1 WCTDM/0/0 FXOLS
2 WCTDM/0/1 FXSKS
3 WCTDM/0/2 FXSKS
4 WCTDM/0/3 FXSKS
/etc/zaptel.conf is:
# Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1
fxols=1
fxsks=2
fxsks=3
fxsks=4
# Global data
loadzone= au
defaultzone = au
I have Googled for help but not found anything. Does anyone have any
suggestions?
TIA
--
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Internet and Open Source lawyer, IT consultant, actor
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Re: [asterisk-users] Unable to load module chan_zap.so
On Mon, 14 Apr 2008, Jeremy Malcolm wrote: I am having trouble with chan_zap.so not loading. When I load it from modules.conf, Asterisk bails out without any error message. When I load it from the console, it just says Unable to load module chan_zap.so no matter what verbose level I am using. dmesg says: Zaptel Version: 1.4.4 Zaptel Echo Canceller: MG2 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Registered tone zone 1 (Australia) lsmod says: wctdm 30912 0 wcfxo 9344 0 zaptel180388 2 wctdm,wcfxo ztcfg -vv says: Zaptel Version: 1.4.4 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXO Loopstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. cat /proc/zaptel/* says: Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 FXOLS 2 WCTDM/0/1 FXSKS 3 WCTDM/0/2 FXSKS 4 WCTDM/0/3 FXSKS /etc/zaptel.conf is: # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 fxols=1 fxsks=2 fxsks=3 fxsks=4 # Global data loadzone = au defaultzone = au Just off hand - isn't that backwards? Isn't it global first before the channels??? Right in the file it says: # Now apply the configuration to the specified channels: # # # We are all done with our channel parameters, so now we specify what # # channels they apply to channels=1-4 What least that is alway what I have done... Brett ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend some good Click 2 Dial Application
On Mon, 14 Apr 2008, Kashif Naeem wrote: Hello All, Can anyone please recommend me some good Click 2 Dial application ? We need to dial using Microsoft Outlook Business Contact Manager. Not used it myself, (Microsoft? Outlook? What that then!) but a couple of my clients are using Snap a number: http://www.snapanumber.com/ Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to load module chan_zap.so
Make sure /usr/lib/asterisk/modules/chan_zap.so is on your system.
If not, my best guess is you compiled asterisk before zaptel.
You'll need to recompile asterisk with the zaptel channeldriver enabled.
Check with: make menuselect
On 17:02, Mon 14 Apr 08, Jeremy Malcolm wrote:
I am having trouble with chan_zap.so not loading. When I load it from
modules.conf, Asterisk bails out without any error message. When I
load it from the console, it just says Unable to load module
chan_zap.so no matter what verbose level I am using.
dmesg says:
Zaptel Version: 1.4.4
Zaptel Echo Canceller: MG2
Freshmaker version: 73
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXO (FCC mode)
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
Registered tone zone 1 (Australia)
lsmod says:
wctdm 30912 0
wcfxo 9344 0
zaptel180388 2 wctdm,wcfxo
ztcfg -vv says:
Zaptel Version: 1.4.4
Echo Canceller: MG2
Configuration
==
Channel map:
Channel 01: FXO Loopstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
4 channels to configure.
cat /proc/zaptel/* says:
Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1
1 WCTDM/0/0 FXOLS
2 WCTDM/0/1 FXSKS
3 WCTDM/0/2 FXSKS
4 WCTDM/0/3 FXSKS
/etc/zaptel.conf is:
# Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1
fxols=1
fxsks=2
fxsks=3
fxsks=4
# Global data
loadzone = au
defaultzone = au
I have Googled for help but not found anything. Does anyone have any
suggestions?
TIA
--
Jeremy Malcolm LLB (Hons) B Com
Internet and Open Source lawyer, IT consultant, actor
host -t NAPTR 1.0.8.0.3.1.2.9.8.1.6.e164.org|awk -F! '{print $3}'
Luxury Perth apartment for sale!
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GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer aficionados are both called users?
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Re: [asterisk-users] Is Asterisk really good??
why yes, my rsync does that just fine, you must not be running
the latest version
Steve Edwards wrote:
On Mon, 14 Apr 2008, Bernd Felsche wrote:
Steve Edwards [EMAIL PROTECTED] wrote:
I'm mainly interested in consistency in configuration. The method has
to be sophisticated enough to handle this box has 2 Ethernet interfaces
so I should configure OpenSER and Asterisk to listen to both IP addresses
on ports 5060 and 5061 respectively. This would preclude rsync.
Why do you think that that would preclude rsync?
Well, it may be based on my ignorance :)
Can rsync mung a stanza from iax.conf like:
[general]
disallow = all
allow = ulaw
mailboxdetail = no
notransfer = yes
port= 5036
register= ${HOSTNAME}:[EMAIL PROTECTED]
trunk = no
and insert the appropriate values?
Can rsync create /etc/sysconfig/openser like:
# Created by ./host-setup.sh on 2008-04-12 17:55:03
OPTIONS=
OPTIONS=$OPTIONS -l a.b.c.d:5060
OPTIONS=$OPTIONS -l a.b.c.e:5060
# (end of /etc/sysconfig/openser)
where a.b.c.d and a.b.c.e are the IP address of eth0 and eth1? (And the
3rd line would only be created if there are 2 interfaces.)
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Recommend some good Click 2 Dial Application
Please check: http://www.voip-info.org/wiki/view/Asterisk+TAPI Configure a TAPI source in windows and Outlook can do click to dial natively using the TAPI Driver. On Mon, 2008-04-14 at 14:24 +0500, Kashif Naeem wrote: Hello All, Can anyone please recommend me some good Click 2 Dial application ? We need to dial using Microsoft Outlook Business Contact Manager. Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to load module chan_zap.so
On Mon, Apr 14, 2008 at 04:43:20AM -0500, Brett Crapser wrote: On Mon, 14 Apr 2008, Jeremy Malcolm wrote: /etc/zaptel.conf is: # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 fxols=1 fxsks=2 fxsks=3 fxsks=4 # Global data loadzone= au defaultzone = au Just off hand - isn't that backwards? Isn't it global first before the channels??? The order doesn't really matter (unless you try to redefine the same thing...). But all of that is Zaptel-level stuff. IT seems to be well defined. In the Asterisk CLI, what happens when you run: module unload chan_zap.so module load chan_aap.so -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cdr_custom outout to serial port
No problem. The program is in Windows. Contact me off line to make arrangements to send you the installation files. C. Savinovich Long ago, I wrote a nice program that reads CDR output from any legacy PBX via the serial port. Not much in use lately, but I will be happy to furbish it with mysql output to anyone who asks. Yes, please. What OS does it run under? Thanks! Doug ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend some good Click 2 Dial Application
I se Snapanumber bt with outlook not BCM but assume it will work the same. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kashif Naeem Sent: Monday, 14 April 2008 5:25 AM To: [EMAIL PROTECTED] Subject: [asterisk-users] Recommend some good Click 2 Dial Application Hello All, Can anyone please recommend me some good Click 2 Dial application ? We need to dial using Microsoft Outlook Business Contact Manager. Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com http://www.haditelecom.com/ Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend some good Click 2 Dial Application
On 14/04/2008, Gordon Henderson [EMAIL PROTECTED] wrote: Not used it myself, (Microsoft? Outlook? What that then!) but a couple of my clients are using Snap a number: http://www.snapanumber.com/ Gordon Oh, that _is_ nice :) Thanks for the pointer! Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Similar option as promiscredir to use in transfer (REFER)
Thanks for the reply, Johansson. Sorry if my question was not very clear... What I need is that asterisk accepts a REFER command from the client, sending the call to a non local domain. The scenario is this: I receive a call from PSTN and dial a sip address that contains one of my applications (running in a separate machine). This application receives input from the user and then transfers the call to another application (in a third machine). The call from PSTN is going to be in asterisk (that got the call in first place) all the time, just the other end will change depending on user input. Bellow is a sip debug from this operation. Asterisk is running in 201.73.67.5:5060 and my first application is at [EMAIL PROTECTED]:5080. This application then tries to transfer the call to a second application located at [EMAIL PROTECTED]:5070, but asterisk ignores the part after the @ from the uri and tries sending the call to the extension 5070 in the context from-sip-external. I had a similar problem with redirects (302), but I solved it using the option promiscredir=yes inside sip.conf. I've already tried setting the option domain= in sip.conf but that didn't help... -- SIP read from 201.73.67.7:5080: REFER sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 201.73.67.7:5080;rport;branch=z9hG4bKPj3r0RqvljQLyTKpBVXgbhce5dADV20tVx Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA To: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 Contact: sip:201.73.67.7:5080 Call-ID: [EMAIL PROTECTED] CSeq: 15651 REFER Event: refer Expires: 300 Accept: message/sipfrag;version=2.0 Allow-Events: presence, refer Refer-To: sip:[EMAIL PROTECTED]:5070 Referred-By: sip:[EMAIL PROTECTED] Content-Length: 0 --- (15 headers 0 lines) --- Transfer to 5070 in from-sip-external Transfer from 0778 in from-sip-external Transmitting (no NAT) to 201.73.67.7:5080: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 201.73.67.7:5080;branch=z9hG4bKPj3r0RqvljQLyTKpBVXgbhce5dADV20tVx;received=201.73.67.7;rport=5080 From: sip:[EMAIL PROTECTED];tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA To: 3130296800 sip:[EMAIL PROTECTED];tag=as26b5df58 Call-ID: [EMAIL PROTECTED] CSeq: 15651 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing Thiago 13 apr 2008 kl. 17.46 skrev [EMAIL PROTECTED]: I made a similar question in a previous thread, but there was no answer, so I think I was not very clear making the question. What I need is some configuration that works like promiscredir=yes in sip.conf that enables me to do the same thing with transfer (REFER), letting me transfer a sip call to a non local sip address. I'm still not really sure what you ask for, but I'll give it a try. The transfer() dialplan application supports generating a REFER from Asterisk to the client. If the call is not answered, it will send 302, if the call is in UP state (answered), Asterisk will send a REFER. Try it. Best regards, /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk SIP Masterclass, Orlando, Florida Next week * A few seats left - register today! Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend some good Click 2 Dial Application
Kashif,
outcall.sourceforge.net
support is at 350 EUR / year
contact me offline if required : steve 'at' bicomsystems {dot} com
Steve
- Original Message -
From: Kashif Naeem
To: [EMAIL PROTECTED]
Sent: Monday, April 14, 2008 11:24 AM
Subject: [asterisk-users] Recommend some good Click 2 Dial Application
Hello All,
Can anyone please recommend me some good Click 2 Dial application ? We need
to dial using Microsoft Outlook Business Contact Manager.
Regards,
--
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
Email: [EMAIL PROTECTED]
MSN: [EMAIL PROTECTED]
Gmail: [EMAIL PROTECTED]
Skype: kashif.naeem
302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.
--
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Re: [asterisk-users] X100M never goes on-hook state
On Fri, Apr 11, 2008 at 11:43:19PM -0300, Marlon Dutra wrote: If I put an DSL filter in series with the line and the card, IT WORKS PERFECTLY!!! The filter imposes 25 ohms over the circuit. Maybe that's causing the card to work. When I put the filter and the ammeter in series, I get zero amper when on-hook and 26 mA when off-hook, that's the expected behaviour. It's possible the 'line relay' on that card is not a physical relay, but electronic, and that its sensitive to too much loop current -- and the DSL filter drops the current far enough for that 'relay' not to pull in spuriously. Telecom guy Mike Sandman has a paper on loop current on his website: http://sandman.com/loopcur.html and, come to that, lots of *really* cool stuff for sale as well; if you've never looked at his site, do. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and the Mitel SX 200 integration
On Sun, Apr 13, 2008 at 11:30:26PM -0500, Doug wrote: At 21:08 4/11/2008, Alexander Lopez wrote: Jorge is correct you will not get the information need via FXO/FXS unless you program the Mitel to do DTMF inband. It is possible but a cludge of a fix at best. We have successfully integrated several Mitel SX200 and SX2000 switches via the PRI (preferred) or T1 using EM_Wink (works but you have delays while waiting for the winks. (wink, wink :-) ). The Mitel is rock-solid Until the floppy disk dies. Then you have a huge doorstop and no phone system. The floppy drives aren't easily replaceable. Hint to the OP: this translates as go start searching for a spare floppy disk drive *NOW*! :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good??
On Sun, Apr 13, 2008 at 04:39:39PM -0700, Steve Edwards wrote: I'm in the midst of rearranging things (which are 2 to 3 times as large as they were then); I'll update that once I'm done. Double-plus cool. I'd be interested in sections like Rolling out a new server or How we maintain all the little configuration files without losing our sanity. I smell a magazine article. :-) That works, but I'm impatient. I'm up for peer review before publication. Understood. Real Magazines tend to be picky about first pub, though. The answer to the second question is likely going to become rsync or cfengine, but I haven't gotten that far yet... and we don't change them all that much anyway. VICIdial has *lots* of knobs. I'm mainly interested in consistency in configuration. The method has to be sophisticated enough to handle this box has 2 Ethernet interfaces so I should configure OpenSER and Asterisk to listen to both IP addresses on ports 5060 and 5061 respectively. This would preclude rsync. True. That's why I was leaning towards cfengine, which I gather is tuned for that sort of thing. I currently do it with shell scripts but I'm looking for something a bit more sophisticated. Puppet (http://reductivelabs.com/trac/puppet/wiki/AboutPuppet) was suggested during the Friday morning VOIP Users Conference. It's open source and written in Ruby. I just feel a bit silly installing yet another language just to support a support tool. Indeed. The shell script approach has the advantage of light weight. I do a minimal Centos 5 install and wget a single script which does everything -- configures the network, installs packages (OpenSER, Asterisk, Zaptel, Libpri, MySQL), adds users, and configures everything from services to timezone. I may stick with it, but it's getting a bit combersome and am interested in what has worked for others. Noted. Our solution may not help you all that much; I gather that with the exception of one small chunk of one file, all our boxen are configured exactly the same. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good??
On Mon, Apr 14, 2008 at 06:02:26AM -0400, Al Baker wrote: Steve Edwards [EMAIL PROTECTED] wrote: Well, it may be based on my ignorance :) Can rsync mung a stanza from iax.conf like: why yes, my rsync does that just fine, you must not be running the latest version April Fools Day was 2 weeks ago, Al. :-) rsync, to the best of my knowledge and belief, does not make any sort of changes to the files it pushes -- it merely pushes the changes you make. So if you need a given file, let's say extensions.conf, to contain different things on different machines, then using rsync to push it from one master will in fact probably not work the way you want it to. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend some good Click 2 Dial Application
siptapi Kashif Naeem schrieb: Hello All, Can anyone please recommend me some good Click 2 Dial application ? We need to dial using Microsoft Outlook Business Contact Manager. Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com http://www.haditelecom.com/ Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] MSN: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf wont load completely
I have seen this issue on both 1.2 and 1.4, was not able to reproduce to find a cause or bug. I have seen this after power failure boot up. show sip peer command shows most of peers, except one or two (in my cases trunk) . if i issue a sip reload command, it will show all of them. I can write a script to reload asterisk after a minute of boot up but i wanted to see if anyone else has seen this issue or has any thoughts. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good??
On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sun, Apr 13, 2008 at 04:39:39PM -0700, Steve Edwards wrote: The shell script approach has the advantage of light weight. I do a minimal Centos 5 install and wget a single script which does everything -- configures the network, installs packages (OpenSER, Asterisk, Zaptel, Libpri, MySQL), adds users, and configures everything from services to timezone. I may stick with it, but it's getting a bit combersome and am interested in what has worked for others. Noted. Our solution may not help you all that much; I gather that with the exception of one small chunk of one file, all our boxen are configured exactly the same. It is actually two small chunks of two small files in Asterisk and one line in the vicidial conf file, and that's about it for unique server configurations, everything else is pretty much the same. We did recently add a custom backup utility to our SVN for VICIDIAL(AST_backup.pl) that will backup all conf files, agi, sound and other files(optionally web files and mysql DB and my.cnf backup) and tar/gz them then send to FTP server. This has worked well for multi-server backups for a couple of our clients so far and it will be included with the next release of VICIDIAL. The idea behind the script is to create a very simple hot-spare solution where all you have to do to replace a running machine is change the IP address of the spare server and un-tar/gz the file on a base-installed system and it will take the place of the failed machine within minutes. We haven't had to use it in production in this capacity yet, but it has worked in testing. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good??
Steve, Is this 'shell script' on the public domain? As it sounds really useful. :) Mark. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: April 13, 2008 7:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is Asterisk really good?? On Sun, 13 Apr 2008, Jay R. Ashworth wrote: On Sat, Apr 12, 2008 at 01:40:44PM -0700, Steve Edwards wrote: On Sat, 12 Apr 2008, Jay R. Ashworth wrote: On Fri, Apr 11, 2008 at 06:11:45PM -0700, Eugen Soare wrote: That was cool! thanks for the pdf. I'm in the midst of rearranging things (which are 2 to 3 times as large as they were then); I'll update that once I'm done. Double-plus cool. I'd be interested in sections like Rolling out a new server or How we maintain all the little configuration files without losing our sanity. I smell a magazine article. :-) That works, but I'm impatient. I'm up for peer review before publication. The answer to the second question is likely going to become rsync or cfengine, but I haven't gotten that far yet... and we don't change them all that much anyway. VICIdial has *lots* of knobs. I'm mainly interested in consistency in configuration. The method has to be sophisticated enough to handle this box has 2 Ethernet interfaces so I should configure OpenSER and Asterisk to listen to both IP addresses on ports 5060 and 5061 respectively. This would preclude rsync. I currently do it with shell scripts but I'm looking for something a bit more sophisticated. Puppet (http://reductivelabs.com/trac/puppet/wiki/AboutPuppet) was suggested during the Friday morning VOIP Users Conference. It's open source and written in Ruby. I just feel a bit silly installing yet another language just to support a support tool. The shell script approach has the advantage of light weight. I do a minimal Centos 5 install and wget a single script which does everything -- configures the network, installs packages (OpenSER, Asterisk, Zaptel, Libpri, MySQL), adds users, and configures everything from services to timezone. I may stick with it, but it's getting a bit combersome and am interested in what has worked for others. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] do cards just instantly go bad
Hi - Been using a TE205P for a number of months - no issues. Today I was talking to someone and I heard click No more phone service. I still have data service on this T1 line. (partial phone) zttool reports the SPAN as OK. calls are not coming in or going out. Does a card just go bad like that? How can I tell if the card is bad? I was expecting/hoping to see something other than OK on zttool. Its reporting OK but still no calls. I made no changes to anything in weeks. I presume there is a chance the carrier (nuvox) is having issues but how can I make sure there isnt something on my end? My host box is centos 5.1, asterisk 1.4.11 libpri -1.4.1 and zaptel 1.4.5.1 Again, been running fine for MONTHS. I did stop asterisk, service zaptel restart, and start asterisk again, same thing though no calls. THanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] do cards just instantly go bad
Quoting Jerry Geis [EMAIL PROTECTED]: you might try an actual power cycle in case some circuit actually needs a hard reset, but other than that, anything is possible, it could have failed. Hi - Been using a TE205P for a number of months - no issues. Today I was talking to someone and I heard click No more phone service. I still have data service on this T1 line. (partial phone) zttool reports the SPAN as OK. calls are not coming in or going out. Does a card just go bad like that? How can I tell if the card is bad? I was expecting/hoping to see something other than OK on zttool. Its reporting OK but still no calls. I made no changes to anything in weeks. I presume there is a chance the carrier (nuvox) is having issues but how can I make sure there isnt something on my end? My host box is centos 5.1, asterisk 1.4.11 libpri -1.4.1 and zaptel 1.4.5.1 Again, been running fine for MONTHS. I did stop asterisk, service zaptel restart, and start asterisk again, same thing though no calls. THanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] do cards just instantly go bad
At 12:52 PM 4/14/2008, Jerry Geis wrote: Hi - Been using a TE205P for a number of months - no issues. Today I was talking to someone and I heard click No more phone service. I still have data service on this T1 line. (partial phone) zttool reports the SPAN as OK. calls are not coming in or going out. Does a card just go bad like that? How can I tell if the card is bad? I was expecting/hoping to see something other than OK on zttool. Its reporting OK but still no calls. Probably not the card. More likely the T1 provider. Contact your carrier. Ask them to put a T1 level trouble ticket. Ask that you be on the phone when the tester is bringing the T1 down and up. Quite often, all that is needed is to BOUNCE the circuit from the switch end. ..mike.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100M never goes on-hook state
Hello, On Mon, Apr 14, 2008 at 10:30 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote: It's possible the 'line relay' on that card is not a physical relay, but electronic, and that its sensitive to too much loop current -- and the DSL filter drops the current far enough for that 'relay' not to pull in spuriously. Yes, I was suspecting it was something like that. I bought a 33-ohm resistor (1/2 watt) and it fixed the problem. Both modules are working properly now. According to the Ohm's law, the loop's overall resistence was 188.46 ohms, once I was measuring 4.9V and 26mA in the circuit. By adding 33 ohms, it went to 221.46 ohms, bringing the current down to 22 mA. Telecom guy Mike Sandman has a paper on loop current on his website: Excellent article and telephony stuff. Bookmarked! Thanks for the reply. Have a nice day. -- MARLON DUTRA Propus GnuPG ID: 0x3E2060AC pgp.mit.edu http://www.propus.com.br/ http://hackers.propus.com.br/~marlon/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good??
-Original Message- I'd be interested in sections like Rolling out a new server or How we maintain all the little configuration files without losing our sanity. Hi, I will contribute my 2-cents on how I maintained consistency on a large application with 64 + Asterisks that all had to have the same config and links back to a central DB. Whenever we needed a new machine, we just We had a master source location.with a master image We cloned the hard drive with linux dd copy of master image boot the new machine with this disk assign appropriate IP address perform some sanity checks prior to shipping Send either disk or full machine to remote COLO for physical install. After the machine came on line, it would have enough configuration to join the other members of the farm of asterisks. For intermediate updates, we used SSL-DSA keys between the master master image machine and each of the 64+ remotes. We would wrote our own script and gave it a list of each machine on which to perform the particular steps. When it was launched, we just went out to lunch or home at night while the remotes were updated. This application had as many as 6,000 simultaneous call running and we wrote the scripts such that each remote were placed in a take no calls status by the script so we did not kill any active traffic. We found that no canned package was useful to do this because each maintenance cycle was addressing a different part of the overall configuration and had slightly different commands that were needed. Any good script writer can do the same for what you described. Regards, ..mike.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
I'm glad so much has been sent about on the thread I create (bloated ego head :) ) It has gotten my curiosity up. What is VICIDIAL? Is it Public Domain? Pay for Software? What's it all about? (not looking for all the features, maybe I should put my understanding of it's functions and people can correct me.) It seems to be a software product that can handle call centers, be they in coming our out going calls. Has modules to take credit cards / and is customizable so that added functionality can be written. This is been very interesting! es Matt Florell wrote: On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sun, Apr 13, 2008 at 04:39:39PM -0700, Steve Edwards wrote: The "shell script" approach has the advantage of "light weight." I do a "minimal" Centos 5 install and wget a single script which does everything -- configures the network, installs packages (OpenSER, Asterisk, Zaptel, Libpri, MySQL), adds users, and configures everything from services to timezone. I may stick with it, but it's getting a bit combersome and am interested in what has worked for others. Noted. Our solution may not help you all that much; I gather that with the exception of one small chunk of one file, all our boxen are configured exactly the same. It is actually two small chunks of two small files in Asterisk and one line in the vicidial conf file, and that's about it for unique server configurations, everything else is pretty much the same. We did recently add a custom backup utility to our SVN for VICIDIAL(AST_backup.pl) that will backup all conf files, agi, sound and other files(optionally web files and mysql DB and my.cnf backup) and tar/gz them then send to FTP server. This has worked well for multi-server backups for a couple of our clients so far and it will be included with the next release of VICIDIAL. The idea behind the script is to create a very simple hot-spare solution where all you have to do to replace a running machine is change the IP address of the spare server and un-tar/gz the file on a base-installed system and it will take the place of the failed machine within minutes. We haven't had to use it in production in this capacity yet, but it has worked in testing. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On 4/14/08, Eugen Soare [EMAIL PROTECTED] wrote: I'm glad so much has been sent about on the thread I create (bloated ego head :) ) It has gotten my curiosity up. What is VICIDIAL? Is it Public Domain? Pay for Software? What's it all about? (not looking for all the features, maybe I should put my understanding of it's functions and people can correct me.) It seems to be a software product that can handle call centers, be they in coming our out going calls. Has modules to take credit cards / and is customizable so that added functionality can be written. This is been very interesting! es Hello, VICIDIAL is call center software for Asterisk. It is designed around Asterisk, not compiled into Asterisk. VICIDIAL takes a different approach to the call center application from how Asterisk inbound Queues/Agents does it, since it uses Meetme rooms to house the agents allowing for more consistency across versions of Asterisk as well as a lot more flexibility in terms of features. The agent web interface is an AJAX application that will run well in most modern web browsers on computers with a PIII 500MHz or higher. With VICIDIAL you can do inbound/outbound/blended call handling and there are all sorts of features for call handling and agent functions. The latest VICIDIAL release is GPLv2, but for future major releases we are moving to the AGPLv2. VICIDIAL is free as in cost and speech. There are currently well over 400 companies using VICIDIAL in over 40 countries(unconfirmed survey results show over 700 company users, with over 17,000 seats total) and the agent interface is available in 9 languages. Hope that helps. For more info go to: http://astguiclient.sourceforge.net/vicidial.html MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good??
On Mon, Apr 14, 2008 at 11:08:21AM -0400, Matt Florell wrote: The idea behind the script is to create a very simple hot-spare solution where all you have to do to replace a running machine is change the IP address of the spare server and un-tar/gz the file on a base-installed system and it will take the place of the failed machine within minutes. We haven't had to use it in production in this capacity yet, but it has worked in testing. I may have just had an even better idea, crribbing from how DSL does overlays. Put said config file on an FTP server on the cluster... named after the IP of the box, the way Bootp files are. Put innocuous files on the box as a base install, and have it prospect the FTP server on boot for an overlay. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On Mon, Apr 14, 2008 at 01:54:22PM -0400, Matt Florell wrote: With VICIDIAL you can do inbound/outbound/blended call handling and there are all sorts of features for call handling and agent functions. The latest VICIDIAL release is GPLv2, but for future major releases we are moving to the AGPLv2. VICIDIAL is free as in cost and speech. I noticed you had gone Affero. Could you expand on that decision, if you have a moment? What's the difference between the two licenses, did you consider GPLv3, and what's your situation on contributed code? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] do cards just instantly go bad
On Mon, Apr 14, 2008 at 01:04:52PM -0400, Jon Pounder wrote: you might try an actual power cycle in case some circuit actually needs a hard reset, but other than that, anything is possible, it could have failed. This is a good point to remember: shutting down modern motherboards *does not* remove all voltage from all lines of the bus connector... and that can keep cards from resetting. As I discovered this week, even a hard power switch on the back isn't always good enough -- we installed a server with hot-swappable dual power supplies, each with its own rocker switch. Turning off both rocker switches *still* did not cause the wake-on-LAN lights on the Ethernet jack to go out. I had to unplug both power cords to accomplish that. I suspect Underwriters' Laboratories Would Not Be Pleased. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
Matt. Thanks for the reply and Link. That should get me started looking at that. Unfortunately, coming from the Nortel world. It may take some time to get up to speed on things. The hardest part (as I see it) is getting hardware/software instructions on setting up and then maybe connecting to someone elses box to play around with the integration of different sites. This looks like a good Fall/Winter project. Need to remodel the basement now. Anyway, I think that's a little off list. :) oops. It looks like there is a link on the web-page of the link that you sent, that provides a "startup from scratch! COOL! Thanks again. Eugen Matt Florell wrote: On 4/14/08, Eugen Soare [EMAIL PROTECTED] wrote: I'm glad so much has been sent about on the thread I create (bloated ego head :) ) It has gotten my curiosity up. What is VICIDIAL? Is it Public Domain? Pay for Software? What's it all about? (not looking for all the features, maybe I should put my understanding of it's functions and people can correct me.) It seems to be a software product that can handle call centers, be they in coming our out going calls. Has modules to take credit cards / and is customizable so that added functionality can be written. This is been very interesting! es Hello, VICIDIAL is call center software for Asterisk. It is designed around Asterisk, not compiled into Asterisk. VICIDIAL takes a different approach to the call center application from how Asterisk inbound Queues/Agents does it, since it uses Meetme rooms to house the agents allowing for more consistency across versions of Asterisk as well as a lot more flexibility in terms of features. The agent web interface is an AJAX application that will run well in most modern web browsers on computers with a PIII 500MHz or higher. With VICIDIAL you can do inbound/outbound/blended call handling and there are all sorts of features for call handling and agent functions. The latest VICIDIAL release is GPLv2, but for future major releases we are moving to the AGPLv2. VICIDIAL is free as in cost and speech. There are currently well over 400 companies using VICIDIAL in over 40 countries(unconfirmed survey results show over 700 company users, with over 17,000 seats total) and the agent interface is available in 9 languages. Hope that helps. For more info go to: http://astguiclient.sourceforge.net/vicidial.html MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On 4/14/08, Eugen Soare [EMAIL PROTECTED] wrote: Matt. Thanks for the reply and Link. That should get me started looking at that. Unfortunately, coming from the Nortel world. It may take some time to get up to speed on things. The hardest part (as I see it) is getting hardware/software instructions on setting up and then maybe connecting to someone elses box to play around with the integration of different sites. This looks like a good Fall/Winter project. Need to remodel the basement now. Anyway, I think that's a little off list. :) oops. It looks like there is a link on the web-page of the link that you sent, that provides a startup from scratch! COOL! Thanks again. Eugen Ah yes, my monster SCRATCH_INSTALL document :) If you run into any problems, please check out our very active VICIDIAL Forums: http://www.eflo.net/VICIDIALforum/index.php Good luck! MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 01:54:22PM -0400, Matt Florell wrote: With VICIDIAL you can do inbound/outbound/blended call handling and there are all sorts of features for call handling and agent functions. The latest VICIDIAL release is GPLv2, but for future major releases we are moving to the AGPLv2. VICIDIAL is free as in cost and speech. I noticed you had gone Affero. Could you expand on that decision, if you have a moment? What's the difference between the two licenses, did you consider GPLv3, and what's your situation on contributed code? We finally decided we would be going to AGPLv2 for our next major release due to a few hosted service providers out there that were altering the code to VICIDIAL, offering VICIDIAL hosted and not contributing their changes back to the project. And under the GPL they have every right to do this as long as the code is not installed on a client-owned machine or transferred to a client. This is known as the GPL-ASP-loophole. AGPL just closes that loophole and says that any customer of a hosted service like that has the right to the source code too. We have not done enough research on GPLv3 yet to want to move to it, and a lot of other GPLv2 projects are staying put as well for the time being. As for contributed code, we require a statement of this is my code and the project can use it and redistribute it from the author. Nothing very detailed at the moment because there are not many code contributors and the project is entirely GPL-based and is not dual-licensed. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E911 Recommendations?
Anybody have recommendations for a reliable, good valued, E911 provider? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On Mon, Apr 14, 2008 at 02:48:38PM -0400, Matt Florell wrote: Ah yes, my monster SCRATCH_INSTALL document :) And (why the hell *not* stick my neck out :-) I'm planning some work on the wiki to merge that and the newer documentation from SVN into sort of an Administrator's Manual in the next, oh, say, month or two. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
Jay R. Ashworth wrote: On Mon, Apr 14, 2008 at 02:05:18PM -0500, Doug wrote: Anybody have recommendations for a reliable, good valued, E911 provider? Wow. E911 providers are *municipalities*, aren't they? :-) No, they're not. There are service companies specialising in the delivery of 911 calls to the correct PSAP along with maintaining and updating E.911 placement/ALI information. I recommend HBF, myself, but they're rather high-end and enterprise-oriented. Also, a well-known local CLEC I've worked for used Intrado for intra-LATA and off-net customers needing to dial 911 and it seemed to work great. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 02:47:12PM -0400, Matt Florell wrote: On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 01:54:22PM -0400, Matt Florell wrote: With VICIDIAL you can do inbound/outbound/blended call handling and there are all sorts of features for call handling and agent functions. The latest VICIDIAL release is GPLv2, but for future major releases we are moving to the AGPLv2. VICIDIAL is free as in cost and speech. I noticed you had gone Affero. Could you expand on that decision, if you have a moment? What's the difference between the two licenses, did you consider GPLv3, and what's your situation on contributed code? We finally decided we would be going to AGPLv2 for our next major release due to a few hosted service providers out there that were altering the code to VICIDIAL, offering VICIDIAL hosted and not contributing their changes back to the project. And under the GPL they have every right to do this as long as the code is not installed on a client-owned machine or transferred to a client. This is known as the GPL-ASP-loophole. AGPL just closes that loophole and says that any customer of a hosted service like that has the right to the source code too. Ok; that's what I *thought* Affero's change was, but it's kind of hard to tell from the actual license... Yes, we had to read it several times ourselves, the version we have in our SVN trunk is what we settled on since there are several different text formats of the AGPL license floating around. We have not done enough research on GPLv3 yet to want to move to it, and a lot of other GPLv2 projects are staying put as well for the time being. I'm not really fond of it myself. I don't know enough about it at the moment to be fond of it or not myself. As more people move to it and it's provisions are tested I will hopefully be able to move from neutral to one side or the other at some point. As for contributed code, we require a statement of this is my code and the project can use it and redistribute it from the author. Nothing very detailed at the moment because there are not many code contributors and the project is entirely GPL-based and is not dual-licensed. Yeah; I was just worried about someone getting pissy about your relicensing from GPL to AGPL. Not that I expect it or anything... :-) I am fairly surprised that I have not heard a single negative comment about it from any members of our VICIDIAL community or anywhere else. There are actually other web-based projects that are moving to it as well(which is how I originally heard about it) and since it became an official OSI-approved Open Source License along with the special provisions that GPL made allowing for AGPL compatibility, there are more people talking about it in the last few months. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
On Mon, Apr 14, 2008 at 3:11 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 02:05:18PM -0500, Doug wrote: Anybody have recommendations for a reliable, good valued, E911 provider? Wow. E911 providers are *municipalities*, aren't they? :-) Could you vague that up a bit, Doug? (Or should I be able to generalize that phrasing into what you actually mean, if I expect to get along here? :-) Cheers, -- jra Wow, that response was completely unnecessary. I think most people (myself included) know what he meant. To actually answer the question - I know many people who have had good experiences with Dash911, now known as Dash Carrier Services (I believe). Good luck Doug! -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and the Mitel SX 200 integration
John covici wrote: OK, this is exactly what I would like to do, can you either write me on or off list for further details. This would be the first baby step toward the 20th Century!! I'd love some pointers on integrating * with a sx-200. I have a system where a fork lift upgrade is impossible. Ideally as we add new extensions, I'd like them to be in *, and have the mitel system know to route the calls correctly. I have a manual for the sx and a few half baked thoughts (put the pstn on *, and have the mitel system send all unknown numbers to *. Then * can route them properly to the outside world or to the new extensions), but it will be slow going. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
On Mon, Apr 14, 2008 at 02:05:18PM -0500, Doug wrote: Anybody have recommendations for a reliable, good valued, E911 provider? Wow. E911 providers are *municipalities*, aren't they? :-) Could you vague that up a bit, Doug? (Or should I be able to generalize that phrasing into what you actually mean, if I expect to get along here? :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On Mon, Apr 14, 2008 at 02:47:12PM -0400, Matt Florell wrote: On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 01:54:22PM -0400, Matt Florell wrote: With VICIDIAL you can do inbound/outbound/blended call handling and there are all sorts of features for call handling and agent functions. The latest VICIDIAL release is GPLv2, but for future major releases we are moving to the AGPLv2. VICIDIAL is free as in cost and speech. I noticed you had gone Affero. Could you expand on that decision, if you have a moment? What's the difference between the two licenses, did you consider GPLv3, and what's your situation on contributed code? We finally decided we would be going to AGPLv2 for our next major release due to a few hosted service providers out there that were altering the code to VICIDIAL, offering VICIDIAL hosted and not contributing their changes back to the project. And under the GPL they have every right to do this as long as the code is not installed on a client-owned machine or transferred to a client. This is known as the GPL-ASP-loophole. AGPL just closes that loophole and says that any customer of a hosted service like that has the right to the source code too. Ok; that's what I *thought* Affero's change was, but it's kind of hard to tell from the actual license... We have not done enough research on GPLv3 yet to want to move to it, and a lot of other GPLv2 projects are staying put as well for the time being. I'm not really fond of it myself. As for contributed code, we require a statement of this is my code and the project can use it and redistribute it from the author. Nothing very detailed at the moment because there are not many code contributors and the project is entirely GPL-based and is not dual-licensed. Yeah; I was just worried about someone getting pissy about your relicensing from GPL to AGPL. Not that I expect it or anything... :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
At 03:05 PM 4/14/2008, Doug wrote: Anybody have recommendations for a reliable, good valued, E911 provider? In my experience, the most reliable service for me has always been associated with commercial PSTN number providers. When it comes to consumer line service, you want E911 to always work correctly as a human life is often at risk. We use a $.$$ per-call pass-thru via a major US carrier. We pass the service along to our wholesale DID clients. Even though it costs, it is such low usage, that very few of our clients pass it along to their retail consumers. I will look with interest at other responses to your question. ..mike.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On Mon, Apr 14, 2008 at 03:24:02PM -0400, Matt Florell wrote: As for contributed code, we require a statement of this is my code and the project can use it and redistribute it from the author. Nothing very detailed at the moment because there are not many code contributors and the project is entirely GPL-based and is not dual-licensed. Yeah; I was just worried about someone getting pissy about your relicensing from GPL to AGPL. Not that I expect it or anything... :-) I am fairly surprised that I have not heard a single negative comment about it from any members of our VICIDIAL community or anywhere else. Well, I'm not, actually... the people who *like* the GPL (that's, y'know, everyone except Trixter :-) would be more inclined to like AGPL, I would think; it merely extends the letter to better reflect the spirit -- which a lot of people think GPl3 does *not* do... There are actually other web-based projects that are moving to it as well(which is how I originally heard about it) and since it became an official OSI-approved Open Source License along with the special provisions that GPL made allowing for AGPL compatibility, there are more people talking about it in the last few months. If OSI approved it don't *they* then have the official language? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
On Mon, Apr 14, 2008 at 03:23:45PM -0400, Kristian Kielhofner wrote: Wow, that response was completely unnecessary. I think most people (myself included) know what he meant. Clearly, no, *I* don't. Or I wouldn't have asked. I think, for my part, that *your* attitude was itself unnecessary. But we can take this off list. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
On Mon, Apr 14, 2008 at 03:20:02PM -0400, Alex Balashov wrote: Jay R. Ashworth wrote: On Mon, Apr 14, 2008 at 02:05:18PM -0500, Doug wrote: Anybody have recommendations for a reliable, good valued, E911 provider? Wow. E911 providers are *municipalities*, aren't they? :-) No, they're not. There are service companies specialising in the delivery of 911 calls to the correct PSAP along with maintaining and updating E.911 placement/ALI information. Hmm. I wasn't aware that either of those functions were permitted to be outsourced. Clearly I have much to learn. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
I'm in the same boat. And we don't need any snide comments because this is a potential liability. Municipalities don't provide E911, they are users of E911 data. If you are not a phone company and you want the E911 data updated with correct addresses, then you need to pay someone to do that for you. That is unless I grossly misunderstand it. On Mon, Apr 14, 2008 at 02:05:18PM -0500, Doug wrote: Anybody have recommendations for a reliable, good valued, E911 provider? Wow. E911 providers are *municipalities*, aren't they? :-) Could you vague that up a bit, Doug? (Or should I be able to generalize that phrasing into what you actually mean, if I expect to get along here? :-) Cheers, -- jra ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
Jay R. Ashworth wrote: On Mon, Apr 14, 2008 at 03:20:02PM -0400, Alex Balashov wrote: Jay R. Ashworth wrote: On Mon, Apr 14, 2008 at 02:05:18PM -0500, Doug wrote: Anybody have recommendations for a reliable, good valued, E911 provider? Wow. E911 providers are *municipalities*, aren't they? :-) No, they're not. There are service companies specialising in the delivery of 911 calls to the correct PSAP along with maintaining and updating E.911 placement/ALI information. Hmm. I wasn't aware that either of those functions were permitted to be outsourced. Clearly I have much to learn. They cannot be outsourced entirely, in any kind of meaningful financial sense. It is still necessary for the end VoIP provider to collect the necessary waivers from the user, to obtain accurate and up-to-date address information from the end-user, and to provide facilities to complete the E911 call. The part that can be outsourced is the actual delivery of the call to the correct PSAP. It doesn't make economic sense for a nationwide VoIP service provider[1] to obtain hard tandem trunks to the various PSAPs in various LATAs all throughout for a service that enjoys at best very incidental use and whose costs cannot be recouped. So, what typically happens is that the VoIP service provider uses some sort of intra-industrial interface gateway, web service, API, etc. to build ALI information for every customer and transmit that to the E911 carrier, which updates it in its own database and actually provides the PSAP connectivity. In other words, there's still a lot of legwork to be done by the VoIP provider, so I don't know that outsourced is really the right term for it. [1] Assuming you think nationwide VoIP service makes sense as a business model ... heh heh. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
On Mon, Apr 14, 2008 at 3:43 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 03:23:45PM -0400, Kristian Kielhofner wrote: Wow, that response was completely unnecessary. I think most people (myself included) know what he meant. Clearly, no, *I* don't. Or I wouldn't have asked. Everyone else was able to provide a helpful, constructive response. I think that speaks for itself. I think, for my part, that *your* attitude was itself unnecessary. But we can take this off list. I won't bother furthering this, on list or off. I'm fine to leave it as a reminder for all of us: never feed the trolls. I forget about that every once in a while. -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] polycom auto answer
I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call 22 and the phone rang it did not auto answer. Did I miss something? exten = 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten = 22,n,SipAddHeader(Alert-Info: Ring Answer) exten = 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten = 22,n,Set(__ALERT_INFO=Ring Answer) exten = 22,n,Set(__SIP_URI_OPTIONS=intercom=true) exten = 22,n,Dial(SIP/404) Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 03:24:02PM -0400, Matt Florell wrote: As for contributed code, we require a statement of this is my code and the project can use it and redistribute it from the author. Nothing very detailed at the moment because there are not many code contributors and the project is entirely GPL-based and is not dual-licensed. Yeah; I was just worried about someone getting pissy about your relicensing from GPL to AGPL. Not that I expect it or anything... :-) I am fairly surprised that I have not heard a single negative comment about it from any members of our VICIDIAL community or anywhere else. Well, I'm not, actually... the people who *like* the GPL (that's, y'know, everyone except Trixter :-) would be more inclined to like AGPL, I would think; it merely extends the letter to better reflect the spirit -- which a lot of people think GPl3 does *not* do... There are actually other web-based projects that are moving to it as well(which is how I originally heard about it) and since it became an official OSI-approved Open Source License along with the special provisions that GPL made allowing for AGPL compatibility, there are more people talking about it in the last few months. If OSI approved it don't *they* then have the official language? Yes and no, they have the official language for AGPLv3, but not AGPLv2 which is the actual license that they first approved on March 12th. I can't find the exact version 2 draft that they approved, since it seems that they moved immediately to post version 3 on their website and just skipped version 2. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
Ok so did anybody have recommendations? How's 911Enable.com? Anybody have recommendations for a reliable, good valued, E911 provider? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 Recommendations?
On Mon, Apr 14, 2008 at 4:45 PM, Adam Moffett [EMAIL PROTECTED] wrote: Ok so did anybody have recommendations? How's 911Enable.com? Anybody have recommendations for a reliable, good valued, E911 provider? I did a while back - Dash 911 / Dash Carrier Services. We looked at 911 Enable. We didn't like their API or they way they handled calls. I don't remember all of the details but you could certainly check out both and see what you think. -- Kristian Kielhofner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap Codec
Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Hi list, After a lot of testing + troubleshooting, I guess I'm observing what I am now calling a regression with zaptel 1.4.10 (is it?) As such I call for peer feedback, before either asking Digium install support or filing a bug. Thanks in advance! System: HP Proliant DL380 G5 with 2x PCI-X + 1x PCIe riser card OS: Centos 5 Kernel: 2.6.18-53.1.14.el5 (also tested under 2.6.18-53.el5) HW: Digium TE220B, the one with HW echo cancellation (configured as 2x E1 via jumpers) Context: Pre-site installation of system, no E1 conectivity (loopbacks tested) /etc/zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=25-39,41-55 dchan=40 span=2,2,0,ccs,hdb3,crc4 bchan=56-70,72-86 dchan=71 Under zaptel 1.4.10, when ztcfg runs this gets logged in the kernel buffer: About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 1: Primary Sync Source VPM400: Not Present VPM450: echo cancellation for 64 channels BUG: soft lockup detected on CPU#0! [c044d448] softlockup_tick+0x96/0xa4 [c042ddc8] update_process_times+0x39/0x5c [c04196f7] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [f89bc1e7] init_vpm450m+0x32d/0x34a [wct4xxp] [f89a3b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f89a7ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042621c] release_console_sem+0x17e/0x1b8 [c0407406] do_IRQ+0xa5/0xae [f8994311] t4_dacs+0x211/0x24b [wct4xxp] [f8a01f6a] zt_ioctl+0x273/0x144f [zaptel] [c0457600] mempool_alloc+0x28/0xc9 [c04ddd33] cfq_resort_rr_list+0x23/0x8b [c04deb6c] cfq_add_crq_rb+0xba/0xc3 [c04dec72] cfq_insert_request+0x42/0x498 [c04d5175] elv_insert+0x10a/0x1ad [c04d908b] __make_request+0x31d/0x366 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04dde27] __cfq_slice_expired+0x8c/0xa5 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04d505d] elv_next_request+0x15c/0x16a [f88bc101] start_io+0x77/0xdc [cciss] [f88bf63e] do_cciss_request+0x32c/0x337 [cciss] [f88ccff0] __split_bio+0x408/0x418 [dm_mod] [f88cd6a6] dm_request+0xce/0xd4 [dm_mod] [c04d6a81] generic_make_request+0x248/0x258 [c04d8734] submit_bio+0xbf/0xc5 [c04548e2] find_get_page+0x18/0x38 [c04719ad] __find_get_block_slow+0xfb/0x105 [c0471cea] __find_get_block+0x15c/0x166 [c0471cea] __find_get_block+0x15c/0x166 [c0471d24] __getblk+0x30/0x270 [f885a485] journal_cancel_revoke+0x8a/0x96 [jbd] [f885a472] journal_cancel_revoke+0x77/0x96 [jbd] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [c041f871] __wake_up+0x2a/0x3d [f8856679] journal_stop+0x1b0/0x1ba [jbd] [c042a209] current_fs_time+0x4a/0x55 [c048626d] touch_atime+0x60/0x8f [c04552ee] do_generic_mapping_read+0x421/0x468 [c045478b] file_read_actor+0x0/0xd1 [c04548e2] find_get_page+0x18/0x38 [c0457319] filemap_nopage+0x192/0x315 [c046048f] __handle_mm_fault+0x85e/0x87b [c047f46b] do_ioctl+0x47/0x5d [c047f6cb] vfs_ioctl+0x24a/0x25c [c047f725] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 2: Secondary Sync Source Completed startup! Soft lockup ?! Hmmm... I'm ignorant on this, but it smells fishy ! For completeness sake, driver was previously loaded ok: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :18:08.0[A] - GSI 19 (level, low) - IRQ 98 Found TE2XXP at base address fdff, remapped to f8854000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x375a2400 Reg 1: 0x375a2000 Reg 2: 0x Reg 3: 0x Reg 4: 0x3101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff2031 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) After trying lot's of things (disable ILO, disable USBs, try different kernel, different TE220B, etc), I figured that this soft hangup does not show under zaptel 1.4.9.2... In all due honesty, I haven't got the faintest idea what kind of impact this could have. Side testing zaptel 1.4.10 on a simpler system, an HP Proliant ML110 (nearly a PC), the error does not show up as well. I checked the zaptel 1.4.10 ChangeLog and there are some changes which I'd suspect: 2008-04-01 16:39 + [r4122] sruffell [EMAIL PROTECTED]: * kernel/wct4xxp/base.c: Work around for host bridges that generate fast back to back transactions which the current version of the quad span cards do not advertise support for.
[asterisk-users] CallerID in NZ
Hi There, We have a Asterisk 1.4 box with a X100P card connected to a analog line with Caller ID serrvices enabled on it. When an incoming call appears we get the following in the log: -- Starting simple switch on 'Zap/1-1' -- Detecting post-CID distinctive ring [Apr 15 10:38:07] NOTICE[7151]: chan_zap.c:6469 ss_thread: Got event 18 (Ring Begin)... [Apr 15 10:38:07] NOTICE[7151]: chan_zap.c:6469 ss_thread: Got event 2 (Ring/Answered)... [Apr 15 10:38:07] NOTICE[7151]: chan_zap.c:6469 ss_thread: Got event 18 (Ring Begin)... -- Detected ring pattern: 299,290,279 -- Checking 0,0,0 -- Checking 0,0,0 -- Checking 0,0,0 -- Executing [EMAIL PROTECTED]:1] ExecIf(Zap/1-1, 0|SetCallerPres|unavailable) in new stack -- Executing [EMAIL PROTECTED]:2] ExecIf(Zap/1-1, 0|Set|CALLERID(all)=unknown 000) in new stack So its not seeing the caller id. What might i have incorrect here? Thanks Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
I want to 3rd this. They admitted some of their hardware runs GPL code (Linux, IPTables, wget and more) yet refuse to provide the source code or evidence of an alternate license agreement with the authors of the software (which I doubt they did I just like to give people that benefit of the doubt). But I do think their engineering is excellent. What a waste. On Thu, Apr 3, 2008 at 1:26 AM, Alex Balashov [EMAIL PROTECTED] wrote: Al lists wrote: Bad memories from AudioCodec :) Por que? I'm curious. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SPA3000 -- dropping calls
Hi, all I have SPA3000 (in Linksys reincarnation) and it has very annoying problem. Sometimes, incoming PSTN call drops the moment one picks up analog phone on FXO port. Most of the times it works, other times phone on FXS rings, I pick it up and all I get is a dial tone. Any ideas what may be wrong? Thanks, Rudolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom auto answer
At 15:06 4/14/2008, Jerry Geis wrote: I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call 22 and the phone rang it did not auto answer. Did I miss something? exten = 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten = 22,n,SipAddHeader(Alert-Info: Ring Answer) exten = 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten = 22,n,Set(__ALERT_INFO=Ring Answer) exten = 22,n,Set(__SIP_URI_OPTIONS=intercom=true) exten = 22,n,Dial(SIP/404) Jerry sip.cfg? voIpProt.SIP.alertInfo.2.value=Ring Answer voIpProt.SIP.alertInfo.2.class=4/ Reboot phones? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
On Mon, Apr 14, 2008 at 07:13:30PM -0400, Andreas van dem Helge wrote: I want to 3rd this. They admitted some of their hardware runs GPL code (Linux, IPTables, wget and more) yet refuse to provide the source code or evidence of an alternate license agreement with the authors of the software (which I doubt they did I just like to give people that benefit of the doubt). But I do think their engineering is excellent. What a waste. Somebody call Eben Moglen. He's *itching* to take this to court, instead of having everyone settle out from under him; I know he is. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about queue
Anyone can update me about the queue sticking by a caller? Is it solved in version 1.4.x? How? On Sat, Apr 12, 2008 at 9:42 AM, Rilawich Ango [EMAIL PROTECTED] wrote: Do you mean autofill works in 1.4.x? But it doesn't work even I set it. On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Codec
This is SIP channel you need to limit. Forcing ulaw only, the Polycom will fall back to ulaw. Per peer, in your sip.conf: disallow=all allow=ulaw From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Monday, April 14, 2008 14:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zap Codec Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good??
On Mon, 14 Apr 2008, Mark Hamilton wrote: Is this 'shell script' on the public domain? As it sounds really useful. :) You're welcome to it. I'll reply with a link off-list. The script is definitely a work-in-progress and not quite ready for prime-time. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom auto answer
Jerry, Did you enable Ring Answer in the phone? Look at your sip.cfg file for: alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ and ringType se.rt.enabled=1 se.rt.modification.enabled=1 DEFAULT se.rt.1.name=Default se.rt.1.type=ring se.rt.1.ringer=2 se.rt.1.callWait=6 se.rt.1.mod=1/ VISUAL_ONLY se.rt.2.name=Visual se.rt.2.type=visual/ AUTO_ANSWER se.rt.3.name=Auto Answer se.rt.3.type=answer/ RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1500 se.rt.4.ringer=13 se.rt.4.callWait=6 se.rt.4.mod=1/ INTERNAL se.rt.5.name=Internal se.rt.5.type=ring se.rt.5.ringer=2 Have a look at: http://www.voicerd.org/index.php/Auto_Pickup On Mon, Apr 14, 2008 at 4:06 PM, Jerry Geis [EMAIL PROTECTED] wrote: I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call 22 and the phone rang it did not auto answer. Did I miss something? exten = 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten = 22,n,SipAddHeader(Alert-Info: Ring Answer) exten = 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten = 22,n,Set(__ALERT_INFO=Ring Answer) exten = 22,n,Set(__SIP_URI_OPTIONS=intercom=true) exten = 22,n,Dial(SIP/404) Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] http://www.shift8.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to load module chan_zap.so
On 14/04/2008, at 5:58 PM, Tzafrir Cohen wrote:
In the Asterisk CLI, what happens when you run:
module unload chan_zap.so
module load chan_aap.so
hostname*CLI module unload chan_zap.so
No such command 'module' (type 'help' for help)
hostname*CLI module load chan_zap.so
No such command 'module' (type 'help' for help)
But /usr/lib/asterisk/modules/chan_zap.so does exist and is readable.
In answer to someone else's response, I didn't compile this Asterisk
or its Zaptel module, I'm using the precompiled packages from
updates.xorcom.com - but presumably they should match.
Thanks.
--
Jeremy Malcolm LLB (Hons) B Com
Internet and Open Source lawyer, IT consultant, actor
host -t NAPTR 1.0.8.0.3.1.2.9.8.1.6.e164.org|awk -F! '{print $3}'
Luxury Perth apartment for sale!
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Re: [asterisk-users] Unable to load module chan_zap.so
On Tue, Apr 15, 2008 at 10:33:43AM +0800, Jeremy Malcolm wrote: On 14/04/2008, at 5:58 PM, Tzafrir Cohen wrote: In the Asterisk CLI, what happens when you run: module unload chan_zap.so module load chan_aap.so hostname*CLI module unload chan_zap.so No such command 'module' (type 'help' for help) hostname*CLI module load chan_zap.so No such command 'module' (type 'help' for help) This is Asterisk 1.2: unload chan_zap.so load chan_zap.so -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] do cards just instantly go bad
If you really suspect the card is at fault, you can try to plug in both ports with a crossover cable and try making a phone call that goes from one port to the other. Or you can have the provider send down a tech with a T-Bird. On Mon, Apr 14, 2008 at 1:05 PM, Mike Trest - On Travel [EMAIL PROTECTED] wrote: At 12:52 PM 4/14/2008, Jerry Geis wrote: Hi - Been using a TE205P for a number of months - no issues. Today I was talking to someone and I heard click No more phone service. I still have data service on this T1 line. (partial phone) zttool reports the SPAN as OK. calls are not coming in or going out. Does a card just go bad like that? How can I tell if the card is bad? I was expecting/hoping to see something other than OK on zttool. Its reporting OK but still no calls. Probably not the card. More likely the T1 provider. Contact your carrier. Ask them to put a T1 level trouble ticket. Ask that you be on the phone when the tester is bringing the T1 down and up. Quite often, all that is needed is to BOUNCE the circuit from the switch end. ..mike.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conferencing..
I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
