Re: [asterisk-users] 1.8.5 Voicemail duration incorrect
Hi Robert, Am Donnerstag, den 25.08.2011, 13:28 -0400 schrieb Robert Huddleston: https://issues.asterisk.org/jira/browse/ASTERISK-16981 Thank You for the link. I already found it a few hours later. I put some debug output in the code and I think I found the location of the issue, but I currently do not know, how to fix it. (See comments in jira). Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk spontaneous reboot
Hello, Today I restart the MySQL-DB (/sbin/service mysqld restart) and I could no longer connect to asterisk (/usr/sbin/asterisk -r) for a few seconds. There is now a core dump present in /tmp : -rw--- 1 root root 88M Aug 26 08:07 core.sip.pbx.tld-2011-08-26T08:07:35+0200 How can I get usefull information about what went wrong ? Because a spontaneous reboot of Asterisk has never happened before when just restarting the MySQL-deamon. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] red alarm on tdm400 fxo (fxs signalled)
On Friday 26 August 2011, Kelvin Chua wrote: has anybody ever seen a red alarm on an fxo port (tdm400) whenever you unplug a pstn line? I think i saw a post on the mailing list a few years back about this, but never actually seen one That's what mine (TDM410P clone with 2 * FXO + 2 * FXS) does if a phone line is disconnected: it gives a red alarm, but the alarm condition clears as soon as the line is plugged back in. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] red alarm on tdm400 fxo (fxs signalled)
i am comparing the experience of using an analog span to a T1 for example: if i have a 3 quad port t1 card, with the initial view of dahdi_tool, i can easily tell if a line is not working now supposed i have a system with 3 24port cards, with just watching the main view of dahdi_tool, i cannot do that, i still need to go into each one, and look at the stats maybe it's just me but i think it's making things easier for everyone if there is some sort of an indication there like please-take-a-look-at-this-as-i-have-a-port-here-without-a-battery (not really yellow, perhaps orange? :D) using callprogress is *experimental* ; This feature can also easily detect false hangups. The symptoms of this is ; being disconnected in the middle of a call for no reason. it's actually ok for me, but for other zones, it might not be adviseable. for example, calling a mobile phone and a landline would give you a different tone and cadence for the ringback thus billing apps will be completely screwed. (unless already anticipated by the billing software) what i have in mind is just plain simple. when the fxo port goes offhook, it waits for the dial tone (based on tonezone) after that, nothing more so as to avoid detection of false hangups. i think this is not too hard as Newman Ventures implemented something like this before. Kelvin Chua On Fri, Aug 26, 2011 at 11:51 AM, Shaun Ruffell sruff...@digium.com wrote: On Fri, Aug 26, 2011 at 10:58:10AM +0800, Kelvin Chua wrote: I should clarify on my post, i can see a red alarm on cat /proc/dahdi/1 but never when using dahdi_tool in cat /proc/dahdi/1: Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 FXSKS (In use) RED(SWEC: OSLEC) 2 WCTDM/4/1 FXSKS (In use) RED(SWEC: OSLEC) 3 WCTDM/4/2 FXSKS (In use) RED(SWEC: OSLEC) 4 WCTDM/4/3 FXSKS (In use) RED(SWEC: OSLEC) In dahdi_tool: Tx/Rx Levels: 0/ 0 Total/Conf/Act: 4/ 4/ 0 but the behavior status alarm is OK a little bit misleading, what i am thinking is adding a sort of yellow alarm or something if one of the configured fxo lines goes down? I understand now. Yes, dahdi_tool is showing the alarm status for the entire span, and for analog cards, the entire span is always essentially green. It's the individual channels that can go into red alarm unlike on digital spans. Why do you want to add a yellow status? What are you trying to accomplish that you cannot with the way things are now? another thing, what if 1 of the configured lines' battery is up, but there is no dial tone? outbound calls will try to go through and the caller will hear dead air. 'callprogress=yes' in chan_dahdi.conf won't help you here? It may be able to detect dialtone before attempting to dial out. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to know how many user is connected
Google linux commands for the purpose.
Not sure about preemptively disconnecting sockets . I think there are
commands like ss in linux which you can use. You need to collect info from
AMI and then use combination of linux commands via php directly to
disconnect anyone (if possible).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, August 25, 2011 6:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to know how many user is connected
Hi Ahmed,
Just realized that maybe youre talking about disconnecting any other
AMI/manger connected user from another manager connection
hhmmm
I dont
think so. Check AMI commands from asterisk wiki. If not, you may need system
command in your AMI connection to close some other socket
Yes I was looking for this :)
Please tell me how to close other socket from current sockets.
one more thing in my case it may be possible that
root 127.0.0.1 may be more then one then how to close them individually?
On Thu, Aug 25, 2011 at 5:09 PM, Gohar Ahmed gohar.ah...@vopium.com wrote:
Just realized that maybe youre talking about disconnecting any other
AMI/manger connected user from another manager connection
hhmmm
I dont
think so. Check AMI commands from asterisk wiki. If not, you may need system
command in your AMI connection to close some other socket.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gohar Ahmed
Sent: Thursday, August 25, 2011 4:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to know how many user is connected
What I understood: you need to disconnect the AMI socket.
1) I want to disconnect connected manager into Asterisk. Is it possible ?
ß Close the $socket after you get the response.
What I understood: you need to maintain the socket until some button is
pressed to stop AMI
2) I want to maintain this socket connection until we disconnect it from web
page. ß Close the $socket on particular action from web-page. This ones
tricky btw maintain a while loop and break loop on a condition toggled by
web-page)
See php section for other examples.
http://www.voip-info.org/wiki/view/Asterisk+manager+Examples
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, August 25, 2011 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to know how many user is connected
Hi List,
Thanks now I am able to get all values from asterisk CLI but I want 2 more
things .
1) I want to disconnect connected manager into Asterisk. Is it possible ?
2) I want to maintain this socket connection until we disconnect it from web
page.
On Thu, Aug 25, 2011 at 1:57 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
Thanks for guide me. Yes I know that CLI command , My motive is to get
information into Php that's why I am finding the solution.
Ahmad Sir, You are right I forget to get information back from CLI to Php
file. Thanks for provide the help link.I will revert back after testing my
code with your guidance
On Thu, Aug 25, 2011 at 12:21 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi
You can use simple cli command
Manager show connected
On Thursday, August 25, 2011, James zhu zhulizh...@live.com wrote:
hi:
please refer this:
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP
and check the manager.conf, make sure the accounts in managers.conf matchs
the managers displayed.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: gohar.ah...@vopium.com
To: asterisk-users@lists.digium.com
Date: Thu, 25 Aug 2011 11:26:53 +0500
Subject: Re: [asterisk-users] How to know how many user is connected
Im not a php expert, but seems your php script is incomplete/ you are
sending to socket (fputs) but note receiving anything(fgets) :
See this page
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP will help
you.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, August 24, 2011 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know how many user is connected
Hi List,
I want to know how many manager is connected into asterisk server. I have
made simple file but I don't have any idea how to get information back from
Asterisk CLI
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: root\r\n);
[asterisk-users] DMTF via rfc2833 and SIP-INFO simultaneously
Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO simultaneously. That has the effect, that asterisk read every dtmf twice. and yes, it's mainly the carriers mistake. but is there a configure option, that asterisk accept only one DMTF method for inbound dtmf? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wanted a modified SIP message body
Hello everybody, I want Asterisk Server to send packets (SIP packets) to some 3Com telephones with the text TZ: 7200\n (ie Time Zone = two hours) in the message body because 3com PBX sends this variable. I would like to know if I it is possible to configure Asterisk to do it, and how. have a nice day! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk spontaneous reboot
use gdb (The GNU Project Debugger) to take a look into the core dump gdb asterisk core.sip.pbx.tld-2011-08-26T08:07:35+0200 Kristijan 2011/8/26 Jonas Kellens jonas.kell...@telenet.be: Hello, Today I restart the MySQL-DB (/sbin/service mysqld restart) and I could no longer connect to asterisk (/usr/sbin/asterisk -r) for a few seconds. There is now a core dump present in /tmp : -rw--- 1 root root 88M Aug 26 08:07 core.sip.pbx.tld-2011-08-26T08:07:35+0200 How can I get usefull information about what went wrong ? Because a spontaneous reboot of Asterisk has never happened before when just restarting the MySQL-deamon. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted a modified SIP message body
Use the *SIPAddHeader(Header:Content)* application in dialplan. I don't think Method specific SIP headers can be done via asterisk. On Fri, Aug 26, 2011 at 3:05 PM, Jaime Lozano jaimelozan...@gmail.comwrote: Hello everybody, I want Asterisk Server to send packets (SIP packets) to some 3Com telephones with the text TZ: 7200\n (ie Time Zone = two hours) in the message body because 3com PBX sends this variable. I would like to know if I it is possible to configure Asterisk to do it, and how. have a nice day! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] red alarm on tdm400 fxo (fxs signalled)
waitfordialtone=yes on chan_dahdi.conf is supposed to be the perfect solution, but does it work on on UK lines? Kelvin Chua On Fri, Aug 26, 2011 at 4:44 PM, Kelvin Chua kel...@gmail.com wrote: i am comparing the experience of using an analog span to a T1 for example: if i have a 3 quad port t1 card, with the initial view of dahdi_tool, i can easily tell if a line is not working now supposed i have a system with 3 24port cards, with just watching the main view of dahdi_tool, i cannot do that, i still need to go into each one, and look at the stats maybe it's just me but i think it's making things easier for everyone if there is some sort of an indication there like please-take-a-look-at-this-as-i-have-a-port-here-without-a-battery (not really yellow, perhaps orange? :D) using callprogress is *experimental* ; This feature can also easily detect false hangups. The symptoms of this is ; being disconnected in the middle of a call for no reason. it's actually ok for me, but for other zones, it might not be adviseable. for example, calling a mobile phone and a landline would give you a different tone and cadence for the ringback thus billing apps will be completely screwed. (unless already anticipated by the billing software) what i have in mind is just plain simple. when the fxo port goes offhook, it waits for the dial tone (based on tonezone) after that, nothing more so as to avoid detection of false hangups. i think this is not too hard as Newman Ventures implemented something like this before. Kelvin Chua On Fri, Aug 26, 2011 at 11:51 AM, Shaun Ruffell sruff...@digium.comwrote: On Fri, Aug 26, 2011 at 10:58:10AM +0800, Kelvin Chua wrote: I should clarify on my post, i can see a red alarm on cat /proc/dahdi/1 but never when using dahdi_tool in cat /proc/dahdi/1: Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 FXSKS (In use) RED(SWEC: OSLEC) 2 WCTDM/4/1 FXSKS (In use) RED(SWEC: OSLEC) 3 WCTDM/4/2 FXSKS (In use) RED(SWEC: OSLEC) 4 WCTDM/4/3 FXSKS (In use) RED(SWEC: OSLEC) In dahdi_tool: Tx/Rx Levels: 0/ 0 Total/Conf/Act: 4/ 4/ 0 but the behavior status alarm is OK a little bit misleading, what i am thinking is adding a sort of yellow alarm or something if one of the configured fxo lines goes down? I understand now. Yes, dahdi_tool is showing the alarm status for the entire span, and for analog cards, the entire span is always essentially green. It's the individual channels that can go into red alarm unlike on digital spans. Why do you want to add a yellow status? What are you trying to accomplish that you cannot with the way things are now? another thing, what if 1 of the configured lines' battery is up, but there is no dial tone? outbound calls will try to go through and the caller will hear dead air. 'callprogress=yes' in chan_dahdi.conf won't help you here? It may be able to detect dialtone before attempting to dial out. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
Great discussion, all of it. Thanks, people. How much power does the home asterisk box need ? I'm using Asus Eee Box (1012Ps) as Myth front ends in another project. About $280 with 320 Gb hard drive and 2 GB RAM. Atom 510 processor. Built in Wifi. Nearly silent. Runs F15 nicely. Would one of them suffice ? FWIW, I'm typing this email on one now because my main system is down. It looks like I am going to need an ATA for the fax machines. Two. My wife informed me yesterday she wants her own in her office. VOIP handles fax machines, right ? I'm wondering what phones everyone is using. Should I stick with analog wireless handsets or are there some good SIP wireless phones out there that I don't know about ??? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted a modified SIP message body
Hello, In which file do I use SIPAddHeader()? Please consider that the packet goes from the PBX to the telephone, and what I want is not a header because the TZ: 7200\n is in the *message body* not in the *message header*. Have a nice day 2011/8/26 Sam Govind govoi...@gmail.com Use the *SIPAddHeader(Header:Content)* application in dialplan. I don't think Method specific SIP headers can be done via asterisk. On Fri, Aug 26, 2011 at 3:05 PM, Jaime Lozano jaimelozan...@gmail.comwrote: Hello everybody, I want Asterisk Server to send packets (SIP packets) to some 3Com telephones with the text TZ: 7200\n (ie Time Zone = two hours) in the message body because 3com PBX sends this variable. I would like to know if I it is possible to configure Asterisk to do it, and how. have a nice day! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
If you really want to go that route, you should also look at AstLinux and install it on an HP thin client such as a 5720. No Hard Drive spinning, and something like 30 watts. No fan either. All the asterisk files can be edited either through SSH or a web interface. I have a bunch out working for a private VOIP network of telephone collectors. Many also integrate PSTN lines through various means 5720's can be had on eBay for a LOT less money John Novack linux guy wrote: Great discussion, all of it. Thanks, people. How much power does the home asterisk box need ? I'm using Asus Eee Box (1012Ps) as Myth front ends in another project. About $280 with 320 Gb hard drive and 2 GB RAM. Atom 510 processor. Built in Wifi. Nearly silent. Runs F15 nicely. Would one of them suffice ? FWIW, I'm typing this email on one now because my main system is down. It looks like I am going to need an ATA for the fax machines. Two. My wife informed me yesterday she wants her own in her office. VOIP handles fax machines, right ? I'm wondering what phones everyone is using. Should I stick with analog wireless handsets or are there some good SIP wireless phones out there that I don't know about ??? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted a modified SIP message body
26 aug 2011 kl. 14:06 skrev Jaime Lozano: Hello, In which file do I use SIPAddHeader()? Please consider that the packet goes from the PBX to the telephone, and what I want is not a header because the TZ: 7200\n is in the *message body* not in the *message header*. That's no longer a SIP header, it's part of the SDP you want to change. You can't do that without changing the source code. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_sms testers required
There are three problems with app_sms in Asterisk 1.8.5.0: - It fails to initialise a 'flags' variable and ends up sending with protocol 2 in some cases when it shouldn't, which obviously fails. - The smsq tool fails to install its call file into the outgoing queue in a way that Asterisk will see it. - If disconnect supervision isn't functioning or inbanddisconnect=yes is set in chan_dahdi.conf, it will keep bitching about 'bad stop bit' over and over again, without even a newline at the end of the message, even when we *told* the peer to go away by sending a REL message. These issues are all fixed with the patch in https://issues.asterisk.org/jira/browse/ASTERISK-18331 Apparently we need a popularity contest even for the obviously-correct parts of the above which have been reported elsewhere¹ in the bug tracking system but ignored. So if any users of app_sms could test the patch and confirm that it works for them, it would be massively appreciated. And would reduce the chances of me getting too frustrated with Asterisk and spending my time helping out with other projects which will appreciate it more. -- dwmw2 ¹ https://issues.asterisk.org/jira/browse/ASTERISK-15361?focusedCommentId=160885#comment-160885 smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
On Fri, 26 Aug 2011, linux guy wrote: How much power does the home asterisk box need ? Much less than you would think. Any modern processor is more than enough. I'm using Asus Eee Box (1012Ps) as Myth front ends in another project. About $280 with 320 Gb hard drive and 2 GB RAM. Atom 510 processor. Built in Wifi. Nearly silent. Runs F15 nicely. Would one of them suffice ? More than enough unless your extended multi-generation family lives with you. And you multiply like rabbits. And you all want to talk at the same time. The LinuxMCE (http://http://linuxmce.org/) project may be of interest to you. They integrate Asterisk, Myth, home automation and the kitchen sink into a single distribution. They just announced their 8.10 release candidate a couple of days ago. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Group not forwaring calls to agents
did you find al solution for this issues? i fight with the same problem. kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Trace to troubleshoot one way of communications
Hi All; How can I get a SIP trace to troubleshoot a one way of communications? I need to see what is happenning in the packets to know the reason of the problem. Thanks Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
Great discussion, all of it. Thanks, people. How much power does the home asterisk box need ? I'm using Asus Eee Box (1012Ps) as Myth front ends in another project. About $280 with 320 Gb hard drive and 2 GB RAM. Atom 510 processor. Built in Wifi. Nearly silent. Runs F15 nicely. Would one of them suffice ? I'm running my small home Asterisk system on an Itox motherboard with an Atom N270, at 1.6 GHz. No CPU-related problems noted. In fact, I'd run it fairly successfully on a Pentium Pro 200, and it worked well enough for simple uses (e.g. no fancy codecs or transcoding). It looks like I am going to need an ATA for the fax machines. Two. My wife informed me yesterday she wants her own in her office. VOIP handles fax machines, right ? This could very well be the most problematic (heart-breaking, frustrating) part of your whole intended installation. Fax - modem - very sensitive to jitter and dropouts. Making fax work over VoIP (using A-law or u-law) is often feasible within a LAN environment, because the jitter and packet-loss rates are low. Making fax work decently on VoIP over the Internet is much harder... jitter and packet-loss rates which would be slightly annoying for a voice conversation can seriously disrupt or abort a fax call. Some (relatively few) VoIP providers support a specialized mode called T.38. in which their far end equipment intervenes in the fax protocol in order to smooth out the process of making fax work over a lossy/jittery/high-latency VoIP connection. This isn't common and seems to be hard to count on. I suspect you'll be better off either: (1) maintaining one analog land-line, and using it for fax (and perhaps backup for VoIP), and/or (2) subscribing to a commercial fax to email gateway service, in which people send their faxes to a number owned by the service provider, and the resulting fax is converted to a compressed PDF and then emailed to you. I imagine that some of these providers also have an email to fax service, operating in the reverse direction... you email a PDF or other file to an address alias they provide, and it's faxed out for you. You *can* operate a sort of hybrid system in your house, if that's convenient to you... e.g. a SIP ATA for your fax machine, to Asterisk, to an analog land-line (via either a dedicated PCI bus card, or an outbound port on a channel bank or certain ATA devices). The jitter and delay on the home LAN would be low enough that this should work reliably. You could also run a combination of hylafax, and iaxmodem on the Asterisk system, and thus use the Asterisk server as a fax-to-email / email-to-fax / document-to-fax gateway. I'm wondering what phones everyone is using. Should I stick with analog wireless handsets or are there some good SIP wireless phones out there that I don't know about ??? Several companies make DECT SIP phones and systems... typically I think they'll handle 4 to 6 DECT handsets, and a couple of independent SIP calls at any given moment. These may or may not be less expensive than buying some one- or two-analog-line SIP systems, and some ATAs; they'd definitely involve less equipment and wiring. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good, inexpensive wireless VOIP handsets ?
I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. Any ideas ? Thanks ! LG -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
On 08/26/2011 02:02 PM, linux guy wrote: get any cheap android device and load linphone. or grandstream works for a wired device. gxp2000 has enough line buttons you can easily route calls for multiple people to a phone and tell who the call is for I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. Any ideas ? Thanks ! LG -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
I was thinking of using a PAP2T-NA for the ATA to handle the fax. It appears to have a large number of fax specific settings. Can anyone comment on using this device with a fax ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
In our house, we need wireless. I have a Grandstream already. I am looking for something with a form factor more conventional than a cellphone. Maybe that is silly ? I see various unlocked large screen Android devices for ~$150. I was hoping to spend on the order of $50 per handset. I don't understand why (other than economies of scale) that I can buy various wireless POTS wireless systems (base and multiple handsets) for $100 and yet a single VOIP wireless handset is $150. Any comments on integrating a wireless POTS system into an asterisk system ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
On 08/26/2011 02:18 PM, linux guy wrote: In our house, we need wireless. I have a Grandstream already. I am looking for something with a form factor more conventional than a cellphone. Maybe that is silly ? I see various unlocked large screen Android devices for ~$150. not sure what you mean more than a cellphone ? get an ata such as an SPA series with 2 channels on it or PAP2T and plug a cordless base station in each one if you want to go cheap. it works but a single line cordless does not at all do justice to what can be done with asterisk. When you say large screen are you talking about an APAD type thing? they work just fine mounted on the wall (I use them for intercom and general control system interface) but I would not walk around holding one to my ear. I was hoping to spend on the order of $50 per handset. I don't understand why (other than economies of scale) that I can buy various wireless POTS wireless systems (base and multiple handsets) for $100 and yet a single VOIP wireless handset is $150. Any comments on integrating a wireless POTS system into an asterisk system ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
On Fri, 2011-08-26 at 12:10 -0600, linux guy wrote: I was thinking of using a PAP2T-NA for the ATA to handle the fax. It appears to have a large number of fax specific settings. Can anyone comment on using this device with a fax ? If you are using POTs to bring in your fax calls you should be ok for home use. I do this with a PAP2T-NA, Hylafax, and iaxmodem. I have iaxmodem accept the fax, then relay it to the PAP2T. I use the second port to drive a Panasonic DECT base station with five satellites, which I have spread around the house. The Panasonic is the only one I have found that has the ability to host a ton of satellites without having a ton of redundant features on the base station (don't really need an answering machine, for example). Its also the only one where the handsets have nice 2.5mm headset jacks, which I use all day. I've never really understood the need for wireless SIP handsets. An ATA plus a normal DECT set seems perfect to me. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
On Friday, August 26, 2011, linux guy wrote: Any comments on integrating a wireless POTS system into an asterisk system ? All you need is an ATA channel per handset ... FWIW, I've got three DECT analog phones in my system: two are hooked into a Linksys PAP2 and the third is hooked into a Grandstream ATA286. These are just like any other SIP client as far as Asterisk is concerned. I also had one hooked up via a Digium IAXy. ATM, the PAP2 route seems the cheapest as there's loads on eBay (at least, there are in UK). Just make sure what you get is unlocked because some VOIP providers use them and lock them into their own networks. HTH, -- Geoff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
Do any of the DECT systems handle multiple incoming phone lines ? How do the DECT systems integrate with the voice mail services on an Asterisk system ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
On Fri, 2011-08-26 at 12:37 -0600, linux guy wrote: Do any of the DECT systems handle multiple incoming phone lines ? How do the DECT systems integrate with the voice mail services on an Asterisk system ? The single line Panasonic that I use doesn't handle multiple phone lines itself, but the ATA will give you call waiting capability, and asterisk will give you the ability to take in multiple DID numbers (via ITSP or POTS). The voicemail in Asterisk is used via the dialplan like you would with any handset. With stock FreePBX, for example, you would dial *97 to access the voicemail associated with the extension that covers that PORT (which would be ALL of the handsets on that basestation). j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
On 08/26/2011 02:26 PM, Jeff LaCoursiere wrote: On Fri, 2011-08-26 at 12:10 -0600, linux guy wrote: I was thinking of using a PAP2T-NA for the ATA to handle the fax. It appears to have a large number of fax specific settings. Can anyone comment on using this device with a fax ? If you are using POTs to bring in your fax calls you should be ok for home use. I do this with a PAP2T-NA, Hylafax, and iaxmodem. I have iaxmodem accept the fax, then relay it to the PAP2T. I use the second port to drive a Panasonic DECT base station with five satellites, which I have spread around the house. The Panasonic is the only one I have found that has the ability to host a ton of satellites without having a ton of redundant features on the base station (don't really need an answering machine, for example). Its also the only one where the handsets have nice 2.5mm headset jacks, which I use all day. I've never really understood the need for wireless SIP handsets. An ATA plus a normal DECT set seems perfect to me. The way I use it I have one device in my pocket, I can get my calls on the couch, in the yard, down the street, at the office or in another city. sip + wifi doesn't just have to be at your house, it can be anywhere there is wifi available. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
On 08/26/2011 01:02 PM, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. I don't have any experience with them, but the Siemens Gigaset A580 IP seems to be about the best price point: http://www.voipsupply.com/siemens-gigaset-a580-ip -- Ian Pilcher arequip...@gmail.com If you're going to shift my paradigm ... at least buy me dinner first. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Trace to troubleshoot one way of communications
Hi, Sip debug can be enabled thru asterisk CLI. SIP signaling trace is printed on the console or terminal if you're connected on asterisk CLI. If you want a complete trace of signaling media packets, packet capturing tools like tcpdump, ethereal wireshark will perfectly address your concerns. Try to be oriented on the packet capturing tools available on the OS your asterisk is installed with. Regards Jesie On Aug 26, 2011, at 11:53 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How can I get a SIP trace to troubleshoot a one way of communications? I need to see what is happenning in the packets to know the reason of the problem. Thanks Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
I like the idea of running multiple ATAs with a single base or handset on each line. Something like the Panasonic KX-TG4111B which sells for about $40 for a handset and base. PAP2s sell for about $50 or $25 per line. Total cost of $65 per handset. Comments on this approach ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
On Fri, Aug 26, 2011 at 12:55 PM, Ian Pilcher arequip...@gmail.com wrote: On 08/26/2011 01:02 PM, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. I don't have any experience with them, but the Siemens Gigaset A580 IP seems to be about the best price point: http://www.voipsupply.com/siemens-gigaset-a580-ip Will those handsets talk directly to the SIP server ? Ie, can one throw away the base station ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_sms testers required
On 08/26/2011 09:29 AM, David Woodhouse wrote: There are three problems with app_sms in Asterisk 1.8.5.0: - It fails to initialise a 'flags' variable and ends up sending with protocol 2 in some cases when it shouldn't, which obviously fails. - The smsq tool fails to install its call file into the outgoing queue in a way that Asterisk will see it. - If disconnect supervision isn't functioning or inbanddisconnect=yes is set in chan_dahdi.conf, it will keep bitching about 'bad stop bit' over and over again, without even a newline at the end of the message, even when we *told* the peer to go away by sending a REL message. These issues are all fixed with the patch in https://issues.asterisk.org/jira/browse/ASTERISK-18331 Apparently we need a popularity contest even for the obviously-correct parts of the above which have been reported elsewhere¹ in the bug tracking system but ignored. So if any users of app_sms could test the patch and confirm that it works for them, it would be massively appreciated. And would reduce the chances of me getting too frustrated with Asterisk and spending my time helping out with other projects which will appreciate it more. That's pretty harsh, David. This has nothing to do with lack of appreciation, and everything to do with a limited set of developer resources and a *massive* pile of issues to deal with. As far as the team of people that Digium employs, the issues they work on are selected by using a process in our department that emphasizes 'return on investment'; a combination of the 'value to the community' of the issue being addressed and the amount of effort it is likely to consume. I hope you'd agree that spending time on issues that affect a large percentage of the user community, even if they take a longer period of time, is a better choice than addressing (possibly many) quick issues that don't affect a significant portion of the user community. Yes, every issue affects *someone* in the user community, and for them, it's important. However, I'd ask you to look over the ChangeLogs for the last few releases, for work that was done by the Digium developers, and point out anything you feel provided less value to the community than fixing problems in app_sms. There are still quite a few community developers actively working on Asterisk, and of course they are free to help move your issues along. In addition, if you are willing to be the 'maintainer' for the areas of Asterisk that interest you, that option is open as well. It's always better for the community as a whole if there are people with direct interest in specific modules/applications/etc. who are maintaining them. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
On Fri, 2011-08-26 at 13:18 -0600, linux guy wrote: On Fri, Aug 26, 2011 at 12:55 PM, Ian Pilcher arequip...@gmail.com wrote: On 08/26/2011 01:02 PM, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. I don't have any experience with them, but the Siemens Gigaset A580 IP seems to be about the best price point: http://www.voipsupply.com/siemens-gigaset-a580-ip Will those handsets talk directly to the SIP server ? Ie, can one throw away the base station ? -- Not the base station - you throw away the ATA. But yes, that is the idea. I have to admit that price point is starting to look more attractive. At least for my house phone my family expects to be able to pick up any handset in the house and be in conference with whatever conversation is going on, and I have solved that. I understand the benefits of a SIP client on a wifi / 3G+ phone, but when I have tried that in the past a few things have gotten in the way: * battery life on smartphones basically sucks !@#$@, and I find my smartphone is plugged into its charger more often than it is in my pocket, so I like the idea of handsets being littered around the house * voice and call quality - at least the last time I tried this (admittedly over a year ago) - was really lacking and inconsistent, and if on the road handoff usually breaks the call between towers * too complex for the wife ;) j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
At 04:28 AM 8/26/2011, you wrote: I'm using Asus Eee Box (1012Ps) as Myth front ends in another project. About $280 with 320 Gb hard drive and 2 GB RAM. Atom 510 processor. Built in Wifi. Nearly silent. Runs F15 nicely. Would one of them suffice ? I have a dual core Atom I use for my home office Asterisk/Samba box. Never seen TOP show over 5% unless I was doing a build or something. I have a Digium Analog card for my POTS lines so I needed a bigger case. I use faxaway.com which costs $12/year for a fax number and essentially unlimited email delivered faxes. Makes my wife happy and I don't have to figure out how to get faxes to work. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
On 08/26/2011 03:17 PM, linux guy wrote: I like the idea of running multiple ATAs with a single base or handset on each line. Something like the Panasonic KX-TG4111B which sells for about $40 for a handset and base. PAP2s sell for about $50 or $25 per line. Total cost of $65 per handset. Comments on this approach ? I use it mixed in with other types of devices. The pap's really are pretty flexible devices for the money. I have one cordless but mostly wired fax, and doorphone. I have a rack of them and each just terminates into the building wiring for phone jacks, so at the user end its just a regular phone plugged in the wall. I used to have t1 card and channel bank, but that became a single expensive point of failure, needs a pc that has pci slots, and really is no more programmable than the sip boxes. So that is why I switched. The cordless handsets I just don't really like for reasons I have gone into already but that doesn't mean they don't work. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
On Friday, August 26, 2011, linux guy wrote: Do any of the DECT systems handle multiple incoming phone lines ? They don't. However, that's not an issue because Asterisk does. Incoming, I have two PSTN lines, three SIP providers, and used to have an IAX2 provider also. Asterisk integrates them all and uses least-cost routing to place outgoing calls through the various trunks. How do the DECT systems integrate with the voice mail services on an Asterisk system ? Exactly the same as all other handsets and softphones. Voicemail comes into my email as well as being available by dialling the appropriate voicemail number from any handset. In another message you mentioned using a PAP2 with a Panasonic phone on each line. This is similar to what I have and it works well for me. HTH, -- Geoff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
On Fri, 2011-08-26 at 07:41 -0700, Steve Edwards wrote: On Fri, 26 Aug 2011, linux guy wrote: How much power does the home asterisk box need ? I use a small box (like those hp thin clients) But these are a bit stronger aluminium housing, instead of plastic, and better foor cooling. Power consumption: 8 Watt under full load CPU: Model: 6.28.2 Intel(R) Atom(TM) CPU Z530 @ 1.60GHz Memory Size: 1 GB Disk /dev/sda: 64.0 GB, 64023257088 bytes This model has just one ethernet port, others have two Size: 10x10 cm hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_sms testers required
On Fri, 2011-08-26 at 14:51 -0500, Kevin P. Fleming wrote: That's pretty harsh, David. Yes, sorry. You're right... especially given that one of my 'obviously correct' fixes to the spool file handling was actually the wrong fix. :) Following the IRC discussion about that, I've now got a working patch to the inotify code which makes it handle hard links correctly. I've filed it against ASTERISK-18331. I certainly don't mean to suggest that the Digium development team isn't doing an excellent job, or that their priorities are wrong. I don't expect you to fix app_sms (or implement app_v110); I'm perfectly happy to do that *myself*. I'm more than happy to be the maintainer for both of those, if you think that's the best way forward. -- dwmw2 smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Friday, August 26, 2011 6:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system On Fri, 2011-08-26 at 07:41 -0700, Steve Edwards wrote: On Fri, 26 Aug 2011, linux guy wrote: How much power does the home asterisk box need ? I use a small box (like those hp thin clients) But these are a bit stronger aluminium housing, instead of plastic, and better foor cooling. Power consumption: 8 Watt under full load CPU: Model: 6.28.2 Intel(R) Atom(TM) CPU Z530 @ 1.60GHz Memory Size: 1 GB Disk /dev/sda: 64.0 GB, 64023257088 bytes This model has just one ethernet port, others have two Size: 10x10 cm Is this a custom build box or does a company sell them preassembled?We are always on the lookout for potential boxes we can use for small installations. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
