Re: [asterisk-users] asterisk as IVR using 3g usb modem
On Thu, 2012-10-18 at 15:45 +, Mahendra Dobariya wrote: hi, I want to use asterisk as IVR system , but to make and receive GSM call, i want to use 3g usb modem.(voice enabled) http://www.huaweidevice.co.in/Products/MobileBroadband/E303c.php and i want to install this system on two different machine 1 on mac os x - 2 raspberry pi- (debian wheezy)--http://www.raspberrypi.org/ thanx in advance.. Are you very sure about the last one (i.e. the r-pi)? These have a very few resourses (cpu, mem) If looking for something small, how about latest pandaboard, a bit more expensive, but less limited: http://www.hardware-modules.com/index.php?page=Browseproduct_type=SBCdesigner=Texas%20Instrumentmodule=Pandaboard%20ES%20(Texas%20Instrument%20-%20OMAP4460)lang=en And still cheaper than most intel-based sff-boards. 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] motif and psi - no sound
19.10.2012 08:40, Dmitry Melekhov пишет: Hello! I'm trying to use psi+ to conect to asterisk using chan_motif and vise versa. Connection looks good, but no sound. As I see there is some traffic (22.229 is my desktop with psi) 08:38:37.463506 IP 192.168.22.229.8010 192.168.22.19.17012: UDP, length 82 08:38:37.481325 IP 192.168.22.229.8010 192.168.22.19.17012: UDP, length 82 08:38:37.481885 IP 192.168.22.19.17012 192.168.22.229.8010: UDP, length 65 08:38:37.501745 IP 192.168.22.19.17012 192.168.22.229.8010: UDP, length 65 but I hear nothing :-( psi-psi works OK. And I have speex16 allowed in motif.conf Any ideas? Thank you! btw, I forget to mention that this is 11-rc2 :-) -- _ -- 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] audio fades out over PRI
On 10/16/2012 10:24 AM, Jerry Geis wrote: I am using asterisk 1.4.43. When I call over the PRI to a single phone and play my recorded message its heard just fine. When I call over the PRI to a single extension (the switch then takes 3 phones offhook in intercom mode) and play my same recorded message the audio is dropping out and the whole message is not heard. What might this be? I think its the other switch - but if we use that switch to call that same extension and speak the audio does not drop out at all What can I look at? Jerry Seems is though this was just the AGE OLD PROBLEM of interrupt sharing on the digium cards. Disabling the serial, parallel, and HDMI on board audio corrects the situation as the card now gets its own interrupt. Jerry -- _ -- 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] AstDB with Sqlite
Just use the standard dialplan db commands - Read
Set(TESTOP=${DB(Nightop/ext)}) write Set(DB(Nightop/ext)=107). It's not as
robust as Mysql or postgres but does seem to do better than the old Berkley
database.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Robinson
Sent: Thursday, October 18, 2012 10:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AstDB with Sqlite
As you may know, asterisk version 10 and high use sqlite. Are any examples
or documentation how to use in dialplan?
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Re: [asterisk-users] Agents in more than one queue at once
In general there is no guaarantee as which call will connect; each queue is independent AFAIK. l. 2012/10/17 Alex Forster a...@alexforster.com My company has been running Asterisk 1.6.2.19-1_centos5 from the official yum repo, and for a while now I've been receiving complaints from our call centers about calls not being routed in the most efficient order. I'll explain with a simplified scenario-- Let's say I have two queues: A and B. I have one agent, Alice, who is a member of both of these queues. While Alice is busy on a call, one person calls in to queue A, and then, several moments later, another person calls in to queue B. At this point, note that both callers waiting on hold are position 1 in their respective queues. A queue show might look like this... A has 1 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s Members: 21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls (last was 533 secs ago) Callers: 1. SIP/Trunk-eb17 (wait: 1:14, prio: 0) B has 1 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s Members: 21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls (last was 533 secs ago) Callers: 1. SIP/Trunk-eb1e (wait: 0:45, prio: 0) My question is: when Alice gets off the phone, which call will she get? My expectation is that she will get the call which has been waiting longer, but I'm not sure that's actually the case. Alex Forster -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- 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 error message so uncommon, not even Google knows abuot it
I'm setting up a test server with a Digium TE122 and am getting the following error on the console, spewing as fast as it can. Does anyone have any idea what this error might be? [Oct 19 11:24:53] NOTICE[2076]: chan_dahdi.c:3108 my_handle_dchan_exception: PRI got event: Event 59 (59) on D-channel of span 2 -- _ -- 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 error message so uncommon, not even Google knows abuot it
On Fri, Oct 19, 2012 at 11:28 AM, Eric Wieling ewiel...@nyigc.com wrote: I'm setting up a test server with a Digium TE122 and am getting the following error on the console, spewing as fast as it can. Does anyone have any idea what this error might be? [Oct 19 11:24:53] NOTICE[2076]: chan_dahdi.c:3108 my_handle_dchan_exception: PRI got event: Event 59 (59) on D-channel of span 2 You have two D channels, why? Some more info would help, like configs and where the PRIs are coming from. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Tell apart between network disruption and asterisk restart via AMI
I have a program that connects to the Asterisk Manager Interface through port 5038 on a remote machine. Suppose I get a TCP disconnection on my program. The program will then attempt to reconnect to the AMI and will eventually succeed. Is there a way to check whether the disconnection was caused by a network disruption, or an Astersk restart/crash? In other words, is the Asterisk process I contacted now the same as the one I was connected before, or is it a different one? The reason I want to know is that I have a cache of information that is costly to parse (scales linearly with the number of extensions) and I want to know how to realize that the information is now stale. -- _ -- 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] Tell apart between network disruption and asterisk restart via AMI
From AMI you can get uptime. If the uptime is short likely Asterisk restarted. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Oct 19, 2012, at 10:31 AM, Alex Villacís Lasso wrote: I have a program that connects to the Asterisk Manager Interface through port 5038 on a remote machine. Suppose I get a TCP disconnection on my program. The program will then attempt to reconnect to the AMI and will eventually succeed. Is there a way to check whether the disconnection was caused by a network disruption, or an Astersk restart/crash? In other words, is the Asterisk process I contacted now the same as the one I was connected before, or is it a different one? The reason I want to know is that I have a cache of information that is costly to parse (scales linearly with the number of extensions) and I want to know how to realize that the information is now stale. -- _ -- 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] Tell apart between network disruption and asterisk restart via AMI
On Fri, Oct 19, 2012 at 12:31 PM, Alex Villacís Lasso a_villa...@palosanto.com wrote: I have a program that connects to the Asterisk Manager Interface through port 5038 on a remote machine. Suppose I get a TCP disconnection on my program. The program will then attempt to reconnect to the AMI and will eventually succeed. Is there a way to check whether the disconnection was caused by a network disruption, or an Astersk restart/crash? In other words, is the Asterisk process I contacted now the same as the one I was connected before, or is it a different one? The reason I want to know is that I have a cache of information that is costly to parse (scales linearly with the number of extensions) and I want to know how to realize that the information is now stale. In the CLI, you can run `core show settings` which will tell you the startup time of the server. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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 1.6.0 disable cdr account logs?
Hi All, I would like to disable the cdr account logs but in 1.6.0 but the 'accountlogs=no' switch is not available till 1.8 as far as I can tell. Is the any switch I can turn off int he Mkae file for the cdr_csv.so module to disable accountcode logs? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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
