[asterisk-users] Asterisk 1.8 Streaming MOH timing interface
We are running Asterisk 1.8.5.0 with an uptime of 40 weeks. Just today our streaming music on hold stopped working. I remember when we had first installed 1.8 we had an issue where the streaming music on hold would not work because Music On Hold was using the DAHDI timing module. We needed the DAHDI timing module loaded so that paging would work. However, at that time we upgraded to 1.8.5.0 and the system loaded properly with both the dahdi and pthread timing module with Music On Hold using the pthread timing module. In that state, everything worked properly - Streaming Music On Hold worked as well as Paging. That has all continued to work properly for the last 40 weeks. I'm wondering of for some reason the Music on Hold service is now using the DAHDI timing module because when I do "module show like timing" I see: CLI> module show like timing Module Description Use Count res_timing_dahdi.soDAHDI Timing Interface 33 res_timing_pthread.so pthread Timing Interface 0 2 modules loaded I believe that the pthread used to show a use count of at least 1 with the Music On Hold service using that timing source. I suspec that if I restart the Asterisk service everything will come back up the way that it did last time. However, I'm wondering if there would be a way to switch the Music On Hold module back to using pthread timing without restarting the Asterisk service. Thanks, Bob -- _ -- 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] Polycom and auto answer
Here's what I have and it works for me in 1.8.5: in sip.cfg in extensions.conf exten => 3500,1,SIPAddHeader(Alert-Info: Auto Answer) exten => 3500,n,Page(SIP/3011&SIP/3021&SIP/3110) ; Shortened for example. I actually have about 20 phones here. exten => 3500,n,Hangup This is working fine in an environment with many 330s, some 450s, some 335s and a 550 all running 3.3.1 Hope this helps you out. Bob -- _ -- 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] use ILBC installed from asterisk yum repositories
On Tue, Aug 2, 2011 at 4:39 PM, Lefteris Zafiris wrote: > You can write a short makefile for just codec_ilbc module, build it and > install it on your running asterisk system. You will have to install the > asterisk18-devel package and get the asterisk source code either from > a tar or from the srpm. If you are familiar with the basics of writing > makefiles its pretty trivial to write one that builds codec_ilbc, I have > done this in numerous systems that use the digium rpms and it works > flawlessly. This method can also be used to build other modules that > are missing from the digium rpms. > Thanks for the pointer. I think I'll give this method a try. I'll see if I can figure out how to write the makefile. -- _ -- 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] use ILBC installed from asterisk yum repositories
I would like to try the ILBC codec on one of our systems. The system is currently running Asterisk 1.8.5.0 installed from the Asterisk provided repositories for Centos 5. Is there a process for installing the ILBC codec under this environment, or will I have to un-install the RPMs and build Asterisk from source? Thanks, Bob -- _ -- 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] Macro issue under 1.8.5
I'm still using macro with Asterisk 1.8.5.0 On Fri, Jul 15, 2011 at 3:17 PM, Paul Belanger wrote: > On 11-07-15 02:18 PM, Doug Lytle wrote: >> >> --[ UxBoD ]-- wrote: >>> >>> I back leveled to 1.8.3 and that works fine. What am I missing as >>> app_macro has been installed okay? >> >> Macro was depreciated in 1.6 and most likely removed in 1.8.5 >> > Removed, no. However in future version of Asterisk it will not be enabled > in menuselect by default. > > @OP: *CLI> module load app_macro.so > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://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] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI
I agree that call files are not an appropriate way to solve this. I would like to move back to using the original Page() application which had always worked for us with 1.4 My initial testing found that MOH from a streaming source such as Shoutcast only worked if I disabled the DAHDI timing module in modules.conf with "noload => res_timing_dahdi.so". I understand the DAHDI timing module is a requirement for the Page() application. Has anybody been able to get streaming MOH to work in 1.8.4.3 with the DAHDI timing module loaded? If not, is there a way to use both the pthreads and DAHDI timing modules and force MOH to use pthreads and force Page() to use DAHDI? Bob On Tue, Jun 28, 2011 at 12:03 AM, Faisal Hanif wrote: > Call file are not suitable for you as asterisk process these files in serial > mode (single threaded) and in case of large number of files processing of > last file can be that much delayed that some portion of message may be > already played or the 1st phone may be hanged. > -- _ -- 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 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI
We just finished an upgrade of our Asterisk system to an HA
environment on a pair of servers using Linux-HA. As part of the
upgrade, we also moved to Asterisk version 1.8.4.3
Most things are working quite nicely on the new system. However, I’m
having trouble getting a paging feature to work. In Asterisk 1.4, we
simply used the Page() application like this:
3400,n,Page(SIP/3011&SIP/3021&SIP/3110&SIP/3120&SIP/3121&SIP/3122&SIP/3124&SIP/3125&SIP/3126&SIP/3127&SIP/3221&SIP/3222&SIP/3223&SIP/3250&SIP/3261&SIP/3262&SIP/3310&SIP/3311&SIP/3324&SIP/3329&SIP/3331&SIP/3332&SIP/3350&SIP/3455&SIP/3457)
However, the Page() application seems to rely on the Meetme()
application which also relies on the DAHDI channel driver for mixing
of the audio streams. I have tried using the DAHDI channel driver on
this system, but that seems to make the Music On Hold application use
the DAHDI timing module instead of the pthread module. With the DAHDI
timing module, Music On Hold does not playback Shoutcast streams which
is also a requirement for this system.
As an alternate solution, we have tried implementing a workaround
which simply uses a set of .call files to dial each phone. Those
phones then auto-answer the call and are placed into a conference
bridge on mute using the ConfBridge application. At this point, the
initiating caller speaks the announcement and the phones automatically
hangup after about 10 second. This worked perfectly in our small scale
tests. However, when we ramped this up to the 25 phones that are
required and tested it this morning, somehow this caused the Asterisk
service to restart. I suspect that processing the 25 call files and
placing them into the conference all at the same time somehow made the
system crash and it immediately started up again.
Here's the relevant dialplan:
exten => 3400,1,Answer
exten => 3400,n,playback(beep)
exten => 3400,n,system(cp /etc/asterisk/testPage/*.call
/var/spool/asterisk/outgoing_staging/)
exten => 3400,n,system(mv /var/spool/asterisk/outgoing_staging/*.call
/var/spool/asterisk/outgoing/)
exten => 3400,n,ConfBridge(testPage,1)
exten => 3400,n,hangup
[testPage]
exten => s,1,Answer
exten => s,n,playback(beep)
exten => s,n,Set(TIMEOUT(absolute)=10)
exten => s,n,ConfBridge(testPage,m)
exten => s,n,hangup
exten => _,1,SIPAddHeader(Alert-Info: Auto Answer)
exten => _,n,Dial(SIP/${EXTEN})
exten => _,n,Hangup()
and here's a sample call file:
channel: Local/3011@testPage
callerid: Page
context: testPage
extension: s
priority: 1
archive: no
waittime: 120
Does anyone have insight into how we could accomplish this paging
feature or of anything that we may have missed?
I suspect we could get this all to work with the original Page()
application if there was a way to force MusicOnHold to use the pthread
timing module instead of the Dahdi timing module. Is that configurable
somewhere?
Thanks for your help,
Bob
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Re: [asterisk-users] Aastra, Asterisk 1.4 and Voicemail
On Mon, Mar 8, 2010 at 7:08 PM, Mike wrote: > This seems like a basic thing to set up, so I have no doubt many people have > done this. Anyone care to point me in the right direction? In our config files, we have: softkey1 type: speeddial softkey1 label: "Voice Mail" softkey1 value: *97 This sets up one of the Softkeys on our 480i phones as a speed dial to *97 which takes them to their voicemail. -- _ -- 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] Astricon
> Or charge for full access! Leave a few teasers, and charge some amount to > see them all. I would pay - even close to attendance price... could only > help you get past break even ;) I agree, I would be quite willing to pay for full access to all the videos from the Conference. Bob ___ -- 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] Switchvox HA options
What are the HA options for Switchvox systems? Is it possible to set up redundant systems with DRBD? I know on the digium website they talk about "Optional cold spare failover" What does this mean? Is this an active spare ready for some sort of automated failover? Thanks for you help, Bob ___ -- 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] Dail in modem
On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote: > I am required to do some thing like Dail in modem . > User will have to call a modem just like we do in dail up connection > now we need to handle that request and retrieve some parameters > from that send a HTTp request to a web server and then after getting > http response send user a feed back .. > Why do you need a modem? What will be dialing into the Asterisk system, a human or a machine? ___ -- 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] show pri usage
On Thu, 2009-03-26 at 07:19 -0700, Vieri wrote:
> Maybe I could simply do something like:
> asterisk -rx "show channels" | grep -c -i zap
> to get the number of zap/dahdi channels in use.
I was actually using a command similar to that up until a few months
ago.
/usr/sbin/asterisk -rx 'show channels' | grep '^Zap/[1-9]-\|
^Zap/1[0-9]-\|^Zap/2[0-3]-' | wc -l
That command counted the number of lines that started with one of the
first 23 Zap channels
Now, I'm using phpagi to monitor from another server. This script polls
both in and out usage on those 23 zap channels where $data1 is In and
$data2 is Out.
connect()) {
$asm->events('off');
$channels = $asm->command("show channels concise");
$channels = $channels['data'];
$channelRows = explode("\n",$channels);
while($row=array_shift($channelRows)){
$rowDetails=explode('!',$row);
if(substr($rowDetails[0],0,3) == 'Zap'){
$zapChannel=substr($rowDetails[0],4,(strpos($rowDetails[0],'-')-4));
if($zapChannel<24){
if(substr($rowDetails[7],0,1) == '9') $data2++;
else $data1++;
}
}
}
$asm->disconnect();
}
echo "$data1!$data2";
?>
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Re: [asterisk-users] Aastra 480i repair?
On Fri, 2009-03-06 at 09:41 -0500, m...@njycamps.org wrote: > Anyone know where I can get an Aastra 480i repaired? The phone works > on speakerphone, but when you lift the receiver offthe hook, the phone > does not engage. There is something wrong with the "hook". The > receiver works fine, on another phone, the phone has had its firmware > flashed, and again, the phone wroks fine using the speakerphone. I hope you don't have too many of those. we bought 80 of them for our office, and I'm sure that 20 of them have already had problems with their off-hook sensor. Even phones that have worked good in the past are now starting to fail. I hear aprox. 1 new phone failure report each week. Now that our phones are off warranty, I've had to ship them straight to Aastra. They recently repaired 7 of them for me for $25 each. Now I'll just wait and see if the fix is really a fix or if they are going to fail again. I dealt with Linda at Aastra: Linda Berendt Needless to say, we're not buying Aastra phones anymore. We're pretty happy with Polycom phones right now. Bob ___ -- 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 Spectralink 8002 Configuration
On Wed, 2009-02-25 at 11:37 -0500, M Hulber wrote: > So I'm thinking, would this work if I had a sip_.conf as well as a > sip_.conf? What the relationship between the LINEs in the > sip_.cfg and the Reg on the phone? What's the relationship > between the AUTH and the LINEn_AUTH? This is just a bit confusing to > me. > > Basically, I want to treat the phone as a multiple extension phone > instead of a single user phone. Where each extension (LINE) represents > itself as a unique peer when communicating with Asterisk and is > registered uniquely. OK, so the confusing thing that was not documented by Polycom is this: At the bottom of page 46, the grey box mentions that each handset needs a config file (which I expected), but it does not clearly state why you would name them sip_JohnDoe.cfg or sip_3001.cfg - This was a little counter intuitive for me until I realized it was related to a username that was entered in the phone's menu. So if you enter the user on the phone as it will pick up the sip_.cfg file and if you enter on the phone it will pick up the sip_.cfg file. I think you would want to break your config out into two files like this: sip_.cfg: AUTH = ; secret LINE1 = LINE1_PROXY = 1 LINE1_CALLID = ABC Tech LINE1_AUTH= ; secret LINE2 = 5 LINE2_PROXY = 1 LINE2_CALLID = ABC Tech LINE2_AUTH= ; secret sip_.cfg: AUTH = ; secret LINE1 = LINE1_PROXY = 1 LINE1_CALLID = ABC Sales LINE1_AUTH= ; secret LINE2 = LINE2_PROXY = 1 LINE2_CALLID = ABC Sales LINE2_AUTH= ; secret Or, you could leave it like you have and have the phone register to both extensions at the same time. I'm not sure what you should do with the first line in that case. Bob ___ -- 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 Spectralink 8002 Configuration
On Wed, 2009-02-25 at 15:13 +, Jeff LaCoursiere wrote: > Aha! Mind posting that config? My sip_allusers.cfg looks like this: CODECS = g711u, g711a PROXY1_TYPE = Asterisk PROXY1_ADDR = 192.168.8.1:5060 #PROXY1_KEYPRESS_2833 = enable PROXY1_KEYPRESS_INFO = disable PROXY1_HOLD_IP0 = disable #PROXY1_PRACK = enable PROXY1_REREG_SECS=3600 PROXY1_KEEPALIVE_SECS=14 PROXY1_DOMAIN = 192.168.8.1 PROXY1_CALLID_PER_LINE = disable PROXY1_MAIL_ACCESS = *97 My sip_.cfg looks like this: AUTH = ; secret LINE1 = LINE1_PROXY = 1 LINE1_CALLID = NOC Tech LINE1_AUTH= ; secret LINE2 = LINE2_PROXY = 1 LINE2_CALLID = NOC Tech LINE2_AUTH= ; secret Bob ___ -- 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 Spectralink 8002 Configuration
Mark, Are you still having trouble with your 8002? I had a lot of trouble with mine initially, but after playing with it for about 8 hours I figured it out. Now it works great all around our office. Our NOC technician loves it! There is a problem with the sample configs that Polycom publishes. I started by un-commenting and modifying the portions that related to an Asterisk setup. However, that seemed to be the source of my problem in the end. I don't know if the phone simply can't parse the length of the sample file, or if there are some errors in the sample file that I missed. As soon as I trimmed the config file down to just the necessary components, the phone started to work! Bob On Mon, 2009-02-23 at 21:07 -0500, M Hulber wrote: > I have a new Polycom Spectralink 8002 and am having trouble with the > configuration or the unit but I can't see what's wrong. The unit does > not seem to even attempt to register with the Asterisk proxy but I can > make calls to it. I have viewed the syslog from the device which it > will actually write to the asterisk server so I know it can be reached. > I have also run a sip debug and see no registration traffic from the > unit. It also pulls the configs from the tftp server on the asterisk > box ok. > > Does anyone have a sample set of configs that work? I have samples for > the Polycom side but haven't seen the match on the asterisk side. Since > I don't even see traffic, I can't think that it's even an authentication > issue. > > When I dial from the device it just sits there, basically. > > MARK. > > -- > > sip_allusers.cfg: (I've tried most variations on theses settings) > > ## FOR PROXY1_TYPE = ASTERISK > > #PROXY1_ADDR = 192.168.2.80:5060# replace the ip address with > the Asterisk Server's Address > PROXY1_ADDR = 192.168.2.80 # replace the ip address with the > Asterisk Server's Address > PROXY1_KEYPRESS_2833 = enable > PROXY1_KEYPRESS_INFO = enable > PROXY1_HOLD_IP0 = disable > PROXY1_PRACK = enable > #PROXY1_REREG_SECS=3600 > PROXY1_REREG_SECS=35 > PROXY1_KEEPALIVE_SECS=14 > #PROXY1_DOMAIN = asterisk# Replace this with your SIP Domain's name > PROXY1_CALLID_PER_LINE = disable > PROXY1_MAIL_ACCESS = 864 # Put Your Voice Mail Sytem's > Pilot Number here > > sip_2000.cfg: > > LINE1 = 2000 > LINE1_PROXY = 1 > LINE1_CALLID = 2000 > #LINE1_AUTH= 2000; 2000 > > sip.conf: > > ; Polycom Spectralink 8002 > [2000] >type=friend >host=192.168.3.123 >;port=5060 >secret=2000 >username=2000 >;fromuser=2000 >;authuser=2000 >qualify=no ; turned this off to stop asterisk side initiated traffic >context=spectra_default >dtmfmode=rfc2833 >disallow=all >allow=ulaw >mailbox...@default >canreinvite=yes >callgroup=1 >pickupgroup=1 >accountcode=Home >nat=no > > > Syslog: > > Feb 23 20:25:06 192.168.3.123 Jan 1 00:18:24.57 0090.7a0a.13f3 > (192.168.003.123) [0007] Call start, AP 0014.d1c2.70fe (-32 dBm) > Feb 23 20:25:09 192.168.3.123 Jan 1 00:18:26.87 0090.7a0a.13f3 > (192.168.003.123) [0008] Number Abufs: 26 > Feb 23 20:25:09 192.168.3.123 Jan 1 00:18:26.87 0090.7a0a.13f3 > (192.168.003.123) [0009] Number Fbufs: 2 > Feb 23 20:25:09 192.168.3.123 Jan 1 00:18:26.88 0090.7a0a.13f3 > (192.168.003.123) [000a] Max Number Abufs: 359 > Feb 23 20:25:09 192.168.3.123 Jan 1 00:18:26.88 0090.7a0a.13f3 > (192.168.003.123) [000b] Max Number Fbufs: 33 > Feb 23 20:25:11 192.168.3.123 Jan 1 00:18:29.57 0090.7a0a.13f3 > (192.168.003.123) [000c] NStat: 0014.d1c2.70fe (-30 dBm), Tx 3704, Rx > 43841, BTx 2, BRx 2766, MTx 0, MRx 0, Tx Drop 3 (0.1%), Tx Retry 96 > (2.7%), Rx Retry 19 (0.0%) > Feb 23 20:25:16 192.168.3.123 Jan 1 00:18:33.87 0090.7a0a.13f3 > (192.168.003.123) [000d] Number Abufs: 46 > Feb 23 20:25:16 192.168.3.123 Jan 1 00:18:33.87 0090.7a0a.13f3 > (192.168.003.123) [000e] Number Fbufs: 3 > Feb 23 20:25:16 192.168.3.123 Jan 1 00:18:34.57 0090.7a0a.13f3 > (192.168.003.123) [000f] NStat: 0014.d1c2.70fe (-36 dBm), Tx 3707, Rx > 43996, BTx 2, BRx 2773, MTx 0, MRx 0, Tx Drop 3 (0.0%), Tx Retry 96 > (0.0%), Rx Retry 19 (0.0%) > Feb 23 20:25:21 192.168.3.123 Jan 1 00:18:39.57 0090.7a0a.13f3 > (192.168.003.123) [0010] NStat: 0014.d1c2.70fe (-36 dBm), Tx 3708, Rx > 44284, BTx 2, BRx 2792, MTx 0, MRx 0, Tx Drop 3 (0.0%), Tx Retry 96 > (0.0%), Rx Retry 19 (0.0%) > Feb 23 20:25:26 192.168.3.123 Jan 1 00:18:44.36 0090.7a0a.13f3 > (192.168.003.123) [0011] Call end, AP 0014.d1c2.70fe (-36 dBm) > > > > > ___ > -- 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
Re: [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5
I found that after moving to Asterisk 1.6 and the latest SVN of ASterisk-GUI, the link changed from: http://localhost:8088/asterisk/static/config/cfgbasic.html to: http://localhost:8088/static/config/cfgbasic.html I don't know if you'll find the same or not... Bob On Mon, 2009-02-23 at 22:17 +0100, Tamer Higazi wrote: > Hi people! > I am not getting really smart. I get the SVN Edition of asterisk GUI > interface, compiled and love to get it to run, what won't work. What am > I doing wrong?! > > svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0 > > make > make checkconfig > make install > > > and If I open one of the URLs: > http://localhost:8088/asterisk/static/config/cfgbasic.html > http://127.0.0.1:8088/asterisk/static/config/cfgbasic.html > > I always get 404 not found! > > > For any advises I would thank you. Here is the manager.conf and http.conf > > manager.conf: > > [general] > displaysystemname = yes > enabled = yes > webenabled = yes > port = 5038 > httptimeout = 60 > bindaddr = 0.0.0.0 > > > > [administrator] > secret = ** > read = system,call,log,verbose,command,agent,user,config > write = system,call,log,verbose,command,agent,user,config > > > and the http.conf: > > enabled=yes > enablestatic=yes > bindaddr=0.0.0.0 > bindport=8088 > > > ___ > -- 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] GUI interface to manage business edition
On Fri, 2009-02-13 at 15:24 -0500, Miguel Martinez wrote: > I am new to the Asterisk world, but have decided to use the business > edition, but am looking for a cost effective gui interface to manage > the software. Does the Asterisk-GUI work with Asterisk Business Edition? ___ -- 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] Broadcast Phone system (for radio)
this link: http://www.telos-systems.com/techtalk/digiphones/digiphones_4.htm States the following: "Generic PBXs will not do for our broadcast application – they just don’t have the features necessary. For example, while lines may certainly be shared to multiple phones, there is no way to switch groups of lines from studio to studio. There is also no way to connect computers for call-screening applications. On the audio side, there is no adaptive hybrid or professional audio outputs. Usually, there is only one or two “Music on Hold” inputs for the entire unit, while we need one for each studio. While you could use a PBX to derive analog lines for the studio telephone interface gear, it will be far superior to make a direct all-digital link. So we will need something like a PBX, but specialized for broadcast." Our company owns 2 radio stations, and they are looking at a new on-air phone system. At the same time, we are looking at installing an Asterisk system for their office PBX. Does anyone know of an asterisk based solution for this type of application? I'm pretty certain Asterisk could handle all the special requirements that this article is claiming a "Generic PBX" can't do. ___ -- 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] Asterisk Appliance
I'm looking for some info on the Asterisk Appliance. I understand it has a gui, but can I still do all the dialplan config that I'm used of doing by hand outside of the gui? If I really wanted to, could I even ignore that the device has a gui and do all my config in the files? I guess I'm just wondering if it will be as flexible as a 'vanilla' asterisk install from source on a linux system. Also, from those who are using these devices, what has your experience been? Are they stable? Do they seem to have enough horsepower and storage space for an SMB with up to 50 phones? Some older specs stated they would be appropriate for businesses with 2-50 users, while the current spec on the Digium site states they are appropriate for 2-20 users. The application I'm thinking of would be VoIP only with a g.711 SIP trunk and g.711 phones. Thanks for your input. Bob ___ -- 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] Asterisk as MGCP client
Has there been any work done on using Asterisk as an MGCP client? I see the 'Asterisk MGCP channels' page on voip-info hasn't been updated since 2006, and I was wondering if anyone has been able to accomplish this yet. I have a situation where it might be helpful to have an Asterisk system register to a local phone service using MGCP. Thanks for your help. Bob ___ -- 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] SendImage() to Polycom ip550 or ip670
I tried really quickly the other day to send an image to these phones from the dialplan like this: exten => 2821,n,SendImage(/var/lib/asterisk/images/asterisk-intro) or exten => 2821,n,SendImage(asterisk-intro) It didn't work for me. Should this work? Is anyone else using this with Polycom Phones? Bob ___ -- 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] Call quality issue across VPN-> POTS vs SIP
On Mon, 2008-11-03 at 13:17 -0500, Lincoln King-Cliby wrote: > It's conceivable, but how would I verify this and how would I change > it if that was the problem? There's a few things you can do here. 1) Check the sip.conf on both sides to see what is defined there for the trunk. Look for some disallow and allow statements. If they are there, that will tell Asterisk what codecs to use on that trunk. 2) You could also check the codec that is in use during a call by looking at the sip channel. From the asterisk CLI, start with "show channel SIP/" and tab it out to complete the command showing the trunk between your two systems. I believe the codecs are listed here as "NativeFormats" and "ReadFormat". You could check this under both of your scenarios to see if there is a different codec in use. 3) If you'd like to try and force the use of a compressed codec such as GSM between your two sites, you would just need to make sure that both sides had the following lines in the definition for the trunk in sip.conf and then do a 'reload chan_sip.so" from the Asterisk CLI: disallow=all allow=gsm ___ -- 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] Call quality issue across VPN-> POTS vs SIP
On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: > Any ideas why the audio quality would be so markedly different when > the only thing that seems to be different is where the call is > originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- 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] hex b1 in CallerID sent by Asterisk On PRI
On Wed, 2008-10-22 at 12:11 -0500, Bob Pierce wrote:
> On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote:
> > Does anyone know what the significance is of the b1 being sent here?
> >
> > Or, is there a way to make Asterisk not send the b1 character as a
> > test?
As a further update to this, I've noticed the following in q931.c at
about line 1236:
static FUNC_SEND(transmit_display)
{
int i;
if ((pri->switchtype == PRI_SWITCH_QSIG) ||
((pri->switchtype == PRI_SWITCH_EUROISDN_E1) && (pri->localtype ==
PRI_CPE)) ||
!call->callername[0])
return 0;
i = 0;
if(pri->switchtype != PRI_SWITCH_EUROISDN_E1) {
ie->data[0] = 0xb1;
++i;
}
memcpy(ie->data + i, call->callername, strlen(call->callername));
return 2 + i + strlen(call->callername);
}
So, I think this is where the b1 is being added.
My question then is, what is the significance of this character?
What's the best way to try sending caller name without this character?
Should I just try changing my switchtype to euroisdn at both sides of
the link?
Bob
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Re: [asterisk-users] hex b1 in CallerID sent by Asterisk On PRI
On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote: > Does anyone know what the significance is of the b1 being sent here? > > Or, is there a way to make Asterisk not send the b1 character as a > test? As an update to this, I noticed the following lines in libpri.h near line 236: /* Network Specific Facilities (AT&T) */ #define PRI_NSF_NONE -1 #define PRI_NSF_SID_PREFERRED 0xB1 #define PRI_NSF_ANI_PREFERRED 0xB2 So, I've tried specifying nsf=none in zapata.conf, but we still see the b1 preceding the caller name in the pri trace. However, I only did a 'reload chan_zap.so' since this is a production system. Should that have changed the nsf settings for this span? My current zapata.conf is pasted below if that helps... Bob [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A104 port 1 [slot:4 bus:13 span:1] switchtype=national context=inbound group=1 signalling=pri_cpe channel =>1-23 ;Sangoma A104 port 2 [slot:4 bus:13 span:2] context=stations group=0 signalling=fxo_ks channel => 25-48 ;Sangoma A104 port 3 [slot:4 bus:13 span:3] switchtype=national nsf=none context=metaswitch group=2 signalling=pri_cpe channel =>49-71 ___ -- 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] hex b1 in CallerID sent by Asterisk On PRI
I'm trying to send CallerID info to a MetaSwitch system over a PRI. The MetaSwitch gets the info exactly as it is sent by Asterisk, but I think it might be having trouble with the Hexadecimal b1 that is being sent just before the first character of the CallerID Name. Does anyone know what the significance is of the b1 being sent here? Or, is there a way to make Asterisk not send the b1 character as a test? I've pasted a portion of the PRI debug trace below. Thanks for your help. Bob -- Making new call for cr 32985 > Protocol Discriminator: Q.931 (8) len=55 > Call Ref: len= 2 (reference 217/0xD9) (Originator) > Message type: SETUP (5) > [04 03 80 90 a2] > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) > Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) >User information layer 1: u-Law (34) > [18 03 a9 83 97] > Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 >ChanSel: As indicated in following octets > Ext: 1 Coding: 0 Number Specified Channel Type: 3 > Ext: 1 Channel: 23 ] > [1e 02 80 83] > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) > Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] > [28 0a b1 54 65 73 74 20 4e 61 6d 65] > Display (len=10) Charset: 31 [ Test Name ] > [6c 0c 21 80 34 30 33 35 35 35 31 32 31 32] > Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) > Presentation: Presentation permitted, user number not screened (0) '4035551212' ] ___ -- 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] FAX over T1 Question
On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote: > I have a Sangoma A104d T1 card, a Rhino 24-port FXS box, and am > running > Asterisk 1.4.21.2 I think you're mostly right on this setup, but I wonder if your A104d is doing some hardware echo cancellation on these calls. If I'm not mistaken, that can mess up fax machine communications. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is shared_lastcall available in 1.4
On Wed, 2008-08-27 at 11:21 +0300, Atis Lezdins wrote: > If you doubt about some part, you're welcome to ask, i'll try to help > you, but i don't want to provide complete backport to you, as i won't > be able to test it :) Thanks Atis, I'll probably try this in a few weeks when I start rebuilding the permanent system that will replace our current temporary system. That should give us the opportunity to test it on the bench instead of playing around with the production box. I'll probably be back to ask for help. Have a great day, Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is shared_lastcall available in 1.4
On Tue, 2008-08-26 at 17:53 +0300, Atis Lezdins wrote: > > Are there any plans to back port this feature into upcoming 1.4 > > releases? > > > > No, new features are added only in trunk, and released in next major > release (1.6). So what would be involved in back porting this feature for our system? Do I simply follow the diff from the link you provided and apply the highlighted changes to the app_queue.c file in my Asterisk source directory before recompiling? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is shared_lastcall available in 1.4
On Tue, 2008-08-26 at 17:30 +0300, Atis Lezdins wrote: > I'd say - go for backport instead. shared_lastcall is commited in > http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820&r2=86985 > and it seems that there are no bugfixes for it since. So, backporting > should be fairly simple. Also i would suggest subscribing to > asterisk-svn and watch for commits to app_queue to not miss any > bugfixes to it. Are there any plans to back port this feature into upcoming 1.4 releases? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is shared_lastcall available in 1.4
On Mon, 2008-08-25 at 17:47 -0500, Bob Pierce wrote: > > I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately, > the > > shared_lastcall option is only in versions 1.6.0 and up. > > > > Does anybody have a workaround for this in 1.4? Or maybe a better question: How stable is 1.6 for production use? Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is shared_lastcall available in 1.4
On Mon, 2008-08-25 at 17:46 -0500, Mark Michelson wrote: > I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately, the > shared_lastcall option is only in versions 1.6.0 and up. > Does anybody have a workaround for this in 1.4? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is shared_lastcall available in 1.4
We have just moved up to Asterisk 1.4.21.2 from 1.2.18 We are now dynamically adding and removing members from our queues. Just like before, our members are shared across multiple queues. Today was our first day with the system in full production, and the only minor glitch that I've run across is that our queue members are mentioning that the queues don't seem to be honouring the wrapuptime. I think I'm running into a similar problem to what is described here: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9777 Has the general option "shared_lastcall" been added to 1.4, or will I need to wait for 1.6? Let me also say, great work guys! I am very happy with some of the improvements I've already seen in 1.4. It was a little bit of work to work around the deprecated agentcallbacklogin function, but it was worth it! Queues are behaving so much better now, and the AutoFill parallel call distribution is really nice. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reverse Scenario
> I know you cannot describe the whole scenario in an email, what I need > is a line or some words for each step :) Or if anyone can do the > whole scenario, please send me an email for further discussions. > Would a combination of call files and conference rooms achieve this? Quick thoughts are: 1) Have box 1 put the caller side in a conference room and send a request to an application on box 2. 2) Application on Box 2 generates a call file for a call between the called side and a conference room on Box1. You may be able to do this even cleaner without the conference room on box 1, but I can't think of how to do that right now. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip or IAX device with professional balanced audio out
Hi all, I've been googling for a solution here and haven't really come up with anything yet. We're doing an Asterisk install for a local radio station, and we're looking for a phone that they can use in their control room hooked up to their mixer board for recording calls. So, when you phone in for some contest or to request a song they record it and play it back a few minutes later on the air. They are currently recording calls from a hacked pots phone, but I was hoping for something a little more elegant with their new system. Has anyone run across a solution that might work nice here, or is there some other way of tackling this problem that I may have overlooked? Thank for your suggestions. Bob ___ -- 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] multiple simultaneous access to single voice mail box
On Thu, 2008-04-10 at 11:25 -0500, Tilghman Lesher wrote: > If you instead use a separate extension, you can use groups to > restrict the number of people accessing a particular mailbox: > Thanks Tilghman, I didn't think of that. I'm sure that will work just fine for what we need. Have a great day. Bob ___ -- 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] multiple simultaneous access to single voice mail box
We are using Asterisk 1.2.18 at this site. One of the users brought this to my attention today. "We have a problem when we take the message off the voice mail. If I am taking off the messages it used to be [on the old phone system] that no one else was able to go in & take off the message. Now I can be taking off the messages & some one else can also be taking off the same messages. We should not be able to do this!!" Has anyone else seen this? Is there a way to setup the voice mail so that each box can only be accessed by one person at a time? Bob ___ -- 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] BRI card with PCI-E interface
On Fri, 2008-02-01 at 15:01 +0100, Alberto Pastore wrote: > Olivier ha scritto: > > Hi, > > > > Does such card exist ? > > It seems all existing models are designed for PCI buses. > > > > Regards > > > > > > > > > > ___ > > -- 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 > > The Sangoma A500BRX is a 2 to 6 bri pci-x interface, > although I've never tested it > (A500BRECX comes with hw echo cancellation). > We've been using the Sangoma A104DE in production for almost a year now and it works great. Bob ___ -- 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] Semi-OT: Best Speakerphone
We're using Aastra 480i phones, and their speakerphone is great. I even have one in our datacenter and the speakerphone is usable even with all the noise of the server fans. I also have a great contact if you happen to be in Canada wanting to buy some. Bob On Mon, 2007-11-26 at 12:30 -0700, Ken Williams wrote: > I'm looking for recommendations on speakerphones for a conference > room. > > We're using Grandstream GXP-2000 which we've been very happy with on > all accounts, except the speaker phone. Speaker phones on these units > are extremely bad, picking up any and all background as well as having > full-duplex issues (that is, when the other end is talking you can't > talk over them to interrupt or whatever). > > So, before I go buy random phones for testing I thought I'd get some > recommendations. > > Thanks, > Ken > ___ > --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] Asterisk on Pcengines Alix board
On Sun, 2007-11-18 at 22:14 +0100, Giuseppe Barichello wrote: > I have successfully compiled and installed Asterisk on an Alix board > (AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian > variant). > I'm using it at home for a month. > That's very interesting! I've been curious about trying this. Did you run across any challenges getting this setup? Bob ___ --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] Queue Statistics reporting
Anyone know of a good package for reporting on Queue statistics from Asterisk? Bob ___ --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] reload manager.conf
On Wed, 2007-10-24 at 13:31 -0700, Richard Lyman wrote: > every time there is a new connection to the asterisk manager > interface, the manager.conf file is reread. > (meaning, it reloads itself) Great. Thanks for your help! ___ --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] reload manager.conf
I've made a change to my manager.conf file in asterisk 1.2.18 Is there a way to reload that config file from the CLI without restarting asterisk? Bob ___ --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] Which Asterisk version to use?
On Thu, 2007-09-27 at 11:48 -0400, Eric B. wrote: > I'm a complete newbie to Asterisk and have been reading through > documentation and sites for the last couple of hours trying to > understand what to do to start my learning curve with Asterisk, and am > very confused. The best starting point IMHO is simply to buy the new O'Reilly Book "Asterisk - The Future of Telephony" and follow the instructions there to install and configure Asterisk 1.4 on top of your favourite Linux Distro. Bob ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
