Re: [asterisk-users] Old Asterisk Release wanting to upgrade ...
On 17/4/14 3:53 am, Lee, John (Sydney) wrote: I have written a lot of AEL2 script in Asterisk 1.4.x and I am not sure if it will still run in 11. If I'm honest, this is why I still have so many 1.4.x boxes around as well. I've been using 11 for new installs, but the thought of having to redo all the AEL macros from 1.4 does not fill me with any enthusiasm to update those boxes. The switch to Gosub() does not seem to be an easy drop-in replacement for Macro(). Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Live Recording on the Storage Server?
Hello, I am wondering has anyone used Live Recording (monitor or mixmonitor) on to Storage Server via network 1 Gigabit connection? Does it perform well, let say about 50 live recordings at the same time. I am planning to make some system changes at work. I would like to put Asterisk VM on a ESXi host and the datastore will be hosted on Storage Server. On a ESXi host, there will be a few VM's: Asterisk VM Windows Server VM Linux Web Server VM Windows 7 VM What I am concern that users on the workstations will browse their files (home shares) and it may interrupt asterisk live recording because it is shared on the same Storage Server? Cheers. -- _ -- 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] Live Recording on the Storage Server?
hi. I would not do that due to network issues. My approach is to record everything locally and every hour or so to move everything to a storage. On Thu, Apr 17, 2014 at 1:52 PM, Shahid H shah...@gmail.com wrote: Hello, I am wondering has anyone used Live Recording (monitor or mixmonitor) on to Storage Server via network 1 Gigabit connection? Does it perform well, let say about 50 live recordings at the same time. I am planning to make some system changes at work. I would like to put Asterisk VM on a ESXi host and the datastore will be hosted on Storage Server. On a ESXi host, there will be a few VM's: Asterisk VM Windows Server VM Linux Web Server VM Windows 7 VM What I am concern that users on the workstations will browse their files (home shares) and it may interrupt asterisk live recording because it is shared on the same Storage Server? Cheers. -- _ -- 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] DUNDi with SIP Mapping
On Wed, Apr 16, 2014 at 10:20 AM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
You are a bit outside of what I have done, but this looks like it might be
what you want to do with SIP:
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP
I had looked at that guide before, but couldn't get it working. I could do
SIP without authentication. This would have worked if I only wanted to
terminate calls to extensions. For future purposes I wanted to include PSTN
routes. In the end I went with IAX and have it up and running. It was
actually simple to integrate with FreePBX. The important piece was setting
ttl to 1 to prevent DUNDi lookup loops, which would cause the box to
sometimes see its own DUNDi extensions.
The one FreePBX box with the PRI will try 10 digits numbers on DUNDi
private then go out the PRI. The other FreePBX boxes try to dial 10 digit
numbers on DUNDi private then use DUNDi to reach the PSTN. This allows me
to add additionally FreePBX boxes with PSTN connections and use weights.
Additionally providing a separate mapping for the PSTN allows toll free to
first try DUNDi private, then a VoIP provider, then the DUNDi PSTN.
cd /var/lib/asterisk/keys
astgenkey -n `hostname -f`
chown asterisk:asterisk *
share .pub keys between all servers
vim /etc/asterisk/dundi.conf
cachetime=60
ttl=1
priv = dundi-extens,0,IAX2,dundi:${
SECRET}@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial
priv = dundi-dids,100,IAX2,dundi:${
SECRET}@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial
pstn = dundi-via-pstn,400,IAX2,dundi:${
SECRET}@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial
;[EID of remote DUNDi peer]
;model = symmetric
;host = IP or FQDN of remote DUNDi peer
;inkey = public key of remote DUNDi peer, without .pub
;outkey = private key of local PBX, without .key
;include = all
;permit = all
;qualify = yes
vim /etc/asterisk/extensions_custom.conf
[dundi-local]
include = dundi-extens
include = dundi-dids
include = dundi-via-pstn
[dundi-local-keepcid]
exten = _X.,1,Set(KEEPCID=TRUE)
exten = _X.,n,Goto(dundi-local,${EXTEN},1)
[dundi-extens]
include = ext-queues
include = ext-findmefollow
include = ext-group
include = ext-local
[dundi-dids]
include = ext-did-0002
[dundi-via-pstn]
include = outbound-allroutes
FreePBX Trunks
Type: DUNDi
Trunk Name: DUNDi Private
DUNDi Mapping: priv
Type: DUNDi
Trunk Name: DUNDi Pstn
DUNDi Mapping: pstn
Type: IAX
Trunk Name: DUNDi
Outgoing Settings:
Trunk Name: dundi
PEER Details:
type=friend
dbsecret=dundi/secret
disallow=all
context=dundi-local-keepcid
allow=ulawg729
FreePBX Outbound Routes
Route Name: dundi
Route Type: Intra-Company
Dial Pattern: NXXX
Trunk: DUNDi Private
Route Name: outbound
Dial Pattern: 1NXXNXX
Dial Pattern: NXXNXX
Trunk: DUNDi Private
Trunk: PRI or DUNDi Pstn
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Re: [asterisk-users] Live Recording on the Storage Server?
On Thu, Apr 17, 2014 at 7:25 AM, binary dreamer dreamer.bin...@gmail.com wrote: hi. I would not do that due to network issues. My approach is to record everything locally and every hour or so to move everything to a storage. +1 save yourself the headache and do this. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Live Recording on the Storage Server?
On 17 Apr 2014, at 16:14, Paul Belanger paul.belan...@polybeacon.com wrote: hi. I would not do that due to network issues. My approach is to record everything locally and every hour or so to move everything to a storage. +1 save yourself the headache and do this. I'll add another +1 to this. I've never been able to get multi-channel recording (even 3 or 4 channels) working reliably over an NFS link to another server. I suspect, with some tweaking of nfs options it might be possible, but if it ain't broke… Just a cautionary note if you do use a cron job to move recordings to a storage device at regular intervals: make sure you use lsof or similar to check the recordings aren't actually open by asterisk at the time, otherwise interesting things will happen. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Live Recording on the Storage Server?
A simple way that we use to do the move is to create a cron job that looks for a .move file. It has the same name as the recorded file. asterisk writes the .move file which is just a text file with some stats in it. The .move file is written from the dial plan at the end of the recording. In the exten = h we write a .delete file for an abandon call. The cron then processes the .move and .delete files at a given interval. We actually write special instructions into our .move files that the cron parses and can then act accordingly. So we have a single smart cron job handling moves for each type of task. In some cases our .delete files are processed as moves to an abandon cache for recovery if a customer did not intend to abandon it. The sky's the limit on how complex you want to make it, but in the long run it is fairly simple and it just works. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Chris Bagnall aster...@lists.minotaur.cc Sent: Thursday, April 17, 2014 11:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Live Recording on the Storage Server? On 17 Apr 2014, at 16:14, Paul Belanger paul.belan...@polybeacon.com wrote: hi. I would not do that due to network issues. My approach is to record everything locally and every hour or so to move everything to a storage. +1 save yourself the headache and do this. I'll add another +1 to this. I've never been able to get multi-channel recording (even 3 or 4 channels) working reliably over an NFS link to another server. I suspect, with some tweaking of nfs options it might be possible, but if it ain't broke. Just a cautionary note if you do use a cron job to move recordings to a storage device at regular intervals: make sure you use lsof or similar to check the recordings aren't actually open by asterisk at the time, otherwise interesting things will happen. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Old Asterisk Release wanting to upgrade ...
I had little problem converting my AEL scripts from 1.4 to 11 -- _ -- 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] Old Asterisk Release wanting to upgrade ...
On 17/4/14 4:53 pm, Eric Wieling wrote: I had little problem converting my AEL scripts from 1.4 to 11 Did they have lots of macros in them? If so, then you, sir, are a better man than I, and I take my hat off to you :-) (and any hints you might want to share in converting 1.4 AEL macros to 11 would be gratefully appreciated) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Live Recording on the Storage Server?
On Thu, Apr 17, 2014 at 11:52 AM, Bryant Zimmerman brya...@zktech.comwrote: A simple way that we use to do the move is to create a cron job that looks for a .move file. It has the same name as the recorded file. asterisk writes the .move file which is just a text file with some stats in it. The .move file is written from the dial plan at the end of the recording. In the exten = h we write a .delete file for an abandon call. The cron then processes the .move and .delete files at a given interval. We actually write special instructions into our .move files that the cron parses and can then act accordingly. So we have a single smart cron job handling moves for each type of task. In some cases our .delete files are processed as moves to an abandon cache for recovery if a customer did not intend to abandon it. The sky's the limit on how complex you want to make it, but in the long run it is fairly simple and it just works. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 We record locally and move the files to the storage server with a cron job once a minute. The script uses lsof to check to see if Asterisk is writing to the file. /usr/sbin/lsof | grep filename | wc -l Thanks, Ryan -- _ -- 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] DAHDI loading issue on Asterisk
Sean, Yes, it is: asteriskpbx@asteriskpbx:~$ lsmod | grep dahdi dahdi 227741 2 oct612x,wcte43x crc_ccitt 12707 1 dahdi asteriskpbx@asteriskpbx:~$ Do you have the kernel module loaded? lsmod | grep dahdi sean -- _ -- 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] Old Asterisk Release wanting to upgrade ...
Is there a specific item you are having problems with? The Gosub and Macro changes in later versions of Asterisk is mostly transparent to the dialplan if you use AELSub() to call AEL from extensions.conf. The AELSub() dialplan application was written do you don't have to worry about macro .vs. gosub with AEL. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: Thursday, April 17, 2014 12:37 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Old Asterisk Release wanting to upgrade ... On 17/4/14 4:53 pm, Eric Wieling wrote: I had little problem converting my AEL scripts from 1.4 to 11 Did they have lots of macros in them? If so, then you, sir, are a better man than I, and I take my hat off to you :-) (and any hints you might want to share in converting 1.4 AEL macros to 11 would be gratefully appreciated) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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
[asterisk-users] Dimensioning asterisk 11
I will be using a dell R320 Xeon E5-2420 2G and 4G RAM. also using a SIP trunk with ulaw/alaw codec. How many calls could I expect to make at the same time? no transcoding or anything. Just call a number and play a gsm file. Thanks, 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] Dimensioning asterisk 11
On Thu, 17 Apr 2014, Jerry Geis wrote: I will be using a dell R320 Xeon E5-2420 2G and 4G RAM.also using a SIP trunk with ulaw/alaw codec. no transcoding or anything. Just call a number and play a gsm file. How will you do ulaw - gsm without transcoding? How many calls could I expect to make at the same time? A whole bunch? It's hard to give any specifics without the same hardware and workload. Here's a datapoint to consider -- testing an HP ProLiant DL320e Gen8 v2 E3-1240v3 8GB. 9300 passmarks vs your 7300 passmarks. (And only $880 from Newegg.) 2 hosts, 1 originating calls, 1 running a simple dialplan, but similar to the expected production dialplan. 500 'participants' - 100 meetme conferences with 5 calls in each. 3000 'participants' - 100 confbridge conferences with 30 calls in each. Meetme() is still a 'single thread' application so you're done when you max out 1 CPU core. 500 calls was my goal, so that's where testing stopped. The hosts aren't in production yet, so I don't know if my testing experience will match production experience. I would expect playback() (without transcoding) to be significantly less CPU hungry than meetme() or confbridge(). -- 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
[asterisk-users] Realtime in asterisk 12 removed/deprecated?
I was just told that realtime was no longer in asterisk 12, but I find enhancements in 12.2-rc2 and no sign in the wiki that this is true. Can someone comment? -- _ -- 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] Realtime in asterisk 12 removed/deprecated?
All significant changes should be listed in the UPGRADE*.txt included in the Asterisk source code. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell Sent: Thursday, April 17, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Realtime in asterisk 12 removed/deprecated? I was just told that realtime was no longer in asterisk 12, but I find enhancements in 12.2-rc2 and no sign in the wiki that this is true. Can someone comment? -- _ -- 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] Realtime in asterisk 12 removed/deprecated?
Yeah, and I didn't find anything there. I was looking for something a little more concrete that it should be... On 04/17/2014 01:16 PM, Eric Wieling wrote: All significant changes should be listed in the UPGRADE*.txt included in the Asterisk source code. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell Sent: Thursday, April 17, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Realtime in asterisk 12 removed/deprecated? I was just told that realtime was no longer in asterisk 12, but I find enhancements in 12.2-rc2 and no sign in the wiki that this is true. Can someone comment? -- _ -- 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] Realtime in asterisk 12 removed/deprecated?
Bruce Ferrell wrote: I was just told that realtime was no longer in asterisk 12, but I find enhancements in 12.2-rc2 and no sign in the wiki that this is true. Can someone comment? Realtime has not been removed or deprecated. A new model for newly written modules has been created, but nothing existing has been migrated to it or even will be (it's a fundamentally shift). -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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] Realtime in asterisk 12 removed/deprecated?
On 04/17/2014 01:34 PM, Joshua Colp wrote: Bruce Ferrell wrote: I was just told that realtime was no longer in asterisk 12, but I find enhancements in 12.2-rc2 and no sign in the wiki that this is true. Can someone comment? Realtime has not been removed or deprecated. A new model for newly written modules has been created, but nothing existing has been migrated to it or even will be (it's a fundamentally shift). Thanks Joshua! Always good to have a definitive statements -- _ -- 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] Realtime in asterisk 12 removed/deprecated?
Bruce Ferrell wrote: On 04/17/2014 01:34 PM, Joshua Colp wrote: Bruce Ferrell wrote: I was just told that realtime was no longer in asterisk 12, but I find enhancements in 12.2-rc2 and no sign in the wiki that this is true. Can someone comment? Realtime has not been removed or deprecated. A new model for newly written modules has been created, but nothing existing has been migrated to it or even will be (it's a fundamentally shift). Thanks Joshua! Always good to have a definitive statements You're welcome. And fundamentally? What the heck was I thinking. *fundamental* shift. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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] Dimensioning
On Thu, 17 Apr 2014, Jerry Geis wrote:
I was thinking transcoding was through PRI card - not gsm to ulaw. :)
You can convert the GSM files to ULAW using sox. I tend to transcode
everything to WAV (PCM not that funky 'GSM in WAV') because it is
relatively cheap (CPU cycles) to transcode from WAV to ULAW and everything
else in the world understands WAV just fine. If you really need to squeeze
out every last cycle, you can schedule a script to transcode WAVs to ULAWs
as needed.
So if all I am doing is originating calls, and using playback() in the
dialplan - then a system() call on completion I can expect upwards or
3000 concurrent calls?
Based on my unsubstantiated testing on my hosts, that seems like a
reasonable conclusion.
What do you do in the program executed by system()?
How do you actually test to make sure without having 3000 users to call.
Crowdsourcing?
No, it's really pretty simple.
On the 'source' host, I have a call file:
# sample-call-file
channel:sip/test@target
application:playback
data:/tmp/total
# (end of sample-call-file)
And a shell script to create the call files:
# create-calls.sh
cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
# (end of create-calls.sh)
Then, on the 'target' host I have a dialplan snippet:
[public]
exten = test,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = test,n, set(GROUP()=TEST)
exten = test,n, set(ROOM=0${GROUP_COUNT()})
exten = test,n, meetme(${ROOM:-2}, cd)
; exten = test,n, confbridge(${ROOM:-2})
exten = test,n, hangup()
Then, on the 'source' host, I can create calls with this command:
./create-calls.sh number-of-calls-to-create
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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