Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: SARK devs on April 12, 2006, 05:01:49 PM
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Hi everyone.
Build 173 is now up on our FTP server. Here are the highlights...
- Added call forward agi entry point so you can invoke call forwarding from custom apps. e.g.
exten=>s,1,agi(selintra,CFToggle,{type},{fromnum},{tonum})
type can be CFIM (immediate) or CFBS (on busy or no answer)
Examples:-
Forward 5002 immediate to 5003
exten=>s,1,agi(selintra,CFToggle,CFIM,5002,5004)
Forward 5002 immediate to Voicemail (aka DND)
exten=>s,1,agi(selintra,CFToggle,CFIM,5002,5002)
*in keeping with our recursive roots a logical CF to voicemail is a CF to self! (programmers should steer clear of humour - they really should).
Clear any Immediate Forward on 5002.
exten=>s,1,agi(selintra,CFToggle,CFIM,5002,)
- Added a new follow-me function (*27*) - this is internal only in this release
*27*{from extension}
This will "follow" you to the phone from which it is invoked - sort of a remote CFIM.
- Physically removed trunk definitions from sip.conf and iax.conf if inactive
- Made trunk inbound routes mutually exclusive (biggish change to trunks)
- Upgraded FOP to handle Queues and 64 entities
- Added new carrier "FreeIpCall"
- cleaned up autosniff and added to the rpm - you should no longer need to run autosniff manually unless you add new hardware.
- fixed port opening issues (again!)
- Fixed fail over bug
- Begun work to support Mitel 52XX series phones (not trivial this - it missed the cut, contact us if you have one of these units and we’ll tell you how to at least make and receive calls)
- Fixed bug with speed dial to external numbers
- RPM now invokes conf-asterisk as its last step so you shouldn't have to do a save after install.
Download the rpm and just -Uvh it over the old system. You should be fine as long as your release is above 145. - YUM soon - we promise.
You still need to do a console-save after a brand new install and we would recommend you do it after this install to ensure the port-opening stuff “fixes” correctly.
Kind Regards
Selintra
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what information do you need to add another carrier?
i have a canadian carrier that would like inclusion if you wish.
and would you explain the syntax needed to add Route Dial Plans..
i can connect to my voip extension with x ten
i can ring my extension from a landline connecting to the did i have added
but me is perplexed how to call out from the xten.. getfastbusy notconfig right response... my hunch is route.. add (do you use asterisk route codes in the route dial plan box?
cheers!
brent 1 :pint: 2 many
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Hello Domainwizard,
You are quite correct, you need to define at least one route. Routes tell SAIL which carrier(s) to use to despatch the outbound call. Yes you use asterisk route codes in the dial plan but you can have multiple entries per line. There is a description on our docs pages - see section 11 at
http://selintra.com/docs/cgi-bin/view/Main/SysVers20
Or simply click on the selintra logo at the bottom of any of the SAIL administration panels.
Adding a new carrier is very easy and it's also covered in the docs, however, if you tell us the name and url of the carrier we will include an automatic definition in the next maintenance release so that everyone can use it.
Kind regards
Selintra
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selintra
Since the update my fop page looks wierd ...
I have 2 lots of extentions ... then trunks ... then queues ... conferences under queues and parks under conferences.
under one lot of extentions there are all my extentions ... second one has nothing ... my queues doesn't have anything.
My Tunks are strange to what I had ... I have 4 trunks to my SPA3000 and 5 trunks to my asrta1.
In my setup I only have 2 trunks setup though ... One to the SPA3000 and one to astra1(Voip)
Is this a change or something gone strange.
Everything works ok
Regards,
Tib
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Since the update my fop page looks wierd ...
Hello Tib,
Nothing wrong (at least not intentionally). We've updated FOP to handle 60 entries. There are 32 extension entries (16 each in column 1 & 2), 16 Trunk entries and 4 Queue, Park and Conference entries.
It generates 4 entries per VOIP line so you can see and manage multiple concurrent calls on the same VOIP trunk. If it's generating 5 then that's a bug and if it's generating 4 for the spa3K that's also a bug.
As it stands, we're not entirely happy with it and we intend to clean it up a bit in 176/177; extensions will be allocated across and down so both columns will get used (without you having to have 17 or more extensions) and we won't display the Queues heading if you have none defined. We'll also clean up the bugs you've spotted.
Kind Regards
Selintra
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Hi All,
I'm very new to this. This is my setup
1. SME Server (6.0.1) Server and the gateway
2. SME Server (7.0RC1) Server only mode
3. 2 windows xp workstation with X-Lite softphone.
4. Ports open from SME 6.X Like this to SME 7.0
TCP 5060 192.168.96.10 5060 Remove
TCP 10000-20000 192.168.96.10 10000-20000 Remove
Problem:
1. I can't hear anything when talk to other PC but i can see incoming call and display everything no sound only ring tone
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Hello Cosy
Not sure where to begin with this. We assume the PC's with the softphones fitted are on the same subnet as the asterisk box. In which case we aren't sure why you are altering iptables on the gateway. Also we don't know why you are issuing a remove (which is anyway incorrectly coded). Finally, asterisk uses UDP, not TCP, to transmit it's data.
So...
Here's what we suggest. To establish a baseline, free of any firewall uncertainties, turn off the firewall on the asterisk box (one of the major differences in 7.0 is that it runs the firewall rules even when it's operating in server-only mode).
config masq setprop status disabled
To be on the safe side, reboot your box and check that the tables are empty.
iptables -L
you should see the three tables, each set to a blanket "ACCEPT".
log in to the asterisk console in highly verbose mode
asterisk -rvvvvvvvvvvvvvvvvvvvv
Now, fetch up your softphones and see if they register with asterisk (you'll see it on the asterisk console if they do). If they register then you know that the phones are finding asterisk OK.
Now try your calls and see how you get on. A good start-point is to issue *56* at one of the phones. This should read-back (from IVR) the extension number of the phone. Again, you will be able to watch this at the asterisk console. If you can see asterisk responding and you are still getting sound problems and you then you need to check that the softphones are correctly configured to use the soundgear on your PC's.
Hope this helps
Selintra
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HI,
Thanks for your reply.
1. All the servers and Workstation in same subnet.
2. I'm not altering iptables. Just cut and paste the Port forward from server-manager
--------------------------------------------------------------------------
Below you will find a table summarizing the current port-forwarding rules installed on this server. Click on the "Remove" link to remove the corresponding rule.
Protocol Source Port(s) Destination Host IP Address Destination Port(s) Action
UDP 10000-20000 192.168.96.10 10000-20000 Remove
UDP 8000-8005 192.168.96.10 8000-8005 Remove
UDP 5060 192.168.96.10 5060 Remove
----------------------------------------------------------------------------------
I just change to UDP now and open port 8000-8005, But not sure this just did it.
3. Turn Off the firewall. " To be on the safe side, reboot your box and check that the tables are empty"
CentOS release 4.2 (Final) - SME Server 7.0rc1
[root@websvr ~]# iptables -L
Chain INPUT (policy DROP)
target prot opt source destination
state_chk all -- anywhere anywhere
local_chk all -- anywhere anywhere
PPPconn all -- anywhere anywhere
denylog all -- BASE-ADDRESS.MCAST.NET/4 anywhere
denylog all -- anywhere BASE-ADDRESS.MCAST.NET/4
InboundICMP icmp -- anywhere anywhere
denylog icmp -- anywhere anywhere
InboundTCP tcp -- anywhere anywhere tcp flags:SYN,RST,ACK/SYN
denylog tcp -- anywhere anywhere tcp flags:SYN,RST,ACK/SYN
InboundUDP udp -- anywhere anywhere
denylog udp -- anywhere anywhere
gre-in gre -- anywhere anywhere
denylog gre -- anywhere anywhere
denylog all -- anywhere anywhere
Chain FORWARD (policy DROP)
target prot opt source destination
state_chk all -- anywhere anywhere
local_chk all -- anywhere anywhere
ForwardedTCP tcp -- anywhere anywhere tcp flags:SYN,RST,ACK/SYN
ForwardedUDP udp -- anywhere anywhere
denylog all -- anywhere anywhere
Chain OUTPUT (policy ACCEPT)
target prot opt source destination
PPPconn all -- anywhere anywhere
denylog all -- BASE-ADDRESS.MCAST.NET/4 anywhere
denylog all -- anywhere BASE-ADDRESS.MCAST.NET/4
ACCEPT all -- anywhere anywhere
Chain ForwardedTCP (1 references)
target prot opt source destination
ForwardedTCP_2165 all -- anywhere anywhere
denylog tcp -- anywhere anywhere tcp flags:SYN,RST,ACK/SYN
Chain ForwardedTCP_2165 (1 references)
target prot opt source destination
Chain ForwardedUDP (1 references)
target prot opt source destination
ForwardedUDP_2165 all -- anywhere anywhere
denylog udp -- anywhere anywhere
Chain ForwardedUDP_2165 (1 references)
target prot opt source destination
Chain InboundICMP (1 references)
target prot opt source destination
InboundICMP_2165 all -- anywhere anywhere
denylog icmp -- anywhere anywhere
Chain InboundICMP_2165 (1 references)
target prot opt source destination
ACCEPT icmp -- anywhere anywhere icmp echo-request
ACCEPT icmp -- anywhere anywhere icmp echo-reply
ACCEPT icmp -- anywhere anywhere icmp destination-unreachable
ACCEPT icmp -- anywhere anywhere icmp source-quench
ACCEPT icmp -- anywhere anywhere icmp time-exceeded
ACCEPT icmp -- anywhere anywhere icmp parameter-problem
denylog all -- anywhere anywhere
Chain InboundTCP (1 references)
target prot opt source destination
InboundTCP_2165 all -- anywhere anywhere
denylog tcp -- anywhere anywhere tcp flags:SYN,RST,ACK/SYN
Chain InboundTCP_2165 (1 references)
target prot opt source destination
denylog all -- anywhere !websvr.ausnetit.local
REJECT tcp -- anywhere websvr.ausnetit.local tcp dpt:auth reject-with tcp-reset
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:pop3
ACCEPT tcp -- anywhere websvr.ausnetit.local tcp dpt:auth
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:auth
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:imap
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:1723
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:ftp
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:ssh
ACCEPT tcp -- anywhere websvr.ausnetit.local tcp dpt:smtp
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:smtp
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:squid
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:ldap
ACCEPT tcp -- anywhere websvr.ausnetit.local tcp dpt:https
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:https
ACCEPT tcp -- anywhere websvr.ausnetit.local tcp dpt:smtps
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:smtps
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:domain
ACCEPT tcp -- anywhere websvr.ausnetit.local tcp dpt:http
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:http
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:http-admin
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:imaps
denylog tcp -- anywhere websvr.ausnetit.local tcp dpt:pop3s
Chain InboundUDP (1 references)
target prot opt source destination
InboundUDP_2165 all -- anywhere anywhere
denylog udp -- anywhere anywhere
Chain InboundUDP_2165 (1 references)
target prot opt source destination
denylog all -- anywhere !websvr.ausnetit.local
denylog udp -- anywhere websvr.ausnetit.local udp dpt:domain
Chain PPPconn (2 references)
target prot opt source destination
PPPconn_1 all -- anywhere anywhere
Chain PPPconn_1 (1 references)
target prot opt source destination
Chain denylog (37 references)
target prot opt source destination
DROP udp -- anywhere anywhere udp dpt:router
DROP udp -- anywhere anywhere udp dpts:netbios-ns:netbios-ssn
DROP tcp -- anywhere anywhere tcp dpts:netbios-ns:netbios-ssn
ULOG all -- anywhere anywhere ULOG copy_range 0 nlgroup 1 prefix denylog:' queue_threshold 1
DROP all -- anywhere anywhere
Chain gre-in (1 references)
target prot opt source destination
denylog all -- anywhere !websvr.ausnetit.local
denylog all -- anywhere anywhere
Chain local_chk (2 references)
target prot opt source destination
local_chk_2165 all -- anywhere anywhere
Chain local_chk_2165 (1 references)
target prot opt source destination
ACCEPT all -- anywhere anywhere
ACCEPT all -- 192.168.96.0/24 anywhere
Chain state_chk (2 references)
target prot opt source destination
ACCEPT all -- anywhere anywhere state RELATED,ESTABLISHED
[root@websvr ~]#
Is that Disable??
4. log in to the asterisk console in highly verbose mode, Now, fetch up your softphones and see if they register with asterisk
asterisk -rvvvvvvvvvvvvvvvvvvvv
asterisk -rvvvvvvvvvvvvvvvvvvvv == Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.3, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.3 currently running on websvr (pid = 3204)
Verbosity was 0 and is now 20
Apr 17 23:01:57 NOTICE[3324]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '88887389 @sip03.astrasip.com.au' timed out, trying again (Attempt #10850)
Apr 17 23:01:57 NOTICE[3324]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '88887389 @sip03.astrasip.com.au' timed out, trying again (Attempt #10856)
Apr 17 23:01:57 NOTICE[3324]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '88887389 @sip03.astrasip.com.au' timed out, trying again (Attempt #10677)
Apr 17 23:02:07 NOTICE[3324]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '88887389 @sip03.astrasip.com.au' timed out, trying again (Attempt #7413)
-- Registered SIP '5000' at 192.168.96.101 port 5060 expires 180
-- Saved useragent "X-Lite release 1105x" for peer 5000
Why this is not registering????
5. Now try your calls and see how you get on. A good start-point is to issue *56* at one of the phones.
I can hear my extention from PBX
-- Executing AGI("SIP/5000-fb6c", "selintra|*56*") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Playback) Options: (vm-extension)
-- Playing 'vm-extension' (language 'en')
-- Playing 'digits/5' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
Apr 17 23:03:17 NOTICE[3324]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '88887389 @sip03.astrasip.com.au' timed out, trying again (Attempt #10854)
Apr 17 23:03:17 NOTICE[3324]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '88887389 @sip03.astrasip.com.au' timed out, trying again (Attempt #10860)
Apr 17 23:03:17 NOTICE[3324]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '88887389 @sip03.astrasip.com.au' timed out, trying again (Attempt #10681)
-- Playing 'digits/0' (language 'en')
-- AGI Script selintra completed, returning 0
Apr 17 23:03:27 NOTICE[3324]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '88887389 @sip03.astrasip.com.au' timed out, trying again (Attempt #7417)
-- Timeout on SIP/5000-fb6c
== CDR updated on SIP/5000-fb6c
-- Executing Hangup("SIP/5000-fb6c", "") in new stack
== Spawn extension (internal, t, 1) exited non-zero on 'SIP/5000-fb6c'
-- Executing Hangup("SIP/5000-fb6c", "") in new stack
== Spawn extension (internal, h, 1) exited non-zero on 'SIP/5000-fb6c'
-- Executing AGI("SIP/5000-66e5", "selintra|*56*") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Playback) Options: (vm-extension)
-- Playing 'vm-extension' (language 'en')
-- Playing 'digits/5' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- AGI Script selintra completed, returning 0
Pease help.
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Build 173 is now up on our FTP server.
Just finished installing on a test system and everything appears to be going smoothly - just a couple of minor issues.
1. I cannot get the automation to work automatically. I have setup a closed period from 1700hrs to 0900hrs with an * under all the other days/months etc so the system should redirect to the CLOSED handler at 1700hrs but it's not working. If I key in *31* from a phone it works. I've tried keying *30* to put it into automatic mode but it still doesn't work. Have to manually put the system into a closed or open state daily. Not too sure if I've missed something here or not.
2. I got all these error messages when trying to open the stats program. Upon checking - the asterisk mysql db wasn't even present. I had to manually copy another db from another system and flush the data, set permissions etc. No idea what happened there.
3. Still a bit confused about how to work the "group ring". I can setup an extension (say 5300) that will ring all 4 phones. Tested and works fine but when I go into the Inbound route settings for the zap channel, 5300 is not available from the list of extensions to divert to. Would I have to create a custom app to direct all incoming calls to "group ring"? At the moment I'm using CFWBS on the first 2 phones so it "hunts" till it finds someone (combined with group pickup) but prefer to use group ring if possible.
On another note - Does anybody know how to program one of the seven speed dial buttons on the GXP-2000 to act as a "group pickup" (*8#). I went into the browser settings for the phone and programmed one of the buttons with *8# but it doesn't work. Just thinking as I'm typing, maybe I could setup an asterisk speed dial number for the group pickup then program the GXP2K speed dial button with this number. I'll try it when I'm down there next.
Other than those minor issues everything seems to be working rock solid.
Regards Lloyd
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HI,
Still my SAIL not registering properley i think SME 6.0.01 not port forward 10000-20000 like this???
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Hi Cosy
Sorry to hear your still having problems with firewalls and the like. You're 7.0 firewall is most definitely running. If it were disabled, this is what you would see
[root@old1 ~]# iptables -L
Chain INPUT (policy DROP)
target prot opt source destination
state_chk all -- anywhere anywhere
local_chk all -- anywhere anywhere
PPPconn all -- anywhere anywhere
Nevertheless, your 5000 extension is registering correctly...
-- Registered SIP '5000' at 192.168.96.101 port 5060 expires 180
-- Saved useragent "X-Lite release 1105x" for peer 5000
It's also sending the *56* correctly to asterisk and asterisk is responding as it should.
-- Executing AGI("SIP/5000-fb6c", "selintra|*56*") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Playback) Options: (vm-extension)
-- Playing 'vm-extension' (language 'en')
-- Playing 'digits/5' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- AGI Script selintra completed, returning 0
-- Timeout on SIP/5000-fb6c
== CDR updated on SIP/5000-fb6c
-- Executing Hangup("SIP/5000-fb6c", "") in new stack
== Spawn extension (internal, t, 1) exited non-zero on 'SIP/5000-fb6c'
-- Executing Hangup("SIP/5000-fb6c", "") in new stack
== Spawn extension (internal, h, 1) exited non-zero on 'SIP/5000-fb6c'
In among all this you have a SIP trunk defined to Astratel which is failing to register. This might be because you don't have your external IP Address filled out in Globals. When communicating with SIP from a server-only asterisk, asterisk needs to send the external IP Address to the SIP provider in the SIP packets. The address you need is the external IP Address that your 6.0 server-gateway is runing at. If you are unsure what this is do
ifconfig eth1
on your 6.0 box....
You can find the manual page for globals here
http://selintra.com/docs/cgi-bin/view/Main/DocChapter06
Kind Regards
Selintra[/quote]
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Hi Lloyd,
Looks like you're doing good with this.
Er...
1. The automation thing looks like a bug mate. We'll have a look at it for you.
2. We've no more idea than you how this might have happened. We haven't seen it here in over three months' continuous running on our main switch.
3. Yup, it's not showing speed dials on the inbound route. You got us again. It's a bug. Looks like 2:nil to you.
We'll release 176 or 177 later this week - hopefully we'll have a fix in for you. However, whichever release it's fixed in, you definitely shouldn't have to build a custom app.
As to programming the GXP, try just sending *8. You shouldn't need the #.
Best
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I cannot get the automation to work automatically.
Hi Lloyd,
We can't recreate this. It works perfectly here. Can you e-mail us the relevant entries from the selintra db?
Still a bit confused about how to work the "group ring"
We've fixed this - it was a bug. We've put 178 up onto the ftp server for you. Speed dials will now appear in the "open" and "closed" drop down menus in Trunks.
Kind Regards
Selintra
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Hi,
Just a few things:
For SPA ATA's (And maybe other adaptors and IP phones), the dialplan you recommend is (*x.|*xx*|x.), but that doesn't work if you want to record a greeting (*60*1234). So it will have to be (*x.|*xx*xxxx|x.).
Since i'm in denmark, which country identifyer will be the best to use in "Global Settings"? What does this funtion do anyway?
If i do a "One Thouch Record" where can i access that afterwards?
Now that we have this beautyfull thing, how do we backup the settings?
If these Q are already documented, then i missed them and apologize.
This is really working well for me now, and i have just ordered a SPA3000 to get my landline integrated in the system, can hardly wait....... :lol:
Per
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Hello Per
For SPA ATA's (And maybe other adaptors and IP phones), the dialplan you recommend is (*x.|*xx*|x.), but that doesn't work if you want to record a greeting (*60*1234). So it will have to be (*x.|*xx*xxxx|x.).
Very good point. Did you also try (*x.|*xx*x.|x.)?
We will update the guide accordingly :-).
Since i'm in denmark, which country identifyer will be the best to use in "Global Settings"? What does this funtion do anyway?
The country identifier finds its way into an asterisk file called indications.conf. It's main job is to mimic the correct cadences for each country so that an inbound dialer hears familiar tones. We believe that the closest to Denmark for tones is Sweden. However, if you have a look at the indications.conf file (it's in /etc/asterisk), you will quickly see how it works. Based upon what's already there and assuming the Danish PTT publishes the information, you can very easily create a bespoke entry for Denmark. If you decide to give it a go then do let us know and we'll create a "dk" entry for you in Globals.
If i do a "One Thouch Record" where can i access that afterwards?
It will turn up in your Primary i-bay in a folder in files. Asterisk records the conversation as two distinct recordings (one for each direction), which is not much good, so we dynamically invoke a sound manager called SOX to merge the two into one synchronised recording at the end of the call. We then create a dynamic link to it from files.
Now that we have this beautyfull thing, how do we backup the settings?
Because SAIL is a true SME server integration, it keeps all of its data in an e-smith database in /home/e-smith/db along with all the other SME databases (like configuration and account). As a result, it should get backed up automatically when you do a regular SME-server backup. Check that a file called /home/e-smith/db/selintra is being backed up and restored.
Kind Regards
Selintra
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Hi Selintra
Can you e-mail us the relevant entries from the selintra db
[root@server ~]# db selintra show dateSeg234621
dateSeg234621=dateSeg
datemonth=*
dayofweek=*
desc=After Hours
month=*
timespan=17:00-09:00
zzDummy=EOR
[root@server ~]# db selintra show dateSeg49751
dateSeg49751=dateSeg
datemonth=25
dayofweek=*
desc=Xmas Day
month=dec
timespan=*
zzDummy=EOR
[root@server ~]# db selintra show dateSeg67455
dateSeg67455=dateSeg
datemonth=1
dayofweek=*
desc=New Years Day
month=jan
timespan=*
zzDummy=EOR
[root@server ~]# db selintra show dateSeg697330
dateSeg697330=dateSeg
datemonth=*
dayofweek=sat
desc=Saturday
month=*
timespan=*
zzDummy=EOR
[root@server ~]# db selintra show dateSeg769071
dateSeg769071=dateSeg
datemonth=*
dayofweek=sun
desc=Sunday
month=*
timespan=*
zzDummy=EOR
Let me know if there are any other entries you need.
Regards Lloyd
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Thanks Lloyd,
Your dateSeg entries look fine, we'll take this off-line as a suspected bug. We've sent you detailed debugging instructions via e-mail.
Kind Regards
Selintra
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Hi Selintra
Very good point. Did you also try (*x.|*xx*x.|x.)?
Yes, that works.
The country identifier finds its way into an asterisk file called indications.conf. It's main job is to mimic the correct cadences for each country so that an inbound dialer hears familiar tones. We believe that the closest to Denmark for tones is Sweden. However, if you have a look at the indications.conf file (it's in /etc/asterisk), you will quickly see how it works. Based upon what's already there and assuming the Danish PTT publishes the information, you can very easily create a bespoke entry for Denmark. If you decide to give it a go then do let us know and we'll create a "dk" entry for you in Globals.
I found something already but I have to test it first. I will see what I can do, but I’m away from home for a while (work,work….)
It will turn up in your Primary i-bay in a folder in files. Asterisk records the conversation as two distinct recordings (one for each direction), which is not much good, so we dynamically invoke a sound manager called SOX to merge the two into one synchronised recording at the end of the call. We then create a dynamic link to it from files.
OK, I found I t there, but it was not merged in to one file!
I think I will make a cron job to move them to an ibay for easy access. I had to go in putty and move them manually, the link does not work with ftp.
Per
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Hi Selintra
It could be nice an easy if there where links in the server manager panel for the fop and stat pages. (And others if i there are more that i don't know of)
Per
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Hi Per,
OK, I found I t there, but it was not merged in to one file
We have performed one touch recording on different installations, and we are unable to reproduce your scenario (2 separate sound files). We always get a single merged file.
If this is still an issue for you please can you send us the asterisk log and the 2 sound files.
Kind Regards,
Selintra
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Hi Per,
OK, I found I t there, but it was not merged in to one file
We have performed one touch recording on different installations, and we are unable to reproduce your scenario (2 separate sound files). We always get a single merged file.
If this is still an issue for you please can you send us the asterisk log and the 2 sound files.
Kind Regards,
Selintra
Hi Selintra
I will do more testing on this as soon as i am home again. I am away from home for work some days.
Can it be because it's a call from extension to extensoin and not an outside line?
Is it possible to have the recording permanent or even for certain extensions or certain incoming numbers?
Per
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Hi Selintra,
The country identifier finds its way into an asterisk file called indications.conf. It's main job is to mimic the correct cadences for each country so that an inbound dialer hears familiar tones. We believe that the closest to Denmark for tones is Sweden. However, if you have a look at the indications.conf file (it's in /etc/asterisk), you will quickly see how it works. Based upon what's already there and assuming the Danish PTT publishes the information, you can very easily create a bespoke entry for Denmark. If you decide to give it a go then do let us know and we'll create a "dk" entry for you in Globals.
The Swedish tones sounds similar to the danish tones, but i have found this in another forum and it looks a little different:
description = Denmark
ringcadance = 1000,5000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/5000
congestion = 425/250,0/750
callwaiting = 425/200,0/500,425/200,0/8000
info = 950/300,0/20,1400/300,0/20,1800/300,0/1000
; dialrecall and record not necessarily these
dialrecall = 425/325,0/25
record = 1400/500,0/15000
I used it until i upgraded to Sail 183, then it was overwritten. It worked fine.
Per
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Hello Per,
Here's a little patch for you to try (this will work for anyone who wants to add a currently unlisted country to the list).
The code fragment which generates the country data for indications.conf is in the following file..
/etc/e-smith/templates/etc/asterisk/indications.conf/30-countries
Add your description for Denmark. Don't forget to give it a header..
[dk]
description = Denmark
ringcadance = 1000,5000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/5000
congestion = 425/250,0/750
callwaiting = 425/200,0/500,425/200,0/8000
info = 950/300,0/20,1400/300,0/20,1800/300,0/1000
; dialrecall and record not necessarily these
dialrecall = 425/325,0/25
record = 1400/500,0/15000
Then, open the "Globals" panel manager...
/etc/e-smith/web/functions/sarkglobals
At line 225 you will find the popup which generates the country list. Simply add 'dk' to the list.
225 $q->popup_menu (
226 -name => 'COUNTRY',
227 -values => ['au','br','cl','de','ee','fi','fr','gr','hu',
228 'it','lt','mx','nl','no','nz','pl','pt','sg',
229 'se','tw','ru','uk','us','za' ],
230 -default => $country))),
You should now be able to test your settings by going to the globals panel, choosing dk for your country and pressing save.
When you are happy with the settings, let us know and we'll commit them to svn.
Kind Regards
Selintra