Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: del on February 09, 2007, 12:30:09 AM
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Hi All,
On a recent visit to the UK I helped a friend install SAIL/Asterisk on his SME7 server. I called him today and it just kept ringing out, no voice mail :shock: I got him to try it ext to ext and this is what happens, the call rings out and then says that the call is established even though no one answers the other ext :shock: The setup is as follows: BT Voyager router connected to the internet, SME server connected to the router's DMZ in server/gateway on one NIC and a second NIC connected to a switch (LAN) 2 other PC's connected to the switch both running XP pro, one with sjphone and the other with xlite. Also connected is a Sipura SPA3000 to which a BT line is connected and a cordless phone. Calls can be made out and received. But no voice mail is ever received (due to the fact it doesn't work!). Is there any where to turn on voice mail even though it was never turned off :D Any pointers appreciated.
EDIT TO ADD:
The above server is using selintra-sail-2.1.14-344.noarch.rpm, I am using selintra-sail-2.1.14-339.noarch.rpm and I see that selintra-sail-2.1.14-406.noarch.rpm is available for download, which one should I be using?
Regards,
Del
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Hello Del,
First of all; voicemail DOES work - 6000 users can't all be wrong.
Having said that, there is clearly however, something wrong with your friend's set up. Did you watch the console log while this call was running? The fact that the call is connecting suggests that voicemail may be running perfectly well but either the system firewall is incorrectly set up and you are getting one-way sound OR voicemail is running but the sound files are missing/corrupt. Without logs it's impossible to tell. Get your friend to run the call with the console log enabled. If it still isn't obvious what's wrong then send us the logs.
If you figure it out from the logs then let us know what it was.
Selintra
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Hi Seleintra,
First of all; voicemail DOES work - 6000 users can't all be wrong.
I meant that it doesn't work on this particular system, not that it "doesn't" work, in fact mine works fine and always as. However on my friends setup it isn't kicking in at the moment. I will get him to run the console log. As for firewall settings they are as per SME server/gateway out of the box, no other firewall is involved. If it was due to one way sound would he be able to make calls? Because he can make and receive calls OK. The reason he has only just found out about the voice mail problem is because he had BT's call minder on the line and we thought this was coming in before Asterisks voice mail and it has taken him 2 months to get it taken off :shock: I will him to reinstall the sounds rpm first and see what happens. Thanks again for your help
Regards,
Del
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Hi Selintra,
The sounds rpm has been reinstalled and the voice mail is now up and running :D Only thing is that on ext 5000, which is the SPA3000 it takes 110 seconds before it comes in, the other two exts, which are both softphones come in on 8 seconds. Could this be a setting in the SPA3000? I have done some searches but have come up with nothing really useful. Thanks again.
Regards,
Del
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look at the ring delay in the database.
do
database show
at the asterisk console.
This will list all the database records - look at ringdelay for the extension the 3K is associated with.
set ringdelay with *26*nn; where nn is the number of seconds.
if it isn't that then it's something in the Spa3K.
Best
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Hi Selintra,
Today I finally got the time to have a look at this :D look at the ring delay in the database.
do
database show
at the asterisk console
I am having a mental blockage (some people would say that I am just mental :D ) and can't remember or work out how to get to the asterisk console :oops: If I run database show as root I get a message saying command not found :shock: So I am obviously not in the correct directory to run this command, can you help me figure out where I should be at :oops: I have got him to try *26*10 and it still doesn't go to VM for over a minute :shock: So I would just like to run the command as you suggested before I condemn the SPA3000.
Regards,
Del
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HI Del
at the root console ....
[root@switch ~]# asterisk -rvvvvv
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.10, 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 1.2.10 currently running on switch (pid = 22157)
Verbosity was 4 and is now 5
switch*CLI> database show
/SIP/Registry/5002 : 192.168.1.123:5060:160:5002: sip:5002@192.168.1.123
/SIP/Registry/5003 : 86.27.126.170:5060:180:5003: sip:5003@86.27.126.170:5060
/SIP/Registry/5004 : 192.168.1.115:5060:160:5004: sip:5004@192.168.1.115
/SIP/Registry/6002 : 192.168.1.127:2051:180:6002: sip:6002@192.168.1.127:2051;line=wl9iquye
/SIP/Registry/6009 : 192.168.1.196:2051:180:6009: sip:6009@192.168.1.196:2051;line=4lcwq6ad
/SIP/Registry/6010 : 81.103.95.24:1000:160:6010:s ip:6010@192.168.1.66:1000
/SIP/Registry/6014 : 81.103.95.24:5060:160:6014:s ip:6014@192.168.1.65:5060
/cfim/6003 : 6002
No idea why asterisk is ringing for so long unless you are using an alias or a ring group with a timeout on it. I'd need to see the console output to help you further.
incidentally - can you change the title of this post - it's not particularly flattering to have "sail voicemail not working" splattered all over the forum, particularly when it's working fine.
J
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Hi Selintra,
This is the output at the asterisk console:home-server*CLI> database show
/SIP/Registry/01905xxxxxx : 192.168.1.246:5060:180:01905 xxxxxx:sip:01905xxxxxx@192.168.1.246:5060
/SIP/Registry/5000 : 192.168.1.246:5060:180:5000: sip:5000@192.168.1.246:5060
/SIP/Registry/5001 : 192.168.1.241:46822:180:5001 :sip:5001@192.168.1.241:46822;rinstance=6f2347f79ae72d1e
/SIP/Registry/5003 : 192.168.1.241:5060:160:5003: sip:5003@192.168.1.241:5060
/SIP/Registry/6001 : 192.168.1.244:12586:180:6001 :sip:6001@192.168.1.244:12586;rinstance=e7aaeef9800caf16
/dundi/secret : ED7+GrQCsSHOOk1UkRAVWg;ffBtR hK+FyDxtcNumQiQVA
/dundi/secretexpiry : 1172430651
/ringdelay/5000 : 0
/ringdelay/5001 : 5
/ringdelay/5003 : 05
Funny thing is if I am correct in reading it it says that ext 5000 which is the SPA3000 is set to no delay :shock: Thanks again for your help and I didn't mean to make the title unflattering, only pointing out that the voice mail wasn't working on this server, not that it didn't work at all.
Regards,
Del
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Hi Del
Thanks for the title change.
Setting ringdelay to no delay (zero) means "ring forever" or until the absolute timeout (I can't rememeber what that is set to). It's usually used as a quick way to turn voicemail off. To change it; use *26* - however, you said you'd already done that... Just to avoid confusion it's *26*nn, all in one dial. To set it to 10 seconds do *26*10 all in one go.
If it doesn't work it will probably be the Spa3K dialplan which is stopping the dial going through. The suggested SAIL dial plan for the spa3K is (*x.|*xx*|x.). This will pass both leading and trailing asterisks to SAIL/Asterisk.
Best
Selintra
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Hi All,
I have now got different problems in my attempt to solve this :shock:
See here:
http://forums.contribs.org/index.php?topic=35971.msg158432#msg158432
Thanks to everyone for looking. :D
Del