Newbie Question - cannot make internal calls (including voicemail)
Submitted by rstille on Mon, 02/08/2010 - 8:16pm.
*** Updating this post *** the problem occurs when any internal extension is called, not just voicemail. I have captured a CLI output and will post it separately ...
All, OK, this is I'm sure a rookie question, but after 2 hours trying to figure it out, I'm finally giving in ..
I have a new Trixbox CE install on v 2.8.0.3. I have polycom IP 500 phones, which I have been able to provision successfully using the IP501 setting in Endpoint Manager.
I have left all the feature setting at default - *97 to reach my voicemail, for instance ... when the extensions were created, I was sure to set the voicemail feature to 'Enabled' and they use the extension number (101 in this case) for the default password.
The problem is voicemail ... when dialing *97 (or *98 for that matter or ANY internal extension for that matter) - the voicemail or extension system does not pickup. Rather, there is a message to the essence of "the number you dialed is not in service."
Did I miss a configuration step somewhere, to configure and enable voicemail?
Sorry if this is overly simple ...
Randy
CLI File
Tue, 02/09/2010 - 3:14pm
Thanks for the tips Scott, and your patience! Here is the CLI file I captured. I believe I am seeing that the system thinks I am dialing extension '10', rather than extension '100' ... but I don't know why, or how this is happening.
Asterisk 1.6.0.10-FONCORE-r40, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core 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 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': == Found
== Parsing '/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.6.0.10-FONCORE-r40 currently running on trixbox1 (pid = 2922)
trixbox1*CLI>
Verbosity is at least 9
[Ktrixbox1*CLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
[Ktrixbox1*CLI>
-- Executing [10@from-internal:1] [1;36;40mResetCDR[0;37;40m("[1;35;40mSIP/101-b7508078[0;37;40m", "[1;35;40m[0;37;40m") in new stack
-- Executing [10@from-internal:2] [1;36;40mNoCDR[0;37;40m("[1;35;40mSIP/101-b7508078[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Ktrixbox1*CLI>
-- Executing [10@from-internal:3] [1;36;40mWait[0;37;40m("[1;35;40mSIP/101-b7508078[0;37;40m", "[1;35;40m1[0;37;40m") in new stack
[Ktrixbox1*CLI>
-- Executing [10@from-internal:4] [1;36;40mPlayback[0;37;40m("[1;35;40mSIP/101-b7508078[0;37;40m", "[1;35;40msilence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer[0;37;40m") in new stack
--
[Ktrixbox1*CLI>
--
[Ktrixbox1*CLI>
== Spawn extension (from-internal, 10, 4) exited non-zero on 'SIP/101-b7508078'
-- Executing [h@from-internal:1] [1;36;40mMacro[0;37;40m("[1;35;40mSIP/101-b7508078[0;37;40m", "[1;35;40mhangupcall[0;37;40m") in new stack
-- Executing [s@macro-hangupcall:1] [1;36;40mResetCDR[0;37;40m("[1;35;40mSIP/101-b7508078[0;37;40m", "[1;35;40mvw[0;37;40m") in new stack
-- Executing [s@macro-hangupcall:2] [1;36;40mNoCDR[0;37;40m("[1;35;40mSIP/101-b7508078[0;37;40m", "[1;35;40m[0;37;40m") in new stack
-- Executing [s@macro-hangupcall:3] [1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/101-b7508078[0;37;40m", "[1;35;40m1?skiprg[0;37;40m") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] [1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/101-b7508078[0;37;40m", "[1;35;40m1?skipblkvm[0;37;40m") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] [1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/101-b7508078[0;37;40m", "[1;35;40m1?theend[0;37;40m") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] [1;36;40mHangup[0;37;40m("[1;35;40mSIP/101-b7508078[0;37;40m", "[1;35;40m[0;37;40m") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/101-b7508078' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-b7508078'
[Ktrixbox1*CLI> quit
Executing last minute cleanups
changed extensions to 20x - partiallly solved
Tue, 02/09/2010 - 5:14pm
So, I can't figure out what I think is the dial plan problem, but I did change the extensions to begin with 200 and, as you would expect, internal calls work. Voicemail does pickup, and you can leave a message. When you do, the Vmail indicator light blinks on the phone and a tone sounds from the handset that a vmail has been received.
BUT ... dialing *97 doesn't work for vmail pickup. When you dial *97, the display shows "enter more digits". If you press "send call", you get a fast busy..... the CLI output shows that no SIP connection is attempted by the phone to the trixbox server.
Thoughts?
SOLVED ...
Tue, 02/09/2010 - 5:46pm
OK, I finally fixed this ... the problem is in the phones themselves. When reset to the factory config, there is a "digitmap" field in the phone config. I logged into the phone via web browser, and removed the settings in this field. After that was completed, calls to voicemail went through no problem.
Also, I subsequently went back to test extension 100 --- worked like a champ.
So the problem was in the phone config ... Polycom IP500, for those reading this months from now ...
A blank dialplan will cause
Tue, 02/09/2010 - 10:59pm
new dial plan ..
Wed, 02/10/2010 - 9:12pm
Skyking, you are correct. I played a bit with the digit map definitions from Polycom. Note that I am not intending for a '9' to be required for outbound calling.
Here is what I came up with. I'm sure it can be improved upon ... I am using 2xx three digit extensions. Our local area codes are 303 and 720 here in Denver.
[3-9]11|211|311|0T|1[2-9]xxxxxxxxx|[3-9]xxxxxxxxx|[3-9]xxxT|2[0-9]X|*9X|*XX
Please note that the firmware doesn't seem to be very intuitive on these digit maps. For instance, if you configure
[1-2]XX|1NXXNXXXXXX
into the digit map, it will not be able to resolve the rule and will send it to an "impossible match" ... dialing "100" sends the number '10' to the SIP server after the second digit is typed.
This probably makes sense (to someone).
Member Since:
2010-02-08