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Visual Voicemail

grantk
Posts: 29
 Member Since:
2010-07-23
Submitted by grantk on Thu, 08/12/2010 - 11:28am.

Back with another question today, I have a 53i and 57iCT I am trying to get the visual voicemail to work on.
I am running trixbox v2.8.04, asterisk 1.6.2.10.

Normal, *97 voicemail checking works fine on these extensions. If I try visual voicemail I can connect, see the message, but not play it.

I can not record any greetings through visual voicemail either. I get an error that says

"sorry you do not have a voicemail box. I am sorry I can not let you do that"( need to add a "dave" onto the end of that recording)

At that point you can not exit out using the done key or any other besides the softkey I have programed to the agent.php script.

The configs are made automatically using the hotdesk feature.

Here is the .prf entry for the 53i

prgkey6 type:xml
prgkey6 label:Voice Mail
prgkey6 value:http://$$AA_XML_SERVER_AA$$/$$AA_XMLDIRECTORY_AA$$/asterisk/vmail.php?ext=$$SIPUSERNAME$$&user=$$AA_SIPUSERNAME_AA$$
prgkey6 states:idle

.prf entry for the 57iCT

softkey4 type:xml
softkey4 label:Voice Mail
softkey4 value:http://$$AA_XML_SERVER_AA$$/$$AA_XMLDIRECTORY_AA$$/asterisk/vmail.php?ext=$$SIPUSERNAME$$&user=$$AA_SIPUSERNAME_AA$$
softkey4 states:idle

the 57iCT section also has a

# Feature override
voicemail script: http://$$AA_XML_SERVER_AA$$/$$AA_XMLDIRECTORY_AA$$/asterisk/vmail.php?ext=$$SIPUSERNAME$$&user=$$AA_SIPUSERNAME_AA$$

in there but does not behave any differently. I imagine that feature overide is not needed.

The extensions were all made with the bulk configuration tool which lead me to a bum context setting intially, I have changed that to default.

The freepbx voicemail section for these phones looks like this:

Status: enabled
VoiceMail Password: pass
Email Address user@example.com
Pager Email Address
Email Attachment: No
Play CID: No
Play Envelope: No
Delete Voicemail: No
VM Options:
VM Context: default

The voicemail module is at version 2.5.1.6 enabled and up to date

I also have no indication lamp on new voicemails.

I set trace and debug = 1 in the server.conf but have only gotten a debug file with very little information.
==> 08122010.debug 12:18:03 PM /var/www/html/aastra/asterisk/aastra_daemon2 Unable to connect to manager localhost:5038 (111): Connection refused
12:18:03 PM /var/www/html/aastra/asterisk/aastra_daemon1 Unable to connect to manager localhost:5038 (111): Connection refused

‹ BLF to monitor DND vs On Call Aastra Scripts 2.3.0 / CE 2.6.2 -- question - how to sync day/night status with * codes? ›


grantk
Posts: 29
 Member Since:
2010-07-23
bump. I got pulled away from
Thu, 11/18/2010 - 6:32pm

bump. I got pulled away from this project for a few months, got back and was searching for a solution to this problem. I forgot I actually posted this.

I changed up the config a bit and got rid of the user variable and still have this problem.

Anyone got an idea?



obeliks
Posts: 877
 Member Since:
2010-03-14
show us the content of
Thu, 11/18/2010 - 7:33pm

show us the content of manager.conf.
It seems the manager is not listening on port 5038 for some reason



grantk
Posts: 29
 Member Since:
2010-07-23
Here is what I
Fri, 11/19/2010 - 7:29am

Here is what I have.

***manager.conf*****
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0

[admin]
secret = amp111
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

#include manager_additional.conf
#include manager_custom.conf

******manager_custom.conf*********
[aastra-xml]
secret = aastra227872
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user,originate
write = system,call,log,verbose,command,agent,user,originate

The manager_additional.conf is empty



grantk
Posts: 29
 Member Since:
2010-07-23
I just loaded up the
Fri, 11/19/2010 - 7:48am

I just loaded up the asterisk console and here is what is shown there when I press the visual voicemail button, this one was using the configuration that passes the username and extension. I logged in, saw one voicemail displayed. I press play then get the recording. 'sorry you do not have a ...' and then the '..Cant let you do that' recording.

== Parsing '/etc/asterisk/asterisk.conf': == Found
== Parsing '/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.6.2.10 currently running on trixbox1 (pid = 3076)
Verbosity was 3 and is now 4
-- Remote UNIX connection
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Executing [vmail@from-internal:1] Answer("SIP/101-000000e3", "") in new stack
== Manager 'aastra-xml' logged on from 127.0.0.1
-- Executing [vmail@from-internal:2] AGI("SIP/101-000000e3", "aastra-vm.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/aastra-vm.php
== Manager 'aastra-xml' logged off from 127.0.0.1
-- AGI Script Executing Application: (MailboxExists) Options: (@default)
-- AGI Script Executing Application: (Playback) Options: (custom/vm-you-no-box)
-- Playing 'custom/vm-you-no-box.slin' (language 'en')
-- AGI Script Executing Application: (Playback) Options: (sorry-cant-let-you-do-that)
-- Playing 'sorry-cant-let-you-do-that.ulaw' (language 'en')
-- AGI Script aastra-vm.php completed, returning 0
-- Executing [vmail@from-internal:3] Hangup("SIP/101-000000e3", "") in new stack
== Spawn extension (from-internal, vmail, 3) exited non-zero on 'SIP/101-000000e3'
-- Executing [h@from-internal:1] Macro("SIP/101-000000e3", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-000000e3", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/101-000000e3", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/101-000000e3", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/101-000000e3", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/101-000000e3' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-000000e3'



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