No voice from the voicemail system and no speaking clock, in fact no voice from the PBX at all

hello,

since a few weeks i cant get the PBX to talk to me at all. neither the voicemail system nor any other “speaking” feature works. i am wondering is it the wrong file format ? i have tried the extra sound packages and have no success with any of those. maybe i am on the wrong path altogether. which sound files have to go to which location and which format do they have to be. in my sip.conf i have the following lines:

disallow=all
allow=ulaw
allow=alaw

does that mean my sound files have to be ulaw or alaw ? does the music on hold configuration have any effect on the voice output of the PBX ? i have moh working with mp3.

[default]
mode=files
directory=/var/lib/asterisk/mohmp3/
random=yes

i have asterisk 1.2.23 and freepbx 2.2.3

regards
Jan

PS: here the output of the logfile if i call the speaking clock, i type in *60 and nothing happens, i hang up than. all phone calls btw are working fine.

Sep 3 12:13:07 DEBUG[3179] chan_sip.c: Setting NAT on RTP to 524288
Sep 3 12:13:08 DEBUG[3179] chan_sip.c: Stopping retransmission on ‘[email protected]’ of Response 101: Match Found
Sep 3 12:13:08 DEBUG[3179] chan_sip.c: Setting NAT on RTP to 524288
Sep 3 12:13:08 DEBUG[3179] chan_sip.c: Checking SIP call limits for device 121
Sep 3 12:13:08 DEBUG[3179] chan_sip.c: build_route: Contact hop: Jan Home sip:[email protected]:5060
Sep 3 12:13:08 DEBUG[3171] channel.c: Avoiding initial deadlock for ‘SIP/121-081b15b8’
Sep 3 12:13:08 VERBOSE[3436] logger.c: – Executing Answer(“SIP/121-081b15b8”, “”) in new stack
Sep 3 12:13:08 VERBOSE[3436] logger.c: – Executing Wait(“SIP/121-081b15b8”, “1”) in new stack
Sep 3 12:13:08 DEBUG[3179] chan_sip.c: Stopping retransmission on ‘[email protected]’ of Response 102: Match Found
Sep 3 12:13:09 VERBOSE[3436] logger.c: – Executing Set(“SIP/121-081b15b8”, “NumLoops=0”) in new stack
Sep 3 12:13:09 DEBUG[3436] pbx.c: Expression result is ‘1188818000’
Sep 3 12:13:09 VERBOSE[3436] logger.c: – Executing Set(“SIP/121-081b15b8”, “FutureTime=1188818000”) in new stack
Sep 3 12:13:09 VERBOSE[3436] logger.c: – Executing Playback(“SIP/121-081b15b8”, “at-tone-time-exactly”) in new stack
Sep 3 12:13:09 DEBUG[3436] channel.c: Scheduling timer at 160 sample intervals
Sep 3 12:13:09 VERBOSE[3436] logger.c: – Playing ‘at-tone-time-exactly’ (language ‘en’)
Sep 3 12:13:20 DEBUG[3436] channel.c: Scheduling timer at 0 sample intervals
Sep 3 12:13:20 VERBOSE[3436] logger.c: == Spawn extension (from-internal, *60, 5) exited non-zero on ‘SIP/121-081b15b8’
Sep 3 12:13:20 VERBOSE[3436] logger.c: – Executing Macro(“SIP/121-081b15b8”, “hangupcall”) in new stack
Sep 3 12:13:20 VERBOSE[3436] logger.c: – Executing ResetCDR(“SIP/121-081b15b8”, “w”) in new stack
Sep 3 12:13:20 DEBUG[3436] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Sep 3 12:13:20 DEBUG[3436] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES (‘2007-09-03 12:13:08’,’“device” <121>’,‘121’,’*60’,‘from-internal’, ‘SIP/121-081b15b8’,’’,‘Playback’,‘at-tone-time-exactly’,12,12,‘ANSWERED’,3,’’,‘1188817988.6’)
Sep 3 12:13:20 DEBUG[3436] app_macro.c: Executed application: ResetCDR
Sep 3 12:13:20 VERBOSE[3436] logger.c: – Executing NoCDR(“SIP/121-081b15b8”, “”) in new stack
Sep 3 12:13:20 NOTICE[3436] cdr.c: CDR on channel ‘SIP/121-081b15b8’ not posted
Sep 3 12:13:20 NOTICE[3436] cdr.c: CDR on channel ‘SIP/121-081b15b8’ lacks end
Sep 3 12:13:20 DEBUG[3436] app_macro.c: Executed application: NoCDR
Sep 3 12:13:20 DEBUG[3436] pbx.c: Expression result is '1’
Sep 3 12:13:20 VERBOSE[3436] logger.c: – Executing GotoIf(“SIP/121-081b15b8”, “1?skiprg”) in new stack
Sep 3 12:13:20 VERBOSE[3436] logger.c: – Goto (macro-hangupcall,s,6)
Sep 3 12:13:20 DEBUG[3436] app_macro.c: Executed application: GotoIf
Sep 3 12:13:20 DEBUG[3436] pbx.c: Expression result is '1’
Sep 3 12:13:20 VERBOSE[3436] logger.c: – Executing GotoIf(“SIP/121-081b15b8”, “1?theend”) in new stack
Sep 3 12:13:20 VERBOSE[3436] logger.c: – Goto (macro-hangupcall,s,9)
Sep 3 12:13:20 DEBUG[3436] app_macro.c: Executed application: GotoIf
Sep 3 12:13:20 VERBOSE[3436] logger.c: – Executing Wait(“SIP/121-081b15b8”, “5”) in new stack
Sep 3 12:13:20 VERBOSE[3436] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/121-081b15b8’ in macro 'hangupcall’
Sep 3 12:13:20 VERBOSE[3436] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/121-081b15b8’
Sep 3 12:13:20 DEBUG[3436] chan_sip.c: update_call_counter(121) - decrement call limit counte

This happened to me many times when I started using Asterisk and related stuff, and problem was always in the firewall settings which was blocking one way voice traffic, or both ways sometimes. This included the scenarios when phones worked fine but sound from the PBX didn’t work.

Your log shows that it is playing the sound files. But obviously you are not getting the voice packets to your phone thats why you can’t hear it.

Try this:

If you are using a standard SOHO router like linksys, d-link etc. connected to a local LAN, then make sure to open ports 5060 and 10000-2000. To make it simpler, just put the IP address of your server in the router DMZ settings. This will open all the ports for your server to communicate. This is considered very unsafe way to do things so do this only for testing. Later you should learn how to do port forwarding if you don’t know this already.

Now make sure in your sip.conf, you have the following lines in the [general] section.

insecure = invite
externip = {your external IP address. You can find this at whatismyip.com}
localnet = {your local network settings, something like =192.168.1.0/255.255.255.0}
nat = yes

With these settings correctly done, you should be able to hear your PBX.

~~~~*
“We do not see things as they are. We see things as we are.”

You want to do those in sip_How To Setup a Remote SIP Extensionnat.conf. Fore more details see:

Philippe Lindheimer - FreePBX Project Lead
http//freepbx.org - IRC FreePBX

hello guys,

firewall is not an issue. i have this effect in the office where the ATA is in the same ip network than the actual asterisk box without any firewalling. i also have a trixbox for testing on which all our phones do get the voices from the voicemails and co. i have no problems with phone calls at all, on none of the machines, i can hear the other side no matter if i go via the sip/iax provider or via the ISDN line. i also get moh no problem. i was hoping to find some obvious difference between the working trix box and the asterisk & freepbx box, but i don’t find anything which would catch my eye. i was thinking od a codec problem but i don’t really understand which modules are involved. if i list the module stats from the asterisk cli i can see a number of modules active. any suggestions in the direction of codecs ?

regards
Jan

So this box is a homebrew???

Did you install the asterisk addons and asterisk sounds from source or what???

Did you install the info module in freepbx??

Bubba

its a debian etch system, i would not call that a homebrew. if you mean the module " Asterisk Info" yes, thats installed. asterisk is installed from source code including the addons which where needed (i think some mysql stuff). don’t get me wrong, the phonesystem works fine, inbound and outbound call are all perfectly fine. only prompts or voice output like voicemailbox or speaking clock do not work.

Jan

It is info services in freepbx which need to check for.
and there is asterisk sounds which needs to installed as well.
asterisk
zaptel
libpri
addon’s
sounds.

Bubba

yes, that module is installed, and i have no missing files. the actual files which contain the voice for the password question or the digits for the speaking clock are there and are played. i have no missing files. i double checked by naming the sounds folder different, i then get an error message complaining about the file cannot be opened. so i have to assume the files are actually played.

any other suggestions ?

Jan

ok
how about *43 echo test???

Bubba

I had removed the asterisk-sounds.noarch module from trixbox…

by doing the echo test i get the following log entry:

—snip—
Sep 5 12:04:03 VERBOSE[15353] logger.c: – Executing Playback(“SIP/120-081b10c8”, “demo-echotest”) in new stack
Sep 5 12:04:03 DEBUG[15353] channel.c: Scheduling timer at 160 sample intervals
Sep 5 12:04:03 VERBOSE[15353] logger.c: – Playing ‘demo-echotest’ (language ‘en’)
—snip—

file is there and is played but i dont hear it.

regarding the module asterisk-sounds.noarch, i dont have that at all. by saying module, do you mean an asterisk module or a trixbox specific piece of software ? i am not running a trixbox.

Jan

i have the same problem. my system is up for over a year now, and i just set up a remote extension. calls are fine (inbound and outbound) but if i call voice mail i dont recive audio. the logs show that the prompts are being played. strange.

snoozer: Did you do a upgrade of asterisk or any other part of your system when the prompts stoped working?

I had a similar issue where MOH would never play. Full audio during conversations, but no MOH.

I checked codecs, file formats, ports, paths, I even downloaded the mpg files to my windows box to make sure they
were playable.

The problem turned out to be the ‘LICENSE’ file in the MOH directory. Asterisk picked up the MOH files and when it opened
the LICENSE file, it was not valid MP3, and I got no audio.

Deleting the LICENSE file fixed it… Hasn’t been a problem under Asterisk 1.4 though…

…i am actually not sure anymore i ever had any prompts of THAT box working. i have spend a lot of time to get MOH working which does work without any problem. i believe i never checked the prompts from the PBX itself. the files for the prompts are not played randomly like the MOH files. any other files on that directory should not be a problem. i am just lost here. i would need to understand which way asterisk is doing his job to go any further. is there something like a flow-chart to see what should happen ? i would need to compare what should be with my present situation.

Jan

for all who are interested, i have solved the problem.

look here:
http://forums.digium.com/viewtopic.php?t=17981

regards
Jan