3 problems - trixbox pro standard edition

Hello,

3 problems with trixbox pro se - maybe anyone can help.
first i'll try to explain our infrastructure. i-net-provider provides i-net + VOIP .behind - the yet not active firewall Linksys - Fileserver (DualCore, 2GB, 2 NICs) under w2k3. theres a virtual machine running at w2k3 ( vmware-server 2.0 ) with Virtually installed Trixbox Pro standard edtion (iso-file from trixbox / march 2009) - 1 NIC only for trixbox (one 192.168.1.10-fileserver, 192.168.1.20-trixbox-vm). Fixed IPs in LAN (both NICs) and 5 Grandstream GXP2000 Telefon-all with fix ip.
After Trixbox Pro Upgrade (must be due to trixbox) today -> new admin gui. all right.

Problems:

1) Callers from outside of (from province, but also from same city) to office are asking where we are located. far east, asia, are something. no...from same city. this confusion comes due to the ringtone which they can hear. We are in middle europe ... but people believe like the call to usa or something
much confusion ... how can i change that behavior? (the ringtone they can hear when our phone is ringing)

2) people told me that phone-connection is sometimes (i dont know if there is much traffic on i-net or fileserver is busy ... but there is REALLY not much traffice - wether i-net nor in lan/fileserver). here's a log of a "connection lost" ... i was alone in the office and there was no traffic on i-net. (snipplet ... asterisk-log (btw. trixbox per tail -f /var/log/asterisk/messages)
-snip-
Mar 20 19:40:03 VERBOSE[32289] logger.c: -- Called telenode/05-dialed-number
Mar 20 19:40:03 VERBOSE[32289] logger.c: -- SIP/telenode-0844d838 is ringing
Mar 20 19:40:03 VERBOSE[32289] logger.c: -- SIP/telenode-0844d838 is making progress passing it to SIP/000B8219B7as-08420e00
Mar 20 19:40:03 VERBOSE[32289] logger.c: -- SIP/telenode-0844d838 answered SIP/000B8219B7as-08420e00
Mar 20 19:40:03 VERBOSE[32289] logger.c: -- Attempting native bridge of SIP/000B8219B7as-08420e00 and SIP/telenode-0844d838
Mar 20 19:40:18 VERBOSE[32289] logger.c: == Spawn extension (internal, 00599994000, 5) exited non-zero on 'SIP/000B8219B7as-08420e00'
Mar 20 19:40:18 VERBOSE[32289] logger.c: -- Executing ResetCDR("SIP/000B8219B7as-08420e00", "w") in new stack
Mar 20 19:40:18 VERBOSE[32289] logger.c: -- Executing NoCDR("SIP/000B8219B7as-08420e00", "") in new stack
Mar 20 19:40:18 NOTICE[32289] cdr.c: CDR on channel 'SIP/000B8219B7as-08420e00' lacks end
Mar 20 19:40:18 VERBOSE[32289] logger.c: -- Executing GotoIf("SIP/000B8219B7as-08420e00", "1?5") in new stack
Mar 20 19:40:18 VERBOSE[32289] logger.c: -- Goto (internal,h,5)
Mar 20 19:40:18 VERBOSE[32289] logger.c: -- Executing Hangup("SIP/000B8219B7as-08420e00", "") in new stack
Mar 20 19:40:18 VERBOSE[32289] logger.c: == Spawn extension (internal, h, 5) exited non-zero on 'SIP/000B8219B7as-08420e00'
Mar 20 19:40:18 VERBOSE[29943] logger.c: Extension Changed 102 new state Idle for Notify User 000B821C4Cxy
-snap-

what happend here? how to interpret this log-entries?
our provider did a small adjustment of our dsl-modem (8192/1024) ... maybe this was the reason - but i dont believe that really.

also ... quality of calls is poor. which codecs do we use. the ones in grandstream-configuration or in trixbox config. i cant figure that out.....

codec-settings of Grandstream GXP 2000
[img=http://img23.imageshack.us/img23/7366/20090320screenshotcodec.th.jpg]

Einstellungen des VOIP-Accounts in Trixbox
[img=http://img25.imageshack.us/img25/5999/20090316rbtrixboxproopt.th.jpg]



3) theres a big problem too. the plan under "edit call menu" doesnt work. i cant figure out why. please have a look at the screenshots. if i call from outside ... and theres a schedule (business hours...) ...it doesnt matter when i call.... nor it rings on a extension which i choose. its always ringing on ONE extension(102). i switched some extension in a blast-group. if i try "blast ring ...group" ...theres also oen phone ringing (102). other phones in the blast-group arent ringing (phones can dial to outside and in lan).
also, before the upgrade, it was possible to call (with given extension) to a specific phone (numberxxxextension). since upgrade this isnt possible. i didnt change the configuration .... If i call "number" only ... one phone rings (102) - no blast call. if i call "number+extension" ...no phone rings ..but i can hear a tone like "modem" oder "fax". there are no entries in asterisk-logs so far i can see - so this must be a trixbox beheavior.

any suggestions?




Trixbox Pro SE settings after Upgrade 20090321 - Teil 1
[img=http://img16.imageshack.us/img16/5157/20090331settingafterupgd.th.jpg]

Trixbox Pro SE settings after Upgrade 20090321 - Teil 2
[img=http://img7.imageshack.us/img7/489/20090331settingafterupgb.th.jpg]


Extension in BLAST-Group, which is not ringing (Extension 101 - 102 is alwasys ringing)
[img=http://img17.imageshack.us/img17/162/20090321viewextensionso.th.jpg]


Extension - overview
[img=http://img19.imageshack.us/img19/5682/20090321extensionblastia.th.jpg]


Auto Answer - Edit Call Menu - "Call-Line" - overview MAIN
[img=http://img22.imageshack.us/img22/6501/20090321autoanswereditc.th.jpg]


Auto Answer - Edit Call Menu BLAST Group
[img=http://img21.imageshack.us/img21/8944/20090321autoanswersubme.th.jpg]

VOIP-Provider settings (Options - VOIP)
[img=http://img25.imageshack.us/img25/5999/20090316rbtrixboxproopt.th.jpg]



so, conclusion:

- how to change the tone which somebody can hear in his phone whenever he/she's calling us? (and not believing to call to far east...)
- what about the direct in calls? why isnt that working right after upgrade / not changed configuration?
- what about this fu**ing call menu. why is always phone extension 102 ringing and not blast? why are schedules not working (is this due to timezone? i really dont expect that, because i changed time ...we are in middle europe ... it seems to be a call menu problem only)?

hoping for useful hints !

thank you
NOR

2/ Couple of possibilities with your call dropping problem, in no particular order:-
a/ Your internet connection isn't up to it and is suffering from packet loss, not unusual on contended and congested ADSL connections. A more compressed codec (G729) might help if things aren't bad, but it really needs the supplier to apply some level of QoS that favours SIP
b/ Your server isn't up to it:
Virtualisation is not an ideal way to run vanilla Asterisk let alone Trixbox and Windows Server will have already gobbled up half of your RAM. If either OS is spooling to virtual memory then you will have problems, though to be fair this tends to show more often as garbled speech

3/ Check the horribly obvious things twice with regard to the phone that isn't ringing.
Is DND active on the phone?
Can you ring the extension specifically?
Has the user put the handset down properly?
Check the full log for an inbound call, you should see whether Trix is trying to ring the extension at all

4/ Trixbox can convert the various signal codecs in to another format, so if your supplier sends voice in G723, and your phone only supports G726 Trixbox can do the relevant transcoding.
As you are virtualising, the less Trixbox has to do, the better the odds of things working, so the one you need to worry about is what your VoISP can supply
If they only offer G711 ULAW/ALAW then that is what you need to set in both Trix and phones.
It is worth splashing the $10 each for some G729 licenses from Digium if they support that

Thanks for your reply

rgower :

2/ Couple of possibilities with your call dropping problem, in no particular order:- a/ Your internet connection isn't up to it and is suffering from packet loss, not unusual on contended and congested ADSL connections. A more compressed codec (G729) might help if things aren't bad, but it really needs the supplier to apply some level of QoS that favours SIP b/ Your server isn't up to it: Virtualisation is not an ideal way to run vanilla Asterisk let alone Trixbox and Windows Server will have already gobbled up half of your RAM. If either OS is spooling to virtual memory then you will have problems, though to be fair this tends to show more often as garbled speech

I think that RAM isn't correlated with our problem.
Our DSL-Provider changed some values (i guess priority of pakets) on our DSL-modem. Maybe this will result in no more (or less) canceled calls...hope so..

rgower :

3/ Check the horribly obvious things twice with regard to the phone that isn't ringing. Is DND active on the phone? Can you ring the extension specifically? Has the user put the handset down properly? Check the full log for an inbound call, you should see whether Trix is trying to ring the extension at all

We solved this problem. It due to the phone number. the phonenumber ends with xxxxx102. And the Extension number was 102 ... i dont know why, but this results in a ringing extension 102, even when it wasnt in the blast group. we changed a setting in ... global id-no extension direct routing or so ... added separate phone numbers for each extension (xxx102101, xxx102102, etc) and told the each extension in tribox to use 'her' specific number. this solved our problem!

rgower :

4/ Trixbox can convert the various signal codecs in to another format, so if your supplier sends voice in G723, and your phone only supports G726 Trixbox can do the relevant transcoding. As you are virtualising, the less Trixbox has to do, the better the odds of things working, so the one you need to worry about is what your VoISP can supply If they only offer G711 ULAW/ALAW then that is what you need to set in both Trix and phones. It is worth splashing the $10 each for some G729 licenses from Digium if they support that

thanks ... i will check this stuff ...


NOR