* working but a few problems (mac os x)

[pix]

Greetings all from London UK,

This is my first post in this forum. I'm a relative newbie (2 weeks) to VoIP and Asterisk.

My Setup:
Mac OS X 10.3.x
G4 1.47MHz
1.5GB RAM
Static IP to * box
2x Grandstream 101
Asterisk CVS-10/28/03

This is what I want to have:

Incoming - voipuser (084498xxxxx) on extension 1000
Incoming - sipgate (0207) on extension 1001

Outgoing - voipuser extension #81
Outgoing - voipbuster extension #82
Outgoing - sipgate extension #83

Voicemail for both phones.
Music on hold

This is what I have:

Both phones can be contacted via SIP, PSTN or local extension number, and the correct phone rings.

Can leave voicemail on either phone but Asterisk crashes and will need restarting straight after.

After restarting Asterisk, you can hear the voicemails left but every message has a loud static noise at the beginning of each recording.

All voicemail boxes are announced as 0000 rather than 1000, 1001 etc

Can dial out via #81(voipuser) and make a call OK.

Dialing out via #82 or #83 (sipgate / voipbuster) doesn't work.

I did manage to get 1 call out via voipbuster once using the same settings but hasn't worked since. ( I left a message on a answer phone, the sound quality was awful.)

Using #82 to phone a BT landline gets this in CLI output:

-- Executing Dial("SIP/1000-cbfe", "SIP/0208xxxxxxx@voipbuster|60") in new stack
-- Called 0208xxxxxxx@voipbuster
== No one is available to answer at this time
-- Executing Congestion("SIP/1000-cbfe", "") in new stack
== Spawn extension (autocontext, #820208xxxxxxx, 2) exited non-zero on 'SIP/1000-cbfe'

Using #83 to phone a BT landline gets error 503 on the phone LCD but nothing is output in CLI.

I've tested both of the above ringing my BT line, the phone didn't ring on either attempt.

'sip show registry': shows all registered OK
'sip show peers': shows both phones, voipuser and voipbuster as "Unmonitored" but sipgate as "UNREACHABLE".

I noticed while testing that if I ring my normal BT landline and hang up the * phone before answering the call, the landline keeps ringing until it's picked up.

Voicemail is setup it seems to work fine, apart from a few things.

After a voicemail has been left the Asterisk server crashes and has to be relaunched via the CLI. This happens if voicemail has been left internally or externally.

When listening to voicemail messages, at the beginning of every message there is a burst of loud static noise. Sample of static http://www.tarling.info/*/static.wav.zip

All voicemail boxes are announced as 0000 rather than 1000, 1001 etc

How do I fix these errors?

Where am I going wrong?

Suggestions - comments, sample code welcome

Thanks in advance

Pix

config files below:

sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
context = autocontext
srvlookup = yes
disallow=all
allow=ulaw
allow=ilbc
allow=g729
allow=gsm
allow=g729
allow=gsm
allow=alaw
allow=all

;
;
register => <Username>:<Password>:< Username >@voipuser.org/1000
register => < Username (7 digits)>:<Password>:<Username (7 digits)>@sipgate.co.uk/<Username (7 digits)>/1001
register=><Username>:<Password>:<Username>@sip.voi pbuster.com


[sip.voipbuster.com]
type=peer
context=autocontext
host=sip.voipbuster.com
fromuser= <Username>
secret= <Password>
fromdomian=sip.voipbuster.com
dtmfmode=rfc2833
insecure=very

[sipgate]
type=friend
context=autocontext
fromuser=< Username (7 digits)>
username=< Username (7 digits)>
authuser=< Username (7 digits)>
secret=< Password >
host=sipgate.co.uk
fromdomain=sipgate.co.uk
dtmfmode=rfc2833
insecure=very
qualify=yes

[voipuser]
type=friend
context=autocontext
username=< Username >
secret=< Password>
host=voipuser.org
fromuser=< Username >
fromdomain=voipuser.org
insecure=very
qualify=no


[1000]
type=friend
secret=1000
host=dynamic
nat=yes
dtmfmode=rfc2833
reinvite=no
canreinvite=no
callerid="1000"<1000>
context = autocontext
mailbox = 1000

[1001]
type=friend
secret=1001
host=dynamic
nat=yes
dtmfmode=rfc2833
reinvite=no
canreinvite=no
callerid="1001"<1001>
context = autocontext
mailbox = 1001


extensions.conf

default extension.conf (do I need it all?), plus...

[incoming_voipuser]
exten => 084498xxxxx,1,NoOp(--- ${CALLERID} calling on VoIPUser (${EXTEN}) ---)
exten => 084498xxxxx,2,Dial(SIP/1000,20)
exten => 084498xxxxx,3,Answer
exten => 084498xxxxx,4,Wait,1
exten => 084498xxxxx,5,Voicemail(u1000)
exten => 084498xxxxx,6,HangUp


[incoming_sipgate]
exten => < Username (7 digits)>,1,NoOp(--- ${CALLERID} calling on Sipgate (${EXTEN}) ---)
exten => < Username (7 digits)>,2,Dial(SIP/1001,20)
exten => < Username (7 digits)>,3,Answer
exten => < Username (7 digits)>,4,Wait,1
exten => < Username (7 digits)>,5,Voicemail(u1001)
exten => < Username (7 digits)>,6,Hangup

[outgoing_voipuser]
exten => _#81.,1,Dial(SIP/${EXTEN:3}@voipuser,60)
exten => _#81.,2,Congestion

[outgoing_voipbuster]
exten => _#82.,1,Dial(SIP/${EXTEN:3}@voipbuster,60)
exten => _#82.,2,Congestion

[outgoing_sipgate]
exten => _#83.,1,Dial(SIP/${EXTEN:3}@sipgate,60)
exten => _#83.,2,Congestion

#include /etc/asterisk/autocontext.conf


autocontext.conf

;
[autocontext]
include => fwd-outgoing
include => vp-outgoing
include => demo

Do I need the above?

include => incoming_voipuser
include => outgoing_voipuser
include => outgoing_voipbuster
include => incoming_sipgate
include => outgoing_sipgate

;
; Connect to voicemmail from another phone
exten => _*1XXX,1,Voicemail(u${EXTEN:1})
exten => _*2XXX,2,Hangup

;
; Connect to voicemmail
exten => 1999,1,VoicemailMain(s${CALLERIDNUM})
exten => 2999,2,Hangup

;
; lower case letter o
; after an extension is reached, pressing zero
; starts voicemail
exten => o,1,voicemailmain

;
; Dialplan entry for user 1000
exten => 1000,1,NoOp(Incoming call for 1000 at ext. 1000)
exten => 1000,2,Dial(SIP/1000,20,tr)
exten => 1000,3,Voicemail(u${EXTEN})
exten => 1000,102,Voicemail(b${EXTEN})
exten => 1000,4,Hangup
exten => 1000,1,Ringing
exten => 1000,2,Wait(2)
exten => 1000,3,VoicemailMain(s${CALLERIDNUM})
exten => 1000,4,Hangup

;
; Dialplan entry for user 1001
exten => 1001,1,NoOp(Incoming call for 1001 at ext. 1001)
exten => 1001,2,Dial(SIP/1001,20,tr)
exten => 1001,3,Voicemail(u${EXTEN})
exten => 1001,102,Voicemail(b${EXTEN})
exten => 1001,4,Hangup
exten => 1001,1,Ringing
exten => 1001,2,Wait(2)
exten => 1001,3,VoicemailMain(s${CALLERIDNUM})
exten => 1001,4,Hangup


voicemail.conf

default voicemail.conf (do I need it all?), plus...

[zonemessages]
gmt=london|'vm-received' q 'digits/at' HM

[default]
1000=><Password>,<User>,<user at somedomain dot com>
1001=><Password>,<User>,<user at anotherdomain dot com>

[jerrydav]

I had the same problem with static when recording voicemail. I got it working by changing the format in voicemail.conf to gsm only (format=gsm). I am using version 1-0-7 on OSX

cheers.

[mberlant]

pix, you've done a rather good job in your configuration, but you have been sloppy with your [context] names. This is likely why calls are not getting routed properly.

Let's look at your configuration elements

[general]
port = 5060
bindaddr = 0.0.0.0
context = autocontext
srvlookup = yes
disallow=all
allow=ulaw
allow=ilbc
allow=g729
allow=gsm
;allow=g729 ; remove the duplication
;allow=gsm ; remove the duplication
allow=alaw
;allow=all ; Why bother disallowing all if you are going to allow them all?

;
;
register => <Username>:<Password>:< Username >@voipuser.org/1000
register => < Username (7 digits)>:<Password>:<Username (7 digits)>@sipgate.co.uk/<Username (7 digits)>/1001
register=><Username>:<Password>:<Username>@sip.voi pbuster.com/1002
Even if you don't want it to ring anywhere, you should point incoming calls somewhere.

[sip.voipbuster.com] <-- Should be [voipbuster], to match the sending context.
type=peer
context=autocontext
host=sip.voipbuster.com
fromuser= <Username>
secret= <Password>
fromdomian=sip.voipbuster.com
dtmfmode=rfc2833
insecure=very

[sipgate]
type=friend <--Should probably be type=peer
context=autocontext
fromuser=< Username (7 digits)>
username=< Username (7 digits)>
authuser=< Username (7 digits)>
secret=< Password >
host=sipgate.co.uk
fromdomain=sipgate.co.uk
dtmfmode=rfc2833
insecure=very
qualify=yes

[voipuser]
type=friend <--Should probably be type=peer
context=autocontext
username=< Username >
secret=< Password>
host=voipuser.org
fromuser=< Username >
fromdomain=voipuser.org
insecure=very
qualify=no


[1000]
type=friend
secret=1000
host=dynamic
nat=yes
dtmfmode=rfc2833
reinvite=no
canreinvite=no
callerid="1000"<1000>
context = autocontext
mailbox = 1000

[1001]
type=friend
secret=1001
host=dynamic
nat=yes
dtmfmode=rfc2833
reinvite=no
canreinvite=no
callerid="1001"<1001>
context = autocontext
mailbox = 1001


extensions.conf

default extension.conf (do I need it all?), plus... These two contexts are superfluous, since you have hard pointed the two incoming lines to 1000 and 1001, respectively, in the register= lines, above.

[incoming_voipuser]
exten => 084498xxxxx,1,NoOp(--- ${CALLERID} calling on VoIPUser (${EXTEN}) ---)
exten => 084498xxxxx,2,Dial(SIP/1000,20)
exten => 084498xxxxx,3,Answer
exten => 084498xxxxx,4,Wait,1
exten => 084498xxxxx,5,Voicemail(u1000)
exten => 084498xxxxx,6,HangUp


[incoming_sipgate]
exten => < Username (7 digits)>,1,NoOp(--- ${CALLERID} calling on Sipgate (${EXTEN}) ---)
exten => < Username (7 digits)>,2,Dial(SIP/1001,20)
exten => < Username (7 digits)>,3,Answer
exten => < Username (7 digits)>,4,Wait,1
exten => < Username (7 digits)>,5,Voicemail(u1001)
exten => < Username (7 digits)>,6,Hangup

[outgoing_voipuser]
exten => _#81.,1,Dial(SIP/${EXTEN:3}@voipuser,60)
exten => _#81.,2,Congestion

[outgoing_voipbuster]
exten => _#82.,1,Dial(SIP/${EXTEN:3}@voipbuster,60)
exten => _#82.,2,Congestion

[outgoing_sipgate]
exten => _#83.,1,Dial(SIP/${EXTEN:3}@sipgate,60)
exten => _#83.,2,Congestion

#include /etc/asterisk/autocontext.conf


autocontext.conf

;
[autocontext]
include => fwd-outgoing
include => vp-outgoing
include => demo

Do I need the above? I don't think so.

include => incoming_voipuser
include => outgoing_voipuser
include => outgoing_voipbuster
include => incoming_sipgate
include => outgoing_sipgate

;
; Connect to voicemmail from another phone
exten => _*1XXX,1,Voicemail(u${EXTEN:1})
exten => _*2XXX,2,Hangup Should probably be _*1XXX

;
; Connect to voicemmail
exten => 1999,1,VoicemailMain(s${CALLERIDNUM})
exten => 2999,2,Hangup As above, should probably be 1999

;
; lower case letter o
; after an extension is reached, pressing zero
; starts voicemail
exten => o,1,voicemailmain

;
; Dialplan entry for user 1000
exten => 1000,1,NoOp(Incoming call for 1000 at ext. 1000)
exten => 1000,2,Dial(SIP/1000,20,tr)
exten => 1000,3,Voicemail(u${EXTEN})
exten => 1000,102,Voicemail(b${EXTEN})
exten => 1000,4,Hangup
exten => 1000,1,Ringing <--These four lines are superfluous, I believe.
exten => 1000,2,Wait(2)
exten => 1000,3,VoicemailMain(s${CALLERIDNUM})
exten => 1000,4,Hangup

;
; Dialplan entry for user 1001
exten => 1001,1,NoOp(Incoming call for 1001 at ext. 1001)
exten => 1001,2,Dial(SIP/1001,20,tr)
exten => 1001,3,Voicemail(u${EXTEN})
exten => 1001,102,Voicemail(b${EXTEN})
exten => 1001,4,Hangup
exten => 1001,1,Ringing <--These four lines are superfluous, I believe.
exten => 1001,2,Wait(2)
exten => 1001,3,VoicemailMain(s${CALLERIDNUM})
exten => 1001,4,Hangup


See if these suggestions help. Your problem with outbound to voipbuster, for example, was that your [outbound_voipbuster] was sending to [voipbuster], but there was only [www.voipbuster.com] listening, so there was no match.

Good luck.

Michael

[pix]

Thanks for the quick replies to my problems. Sorry for delay in posting the outcome from the info provided, well here's a update.....

Thanks jerrydav,

I changed my format for voicemail to "format=gsm" and it fixed my static problem.:lol:
I found out while testing that "format=gsm|wav" also works, handy for voicemail messages sent via email to Mac users, they can now be opened with quicktime.

voicemail.conf

format=gsm|wav

mberlant,

Thank you also for your suggestions.

I removed all of the things you mentioned apart from

extensions.conf

[incoming_sipgate]
exten => < Username (7 digits)>,1,NoOp(--- ${CALLERID} calling on Sipgate (${EXTEN}) ---)
exten => < Username (7 digits)>,2,Dial(SIP/1001,20)
exten => < Username (7 digits)>,3,Answer
exten => < Username (7 digits)>,4,Wait,1
exten => < Username (7 digits)>,5,Voicemail(u1001)
exten => < Username (7 digits)>,6,Hangup

It seems that it is still needed for the sipgate account to work.

+++++++++++++++++++++++++++++++++++++++++++++++

I recreated all the mailboxes one at the time, and connected to each voicemail box in turn via my SIP phone.

Doing this on my Mac OS X * box seems to create voicemail boxes correctly.

Now when connected to voicemail all voicemail boxes are announced by the correct extenision, compared to "extension 0000" for all voicemail boxes, before the above changes where made.

autocontext.conf

;
; Connect to voicemail from another phone
exten => _*1XXX,1,Voicemail(u${EXTEN:1})
exten => _*1XXX,2,Hangup


I'm still having problems with....

Voicemail:

After a voicemail has been left, or even if it's just been accessed without leaving a message it always makes asterisk crash.

Anyone know why - what is causing this?

Outgoing calls:

Still can't call out via voipbuster.com or sipgate, but I think I need to purchase some outgoing credits for sipgate to work. Not sure what's stopping outgoing via voipbuster.

Suggestions?

And finally....

During all testing I've found that if I call a PSTN line and hang up before it gets answered.
The PSTN line that's been called keeps ringing until picked up, but of course no ones there because it's been hung up.

Annoying, not for me, but the person who I phoned will have a constantly ringing phone.

Fixes anyone?


Pix