Sip and Xlite problem

Asterisk Asterisk Support
1

I have changed my context from what I have read in these forums and in the Oreily Book. But I cannot call in or out. My Xlites can’t even call each other. Can someone tell me what I am doing wrong.

Sip.conf

context=default
allowguest=yes
allowoverlap=yes
allowtransfer=yes
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
pedantic=no
tos_sip=cs3
tos_audio=ef
tos_video=af41
maxexpiry=3600
minexpiry=60
defaultexpiry=120
t1min=100
vmexten=voicemail
allow=ulaw
allow=ilbc
mohinterpret=default
mohsuggest=default
language=en
trustrpid=no
sendrpid=yes
usereqphone=yes
dtmfmode=rfc2833
compactheaders=yes
videosupport=yes
maxcallbitrate=384
callevents=no
alwaysauthreject=yes
g726nonstandard=yes
rtptimeout=60
rtpholdtimeout=300
rtpkeepalive=100
sipdebug=yes
recordhistory=yes
dumphistory=yes
allowsubscribe=yes
subscribecontext=default
notifyringing=yes
notifyhold=yes
t38pt_udptl=yes
register=5598921147@sip.broadvoice.com:******:5598921147@sip.broadvoice.com/100
registertimeout=20
registerattempts=10
externhost=asterisk1.homedns.org
externrefresh=10
localnet=192.168.1.0/255.255.255.224
nat=yes
directrtpsetup=yes
canreinvite=update
rtcachefriends=yes
rtsavesysname=yes
rtupdate=yes
rtautoclear=yes
ignoreregexpire=yes
jbenable=yes
jbforce=no
jbmaxsize=200
jbresyncthreshold=1000
jbimpl=fixed
jblog=no

[sip.broadvoice.com]

type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=5598921147
secret=*********
username=5598921147
insecure=very
context=from-broadvoice
authname=5598921147
dtmfmode=inband
dtmf=inband
canreinvite=no
qualify=yes
[100]
context=phones
type=friend
username=100
secret=100
callerid=“Eddie” <100>
host=192.168.1.8
nat=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw
mailbox=100@default
qualify=yes

extensions.conf

[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1
include=daytime|9:00-17:00|mon-fri||
include=weekend||sat-sun||*
include=weeknights|17:02-8:58|mon-fri||
Eddie=SIP/100
Elaina=SIP/101
Val=SIP/102

[phones]
include=internal
include=outgoing

[from-broadvoice]

exten=100,1,Answer()
exten=100,n,Background(enter-ext-of-person)
exten=100,n,WaitExten()
exten=1,1,Dial (${Eddie},30)
exten=1,n,Playback(vm-nobodyavail)
exten=1,n,Hangup()

[internal]

exten=100,1,Dial (${Eddie})
exten=eddie,1,Dial(${Eddie})
exten=101,1,Dial (${Elaina})
exten=elaina,1,Dial (${Elaina})
exten=102,1,Dial (${Val})
exten=val,1,Dial (${Val})
exten=850,1,VoiceMailMain()
[outgoing]

exten=_1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_1NXXNXXXXXX, 2, congestion()
exten=_1NXXNXXXXXX, 102, busy()
exten=_01130.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01131.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01132.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01133.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01134.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011351.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011352.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011353.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011378.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01139.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01141.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011420.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01143.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01144.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01145.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01146.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01147.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01148.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01149[2-9].,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01154.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01155.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01156.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01160.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01161.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01164.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01165.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01181.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01182.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011852.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01186.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011886.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011972.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011.,2,congestion()
exten=_011.,102,busy()

This is my CLI output>

<— SIP read from 147.135.8.128:5060 —>
SIP/2.0 200 OK
Call-ID: 4813dd79635384a9436dea542c32d713@127.0.0.1
CSeq: 102 OPTIONS
From: “asterisk” sip:asterisk@127.0.0.1;tag=as4086c96d
To: sip:sip.broadvoice.com
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK754c074e;received=67.187.177.30;r port=5060
Supported: 100rel
Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
Content-Length: 0

CSeq: 102 OPTIONS
From: “asterisk” sip:asterisk@127.0.0.1;tag=as4086c96d
To: sip:sip.broadvoice.com
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK754c074e;received=67.187.177.30;rport=5060
Supported: 100rel
Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
Content-Length: 0

<------------->
— (12 headers 0 lines) —

Retransmitting #3 (NAT) to 192.168.1.8:5060:
OPTIONS sip:192.168.1.8 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK4e84573a;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as29e5dce3
t: sip:192.168.1.8
m: sip:asterisk@192.168.1.26
i: 3819414c0d5d5e802cd820113e940330@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 16 Feb 2008 18:48:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0

When I call in, I see it going through the motions in the CLI but nothing rings in.

My Xlite config.

username=100
password=****
auth user name=100
domain=192.168.1.26
domain proxy unchecked and target domain highligted

Topology all discover

I really want this to work. I have been using my 100 minutes doing test calls.

2

Yes, the problem was resolved- I had an old asterisk version 1.2.24 installed.

I just did

[b]ipkg uninstall asterisk

ipkg install asterisk14

reboot[/b]

Now I am having such an error:

[Feb 17 02:10:55] WARNING[593]: file.c:568 ast_openstream_full: File vm-theperson does not exist in any format [Feb 17 02:10:55] WARNING[593]: file.c:871 ast_streamfile: Unable to open vm-theperson (format 0x4 (ulaw)): No such file or directory

3

You need to install/upgrade the asterisk-sounds package.

:smile:

4

I have 1.4.17 package, but I don’t think I have sound issues. I think I have scripting issues, right?

5

Asterisk appears to be struggling to find the voice mail sound file ‘The person you are calling is…’

So either you have it installed in the wrong place or not at all.

I’ll have a look tonight and see where my asterisk sounds are located. I’m in a hurry to get to work just now.

6

I see, so you are saying it can not ring because the ringing sound is not available. OK, I will check my sound files

7

My sound files are in 'var/lib/asterisk/sounds/'
Check to see if you have a bunch of gsm files in that directory.
I have 1340 of them.

Connect to asterisk from a terminal with this command ’ asterisk -vvvvvvvgrc’ then receive a call.
You will see asterisk work through the extensions file. Any errors will show.

8

Thank you for the response, I have my * box up now.