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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: del on September 27, 2008, 04:43:06 PM
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Is anyone using voicenetwork.ca on Sail? I recently attemted to use them and added them as a carrier and the wierd thing is when I create a trunk it connects and shows as registered their end but when I call the DiD number it says I'm off line :? I spoke to their support but they don't have any knowledge of Sail and they said that the call is going out from them and then getting bounced back to them :shock: Is there some command that I can use to see what is happening my end? They told me to set it up as per their Trxbox setup and to make sure that insecure was set to very and to allow there IP access to my server :? I have tried as SIP and IAX2, I can't get it to register as an IAX2 :( below is the example SIP setup for Trixbox
PEER DETAILS:
host=sipgw.voicenetwork.ca
context=from-trunk
type=friend
username=1390976565
fromuser=1390976565
sendrpid=yes
qualify=yes
canreinvite=no
insecure=port,invite
secret=password
USER DETAILS:
type=peer
host=sipgw.voicenetwork.ca
qualify=yes
canreinvite=no
context=from-trunk
and the Asterisk example:
This register line must appear after the [global] section
; and BEFORE any other peer definitions
register => 1390976565:password@sipgw.voicenetwork.ca/1390976565
;
; Make sure that you have checked the Send DID option under the EDIT DID option
;
; Edit your SIP.CONF file and place the following lines in it
[voicenetwork-in]
type=peer
host=sipgw.voicenetwork.ca
fromuser=1390976565
dtmfmode=rfc2833
qualify=yes
canreinvite=yes
context=voicenetwork-incoming ; incoming DID calls will arrive in the voicenetwork-incoming context
[voicenetwork-out]
host= sipgw.voicenetwork.ca
type=friend
username=1390976565
fromuser=1390976565
fromdomain=voicenetwork.ca
sendrpid=yes
qualify=yes
secret=password
dtmfmode=rfc2833
disallow=all
allow=ulaw
context=voicenetwork-incoming ; incoming DID calls will arrive in the voicenetwork-incoming context
insecure=port,invite
;
; Edit your EXTENSIONS.CONF and place the following information in it
;
[voicenetwork-incoming]
exten => _X.,1,Answer
exten => _X.,2,Wait(1)
exten => _X.,3,SayDigits(${EXTEN})
exten => _X.,4,Wait(1)
exten => _X.,5,Echo
exten => _X.,6,Hangup
; To use VoiceNetwork.ca to termination your calls
; add the following line to your extensions.conf file
;
exten => _X.,1,Dial(SIP/voicenetwork_out/${EXTEN})
Thanks in advance,
Del
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Hi Del,
They've told you what they need...
A registration string and insecure=yes (or "port, invite" for 1.4).
So....
register => 1390976565:password@sipgw.voicenetwork.ca/1390976565
host=sipgw.voicenetwork.ca
context=mainmenu
type=peer
username=1390976565
fromuser=1390976565
sendrpid=yes
qualify=yes
canreinvite=no
insecure=port,invite
secret=password
Modify your carrier entry to include the registration string (username:password@sipgw.voicenetwork.ca/username) and make sure your peer stanza looks like the one above.
Best
S
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Hi Selintra,
I have done all that and it still does not ring in, I still get their message telling me I am off line :o
On my trunk I have (in the left box)
host=sipgw.voicenetwork.ca
context=from-trunk
type=friend
username=1390976565
fromuser=1390976565
sendrpid=yes
qualify=yes
canreinvite=no
insecure=port,invite
secret=password
and in the righthand box:
type=peer
host=sipgw.voicenetwork.ca
qualify=yes
canreinvite=no
context=from-trunk
The registration string is:
1390976565:password@sipgw.voicenetwork.ca/1390976565
Can you see any thing I am doing wrong? Thanks
Del
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Leave the righthand box empty.
In the lefthand box your context should be mainmenu and the type should be peer (look at my example).
I'm also assuming that your registration string contains your real password (and not just the string "password").
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Hi Selintra,
This is what I have now,
Righthand Box:
host=sipgw.voicenetwork.ca
context=mainmenu
type=peer
username=1390976565
fromuser=1390976565
sendrpid=yes
qualify=yes
canreinvite=no
insecure=port,invite
secret=password
Registration String:
1390976565:password@sipgw.voicenetwork.ca/1390976565
I still get told that I am offline when I ring the number, so I don't see what I am doing wrong :???:
If I log into my account the peer shows as connected and if I click on the state icon in server manager I get this:
* Name : 1390976565
Secret : <Set>
MD5Secret : <Not set>
Context : mainmenu
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
FromUser : 1390976565
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : No
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : port,invite
Nat : RFC3581
ACL : No
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : Yes
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost : sipgw.voicenetwork.ca
Addr->IP : 64.34.135.88 Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username: 1390969200
SIP Options : (none)
Codecs : 0xc (ulaw|alaw)
Codec Order : (alaw:20,ulaw:20)
Auto-Framing: No
Status : OK (55 ms)
Useragent :
Reg. Contact :
Password is the real password.
Thanks again for your help
Del
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All looks good...
What does "sip show registry" show?
If it is registered then turn all your phones off except one. At the asterisk console turn on sip debugging with
sip set debug
Run an inbound call and grab any output and post it to me at admin@aelintra.com
Turn debugging off with
sip set debug off
thx
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Hi Selintra,
I get this message:
sip: command not found
for all of the commands you have listed :shock:
I am logged in as root, do I need to change directories?
Sorry I am being a bit slow here :( but I promise to try and get up to speed :)
Del
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at the asterisk console Del.
You are typing into the linux console.
asterisk -rvvvv
Then try your commands again
Best
S
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Hi Selintra,
I have done that, the report is on it's way to you :)
I saw the word "destroy" in the debug and I am guessing that is not good news :shock:
Thank you for taking the time to look at this for me.
Del