Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: wifi on March 18, 2008, 01:51:02 PM
-
Now that I have setup SAIL (thank you selintra) on my SME server, I would like to start adding my Voip accounts.
One of my voip providers is Voipbuster (I do have an IN number).
Will / can SAIL work with Voipbuster?
I've tried adding the account (in Trunklines), and it does look like it's setup okay, but when I call my voip IN number I get a "does not exist" tone code (not sure if this is what you call it)
(http://www.data-hosting.eu/vip/001055/voipbuster.gif)
As you can see I've set the "Open Inbound Route" to "Leave Voicemail", so should this "does not exist" tone code not be some kind of "Leave Voicemail" option?
Is there some log file that I can look at to see what the error is, and if the Voipbuster account that I've setup is actually working?
-
select "edit" on your voipbuster trunk and add the following line somewhere in the left hand window...
insecure=very
try your call again...
If it doesn't work.
do the following....
log into linux and show me the output from the following two commands...
asterisk -rx "sip show registry"
asterisk -rx "sip show peers"
Thx
S
-
You MUST have at least one extension defined to the system (even if it isn't connected) and you MUST have that extension defined as the operator in globals.
If you've done that, then...
select "edit" on your voipbuster trunk and add the following line somewhere in the left hand window...
insecure=very
try your call again...
If it doesn't work.
do the following....
log into linux and show me the output from the following two commands...
asterisk -rx "sip show registry"
asterisk -rx "sip show peers"
Thx
S
-
Hi again,
Sorry for the late reply. I needed to go somewhere.
I've added the "insecure=very", and it now looks like this:type=peer
host=sip.voipbuster.com
fromdomain=sip.voipbuster.com
qualify=3000
canreinvite=no
username=my_user_name
fromuser=my_phone_nr_in_0031_format
secret=my_password
disallow=all
allow=alaw
allow=ulaw
insecure=very
(note that I have removed the username, fromuser and secret)
re: asterisk -rx "sip show registry"
[root@files ~]# asterisk -rx "sip show registry"
Host Username Refresh State Reg.Time
[root@files ~]#
re: asterisk -rx "sip show registry"
[root@files ~]# asterisk -rx "sip show peers"
Name/username Host Dyn Nat ACL Port Status
my_user_name/my_user_name 194.221.62.198 5060 OK (29 ms)
5000/5000 10.0.0.4 D 5060 OK (5 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
[root@files ~]#
(again I've changed the username)
To me it looks like something is wrong with sip show registry, as it's empty.
Thank you for looking after me.
-
ok
fromuser and username should be the same (your voipB user-id)
You need to add a registration string in the trunk (in the registration string field) and it should be...
{yourVoipBusername}:{yourVoipBpassword}@sip.voipbuster.com/{the inbound DiD which VoipB allocated to you}
DON'T include the curly braces, they are just there to delineate the various entries.
-
Hi,
This is what I now have (see) when I select Trunklines > Edit "Modify a Trunk or Gateway"
type=peer
host=sip.voipbuster.com
fromdomain=sip.voipbuster.com
qualify=3000
canreinvite=no
username=voipB user-id
fromuser=voipB user-id
secret=voipB password
disallow=all
allow=alaw
allow=ulaw
insecure=very
voipB user-id:voipB password@sip.voipbuster.com/0031300000000
(I've changed the voipB user-id, voipB password and 0031300000000)
asterisk -rx "sip show registry" is still showing nothing.
I'm 100% sure that the voipB user-id, voipB password and 0031300000000 are correct.
I feel so stupid.
-
voipB user-id:voipB password@sip.voipbuster.com/0031300000000
correct string, incorrect location - it goes into the "registration string" field above and to the right, not into peer window.
Kind Regards
S
-
Are you a machine? Do you never sleep?
Anyway. We are getting closer I think.
I'm still getting a "does not exist" sound when I call the number (sounds like tu ta taa)
This is how my "trunklines" looks like (when I click on edit).
(http://www.data-hosting.eu/vip/001055/voipbuster2.gif)
I've still a lot of things missing in it as you can see.
asterisk -rx "sip show registry" is now showing the following.
[root@files ~]# asterisk -rx "sip show registry"
Host Username Refresh State Reg.Time
sip.voipbuster.com:5060 voipB user-id 145 Registered Tue, 18 Mar 2008 20:29:34
[root@files ~]#
I've added some more SJphones to some PC's on the LAN, and I can call any phone on the system now (within the LAN)
Once again thank you. You are doing a great job.
If you give me a donate link I will make a donation.
-
OK - you're almost there. The final trick with carriers is to figure out what number they're delivering against and what number you are "matching" them to in your sip.conf. The final bit of the registration string (after the /) tells the carrier what number to deliver against. This should match the DiD you defined for the trunk when you created it. Now, just looking at your screenshot, the DiD (top left), which you've sensibly greyed out, doesn't look long enough to match your 0031xxxxxxx number so I'm guessing that you've specified the VoipB user-id here. Correct?
If this is true then simply change the tail of the registration string to match and you should be good to go. There is another way to do it... but we'll leave that for another day.
Kind Regards
S
-
selintra, your last post got it to work for 1/2 now :-)
re: so I'm guessing that you've specified the VoipB user-id here. Correct?
Yes I did. After recreating the profile (with the DiD) I got it to kind of work..
Calling the number does connect to my SJphone, and will show a popup ignore / answer.
When I click on answer, it looks / sounds like I (SJphone site) picked up the phone, but the calling person (non voip) still hears the phone ringing.
When I click on ignore, the calling person gets a message saying that I'm on the phone, and to leave a message.
Also. When I try to make a real call with the SJphone, I keep getting a "Call reject: 603 Declined"
I'm sure this has something to do with "Routes"..
So close, but still so far away :/
-
Okay.. Got a bit closer today :-)
Problem with the not being able to answer the call was that in sip.conf the localnet was set to: 192.168.1.0/255.255.255.0
My localnet is: 10.0.0.0/255.255.255.0
After changing this in Headers > sip.conf, calls made to my Voipbuster IN number are working (I can pickup the phone, and actually talk to the other person.. And yes.. They can hear me.. And yes I can hear them too)
Problem now is that when I make a call from the Voip phone (in this case X-Lite softphone) to a number outside my network I get a "call failed: declined".
I'm sure that this has something to do with Routes, but what?
-
Excellent... Well done. I'm impressed you found that localnet parameter, it catches a lot of people out.
So...
What have you defined in your route?
Best
S
-
I did some googling on the localnet part, and saw some example files of sip.conf
The hardest part was finding the correct sip.conf file.
Anyway. This is how my route looks like.
Please note that for the example I've Photoshoped the Primary Path number.
The official Primary Path number is the IN number that I got from Voipbuster.
When calling this number the X-lite softphone rings and works fine.
It's OUT that I've got a problem with.
(http://www.data-hosting.eu/vip/001055/routes.gif)
One more thing. PLEASE give me a donate link! I really want to give you some cash for helping me!
-
You have no dialplan in your route.
The route works by looking for number patterns. Have a look at this entry in the documentation
http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter112
Then google for asterisk dial plans.
The dial plan specifies number patterns that you want this route to catch and distribute. For example if I wanted to catch numbers which were 10 digits in length and always began with a zero then the dialplan would be _0XXXXXXXXX The underscore at the beginning tells asterisk that this is pattern.
If I knew what dutch telephone numbers looked like I might be able to help more but this above should point you in the right direction.
ps - you might want to turn authorization OFF otherwise it will ask you for a pin number when you attempt to dial out.
KInd Regards
S