TUTORIAL IPKall

[mruge]

I have seen many different ways to setup IPKall directly into my asterisk pbx.

Which way is the best way to set this up?

Thank you,
Michael

 

[mruge]

Ok, i got it to work except for one big problem. I am getting one way audio. It outputs audio, but I there is no input.

I am trying to set it up to work as a vm dialin number. The system does not hear the dtmf tones.

I have the NAT funtions turned on in the sip_nat.conf file.

Here are my incoming settings for the trunk:

User Context: from-trunk
 
allow=ulaw&alaw
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=voiper.ipkall.com
insecure=very
type=peer
nat=yes

The ports are open on my router. Other SIP trunks work without a hitch.

Any ideas?

Thanks!
Michael

 

[rmavillarica]

Not sure if this will help- my settings are almost exactly the same as yours but it works quite well most of the time:

Outgoing settings (Trunkname ipkall):
type=peer
qualify=yes
nat=yes
insecure=very
host=voiper.ipkall.com
dtmfmode=rfc2833
disallow=all
allow=ilbc&ulaw&gsm


Incoming settings (Trunkname ipkallin):
type=peer
nat=yes
insecure=very
host=voiper.ipkall.com
dtmfmode=rfc2833
disallow=all
allow=ulaw&gsm&ilbc


In ipkall.com, I think I have the standard setup:
SIP phone number: 360xxxxxxx
SIP proxy: myaddress.dyndns.com

And I have the usual stuff in sip_nat.conf:
externhost=myaddress.dyndns.com
localnet=192.168.1.0/255.255.255.0
nat=yes
externrefresh=60


Just noticed that ipkall has an IAX tick box. No idea how to set that up though.

Hope this helps.

Martin

 

[cliff]

I didn't setup a trunk for IPKall. I "Allow Anonymous Inbound SIP Calls" in general settings and have an "inbound route" set up for the number. I list that number in "SIP Phone Number:" on my IPKall setup page. This has worked for 18 months, routes to a ring group, forward or extension and responds very quickly. My experience is IPKall works as well or better than most paid providers.

 

[kwest]

Ok, i got it to work except for one big problem. I am getting one way audio. It outputs audio, but I there is no input.

I am trying to set it up to work as a vm dialin number. The system does not hear the dtmf tones.

I have the NAT funtions turned on in the sip_nat.conf file.



Any ideas?

Thanks!
Michael

Did anyone ever figure out this problem? I have tried everything!

 

[someguysname]

I used IAX instead of sip...no audio problems at all.

 

[kwest]

I used IAX instead of sip...no audio problems at all.

Thanks, yeah i tried that, changed it in my account but just get busy signal now, I did change the incoming trunk to IAX but still no connect, will work on it some more

 

[someguysname]

I set the IAX proxy on IPKALL web page to read xx.xx.xx.xx/1234567890 where xx.xx.xx.xx is my external IP address and 1234567890 is my IPKALL number. If you're using dynamic DNS just use your dns name instead of the IP address. I forwarded port 4569 from the outside interface of my router to my asterisk box and put an acl on it to only allow connections from IPKALL on that port.

The freepbx setup was just a trunk with the name as my IPKALL number and then the user context set as from-trunk and the details as:

allow=g729
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=voiper.ipkall.com
insecure=very
type=peer

and I also have this in the iax_custom.conf, I;m not sure if both were needed, I set it up back when I first started with asterisk and never bothered to see which one or if both made it work:

[1234567890] <-- IPKALL number
type=user
insecure=very
context=from-trunk
auth=plaintext
secret=ipkallloginpassword

 

[mspieker]

IP Kall is the only provider that I've had one way audio problems with. I too had to set it up as IAX to fix that problem. I have also had problems with them for quite some time now, that all my calls don't go through. Some callers just get dead air, again they are the only provider I've had this problem with. But Hey THEY ARE FREE

 

[Boolah]

In order for IPKall to work correctly with SIP, you will need to forward UDP port 5060 and UDP ports 10000-20000 (by default) to your asterisk box. Additionally, you will need to set "Allow Anonymous Inbound SIP Calls" to YES on the General Settings page.

 

[kwest]

IP Kall is the only provider that I've had one way audio problems with. I too had to set it up as IAX to fix that problem. I have also had problems with them for quite some time now, that all my calls don't go through. Some callers just get dead air, again they are the only provider I've had this problem with. But Hey THEY ARE FREE

May I ask what are you trunk settings? I cannot get mine to work caller receives busy.I tried some examples I found on the web but no go.

 

[completetech]

outgoing peer details:

type=peer
qualify=yes
nat=yes
insecure=very
host=voiper.ipkall.com
dtmfmode=rfc2833
disallow=all
allow=ilbc&ulaw&gsm

incoming user details:

type=peer
nat=yes
insecure=very
host=voiper.ipkall.com
dtmfmode=rfc2833
disallow=all
allow=ulaw&gsm&ilbc

user context = phone number from ipkall


then i just setup a second trunk with a little different name to allow the voiper2.ipkall.com address

works perfect everytime. i dont reall use it like that much but works great though

 

[kwest]

outgoing peer details:

type=peer
qualify=yes
nat=yes
insecure=very
host=voiper.ipkall.com
dtmfmode=rfc2833
disallow=all
allow=ilbc&ulaw&gsm

incoming user details:

type=peer
nat=yes
insecure=very
host=voiper.ipkall.com
dtmfmode=rfc2833
disallow=all
allow=ulaw&gsm&ilbc

user context = phone number from ipkall


then i just setup a second trunk with a little different name to allow the voiper2.ipkall.com address

works perfect everytime. i dont reall use it like that much but works great though

With these settings you used IAX? I just get busy signal, I am starting to think it is not worth it, every provider I have ever setup work just fine except ipkall, pain in the %^$#@

 

[aster1sk]

Nat?

May I suggest reading this :

http://pbxinaflash.com/community/threads/alternative-nat-ip-sh-tool.3075/?t=3075

 

[mspieker]

Again, for me, IPKall is the only trunk I've had problems with, 1 way audio even though I've done all the one way audio fixes out there. Those fixes DID fix my problems with other providers but not ipkall. It wasn't until I changed it to IAX that it started working. Now the only problem is if it has been a while since someone dialed the number, an incoming call may take up to a minute to actually ring through. Most people hang up with that amount of time of dead air.

completetech, I'm not sure why but your settings didn't work for me either, I too just a busy signal

kwest, this is what worked for me, keep in mind I'm a newb, and I'm not certain what is needed and what isn't, but it works for me =)

for ipkall IAX trunk
in trunks
trunk name =>ipcall (yes I used a c, I don't remember why)

peer details
type=peer

user context=>ipkall
type=user
secret=*******My ipkall web password*****
insecure=very
context=from-trunk
auth=plaintext

then on the ipkall website
IAX User Name:ipkall
IAX Proxy: *****My dyndns.org fqdn***** followed by /XXXXXXXXXX this being the ip kall number, this is what allows asterisk to recognize the ipkalls DID number
email is my email address
password: my password used above as secret, which is also my ipkall website login password
seconds to ring:120

Like I said, I'm a newb and this may be totally jacked, but it works for me.
Hope it helps

 

[completetech]

i am using SIP not IAX.

 

[Dieter Preiser]

I didn't setup a trunk for IPKall. I "Allow Anonymous Inbound SIP Calls" in general settings and have an "inbound route" set up for the number. I list that number in "SIP Phone Number:" on my IPKall setup page. This has worked for 18 months, routes to a ring group, forward or extension and responds very quickly. My experience is IPKall works as well or better than most paid providers.

That is correct. I've been running PIAF on Raspberry Pi and Raspberry Pi2 without setting up a trunk, only inbound routes. Be sure you have the IPKall IP address or domain name in IPTABLES.