VOX SIP Account on Asterisk or Elastix Yahclick System

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Rory Pohl

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I am at wits end trying to get this going. Has anybody successfully been able to implement an Elastix box on the Yahclick system. The latency is too much to use a Voice Provider of my choice in South Africa so we are forced to use VOX as the service provider, also because of the different ports they use on their system. When I connect a VoIP phone directly through the internet to VOX we are able to make incoming and outgoing calls perfectly. When running on the PBX we can make outgoing calls and have no hassle what so ever in regards with outgoing calls. I am getting one way speech on the incoming calls,users in office are not able to hear the caller phoning in to the system but I am able to hear them clearly, however it does sometimes work and seems to be an intermittent problem but mostly I get the one way speech. I am also purchased a paid for g729 codec from Digium. VOX are not very helpful or not willing to assist I am hoping someone could guide me in the right direction.

This is my configuration.

Trunk Name: OutTrunk

PEER
host=196.41.24.53
username=SIP number
secret=SIP password
nat=no
insecure=very
fromuser=SIP number
fromdomain=yc.vphone.co.za
dtmfmode=auto or rfc2833
type=peer
qualify=yes
trunk=yes
trunktimestamps=no
disallow=all
allow=g729
context=from-trunk



USER Context: SIP number
Incoming Settings
secret=Password of SIP
type=user
context=from-trunk
nat=no
insecure=very
host=196.41.24.53
fromdomain=yc.vphone.co.za
dtmfmode=auto or rfc2833
disallow=all
allow=g729

Register String: SIP number:SIP [email protected]:5070/SIP number
 
If your router has SIP ALG or SIP Passthrough enabled, turn it off.
 
Obvious question, but can you make calls 'locally' via the PABX from one SIP phone to another without a problem?
 
This is registering over the internet. You will have issues as they can't guarantee QoS. If you can afford to move to their mpls and have your dedicated vrf, do that
 
I have been using it for a long time and it works 100% below is what I have...

Trunk Name: 2787XXXXXXX
Outbound Caller ID: 2787XXXXXXX

Trunk Name: 2787XXXXXXX

Peer Details:
username=2787XXXXXXX
type=peer
qualify=yes
secret=YOUR-PASSWORD
nat=yes
insecure=invite,port
host=196.3.177.10
fromuser=2787XXXXXXX
fromdomain=196.3.177.10
dtmfmode=rfc2833
context=from-trunk
disallow=all
allow=g729
canreinvite=no

User Context: (leave blank)
User Details: (remove anything and leave blank)

Register String: 2787XXXXXXX:[email protected]/2787XXXXXXX

Create an inbound route using your VOX number as the DID Number e.g. 2787XXXXXXX, and choose a description anything you like. leave everything else blank choose an extension for inbound route...

Also you can use g729, alaw or g722 with VOX......
g729 needs a license from Digium

If you need your SIP Password speak to Lesley Liebenberg [email protected]
0878059111 choose the option for dealer support and ask for Lesley... He is very Helpful.

The above is over ADSL & Diginet.

But in Asterisk/FreePBX you need to make sure of the following.

Under Tools -> Asterisk SIP Settings.

NAT = YES
IP Configuration = Dynamic IP (or Public IP)
Local Network = Must match local IP range
Codecs = Tick - G729 only (unless you need others for internal phones)
Registration Times = 90 (minexpiry) - 120 (maxexpiry) - 105 (defaultexpiry)

Next the field "qualify=yes" change this to "qualify=10000" this basically tells the PBX to wait 10 seconds before saying the line is unavailable. It is useful for high latency links.
I forgot, 1 way audio problems can usually be solved by adding "canreinvite=no"

Also I notice you populated the "User Context" it must be left blank, as the "register string" is used for incoming calls.

Hope that helps
 
Last edited:
Thanks for your reply
This is another thing which I am actually cheesed off about is the fact that in my discussions with VOX (before purchasing), from my understanding is that we would have been supplied with a CPE with a separate voice port and one for data (My understanding of that was 2 physical Ethernet Ports). The CPE which was supplied to us is the (Huges) HN9260 router (which only comes with one physical Ethernet port) and in my discussions with VOX (when we started experiencing issues) they assured me that this router is correct / ok and the port 5070 is still prioritized with that unit and it can just be connected to your switch.

In a another separate conversion to different technical consultant he said to be it definitely a firewall / port forwarding issue and started guiding me over the phone to open up all traffic to my pbx and the relevant ports etc. That's when we discovered that we are running the HN9260 , which you actually cannot configure and it seems this is a very basic unit in terms of CPE.

So to answer your question , no , unfortunately access is basic / limited to the router and controlled by VOX and I wont be able to tell you if these settings are enabled or disabled.

Thanks
 
Hummercellc said:
I have been using it for a long time and it works 100% below is what I have...

Trunk Name: 2787XXXXXXX
Outbound Caller ID: 2787XXXXXXX

Trunk Name: 2787XXXXXXX

Peer Details:
username=2787XXXXXXX
type=peer
qualify=yes
secret=YOUR-PASSWORD
nat=yes
insecure=invite,port
host=196.3.177.10
fromuser=2787XXXXXXX
fromdomain=196.3.177.10
dtmfmode=rfc2833
context=from-trunk
disallow=all
allow=g729
canreinvite=no

User Context: (leave blank)
User Details: (remove anything and leave blank)

Register String: 2787XXXXXXX:[email protected]/2787XXXXXXX

Create an inbound route using your VOX number as the DID Number e.g. 2787XXXXXXX, and choose a description anything you like. leave everything else blank choose an extension for inbound route...

Also you can use g729, alaw or g722 with VOX......
g729 needs a license from Digium

If you need your SIP Password speak to Lesley Liebenberg [email protected]
0878059111 choose the option for dealer support and ask for Lesley... He is very Helpful.

The above is over ADSL & Diginet.

But in Asterisk/FreePBX you need to make sure of the following.

Under Tools -> Asterisk SIP Settings.

NAT = YES
IP Configuration = Dynamic IP (or Public IP)
Local Network = Must match local IP range
Codecs = Tick - G729 only (unless you need others for internal phones)
Registration Times = 90 (minexpiry) - 120 (maxexpiry) - 105 (defaultexpiry)

Next the field "qualify=yes" change this to "qualify=10000" this basically tells the PBX to wait 10 seconds before saying the line is unavailable. It is useful for high latency links.
I forgot, 1 way audio problems can usually be solved by adding "canreinvite=no"

Also I notice you populated the "User Context" it must be left blank, as the "register string" is used for incoming calls.

Hope that helps
Click to expand...

Thanks , I am going to try / check these settings next week and get back to you
 
Still no luck on my side. I have tried the settings above exactly as you specified and still seem to be getting 1 way speech from time to time (basically at least 80% of the time). The only thing I that was different was the following:

host=196.41.24.53 (for Yahclick users) or yc.vphone.co.za
fromdomain=196.41.24.53

Register String has to be 2787XXXXXXX:[email protected]:5070/2787XXXXXXX (you have to specify port 5070 or it will not register)

Just for the record VOX initially told be to add the relevant fields to the User Context field. I could not edit the "Under Tools -> Asterisk SIP Settings" as I don think Elastix comes with this Add on Module, however I have specified the allow=g729 & disallow all under the sip_custom.conf & sip_general_custom.conf files.

Also just to answer some of the questions above , local / internal calls are fine. We have indeed purchased g729 codec's , calls always hit the pbx so definitely not connection issues. I am thinking of purchasing a conventional pbx but this would be sad because then I would loose the flexibility of the Elastix system as well as the benefit of the telephone recording. Or else try different distribution Asterisk Now / Freepbx or maybe even Freeswitch , any thoughts would be greatly appreciated.
 
Rory Pohl said:
Still no luck on my side. I have tried the settings above exactly as you specified and still seem to be getting 1 way speech from time to time (basically at least 80% of the time). The only thing I that was different was the following:

host=196.41.24.53 (for Yahclick users) or yc.vphone.co.za
fromdomain=196.41.24.53

Register String has to be 2787XXXXXXX:[email protected]:5070/2787XXXXXXX (you have to specify port 5070 or it will not register)

Just for the record VOX initially told be to add the relevant fields to the User Context field. I could not edit the "Under Tools -> Asterisk SIP Settings" as I don think Elastix comes with this Add on Module, however I have specified the allow=g729 & disallow all under the sip_custom.conf & sip_general_custom.conf files.

Also just to answer some of the questions above , local / internal calls are fine. We have indeed purchased g729 codec's , calls always hit the pbx so definitely not connection issues. I am thinking of purchasing a conventional pbx but this would be sad because then I would loose the flexibility of the Elastix system as well as the benefit of the telephone recording. Or else try different distribution Asterisk Now / Freepbx or maybe even Freeswitch , any thoughts would be greatly appreciated.
Click to expand...

Very strange :wtf:

It should work, I use elastix 2.4 and I have the SIP Settings Module by default.

I have PM'ed you settings, just place number and password...
 
One way audio is usually a NAT issue. Make sure your NAT is setup properly everywhere - especially your router!. You can try nat=force_rport,comedia instead of nat=yes.

Seeing that you are using a satellite connection try and enable jitter buffer. You might have to go to FreePBX unembedded mode to do this and check under Advanced - Sip Settings. Set your jitter buffer to: Enabled, Force, 200ms

Hope this helps
 
Rory Pohl said:
I am at wits end trying to get this going. Has anybody successfully been able to implement an Elastix box on the Yahclick system. The latency is too much to use a Voice Provider of my choice in South Africa so we are forced to use VOX as the service provider, also because of the different ports they use on their system. When I connect a VoIP phone directly through the internet to VOX we are able to make incoming and outgoing calls perfectly. When running on the PBX we can make outgoing calls and have no hassle what so ever in regards with outgoing calls. I am getting one way speech on the incoming calls,users in office are not able to hear the caller phoning in to the system but I am able to hear them clearly, however it does sometimes work and seems to be an intermittent problem but mostly I get the one way speech. I am also purchased a paid for g729 codec from Digium. VOX are not very helpful or not willing to assist I am hoping someone could guide me in the right direction.

This is my configuration.

Trunk Name: OutTrunk

PEER
host=196.41.24.53
username=SIP number
secret=SIP password
nat=no
insecure=very
fromuser=SIP number
fromdomain=yc.vphone.co.za
dtmfmode=auto or rfc2833
type=peer
qualify=yes
trunk=yes
trunktimestamps=no
disallow=all
allow=g729
context=from-trunk



USER Context: SIP number
Incoming Settings
secret=Password of SIP
type=user
context=from-trunk
nat=no
insecure=very
host=196.41.24.53
fromdomain=yc.vphone.co.za
dtmfmode=auto or rfc2833
disallow=all
allow=g729

Register String: SIP number:SIP [email protected]:5070/SIP number
Click to expand...



i found the issue and it work.
enter details as below

username=2787802####
fromuser=2787802####
secret=S0Se#####r
host=41.193.38.17
fromdomain=41.193.38.17
type=friend
context=from-trunk
insecure=port,invite
trustrpid=yes
sendrpid=yes
directmedia=no
qualify=yes
keepalive=45
nat=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw&alaw
 
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