Topic: grand stream fxo gateway gxw410 config

[rrkelly]

has anyone configured a gxw 4104 or 8 to run under sail. the only mention was in Italian and the translation did not appear to cover configuration. 

This part is for selintria:
have you ever heard of a configuration requiring a routable public ip for every phone? The sip provider claims that is how their switch is setup. below is what they tell me they have

Sylantro Firmware version 4.2 PB1
Session Border Controller Acme Packet
Switch Telica now Lucent Compact Switch now known as the Lucent-Alcatel Gateway Platform
 
This is a sip provider that for political reasons i'm stuck with. right now they are setting the roll over at the
switch and giving me a an adtran fsx interface which will be hardwired to grandstream 4 port fx0 to go back into sail which is completely nuts.

[bbialy]

incoming are easy to configure.
3 places
FXO Channels
Profile (ex. 1)

and most important
unconditional forward to Voip

as i remeber
ch1-4:[sail account]  @ p1:;   5060

i will check it today evening
and post config

[SARK devs]

4104/4108 set up is very straightforward.  Runs fine and sound quality is good.  We had problems detecting far-end hang-up because the unit was built to work under Bellcore (USA) type networks and couldn't detect ETSI K-break hangup.  Other than that its fine.

I have seen set-ups with all phones publicly addressable.  Usually it is used with low-function (i.e.not very clever) PBX or with pure SIP routers such as SER and openSER which rely on the phones themselves to do a lot of the switching work.  It also keeps load off the PBX by allowing SIP reinvites during transfer so that the phones communicate directly with one another without their media streams going through the switch.  It tends to switch very fast but is often incapable of even very basic PBX functions, such as call pick-up.  All of the set-ups I have seen used phones with very strong SIP functionality of their own (i.e. Snom) 

Best

S

   



 

[rrkelly]

im trying to figure out how i should set this up --  i need the 4 pots lines for incoming calls -- these are roll overs from a main number that will be forwarded to my sip provider  but i would also like to use them as needed for local calls. Do i make them extentions or trunks as far as sail is concerned?
thanks
rob

[SARK devs]

Always set a gateway up as a trunk.  The main difference is that this is s SIP trunk that will register with you (rather than the other way 'round) so you should set the trunk IP address as dynamic.  The SIP settings will be much the same as the ones we use for Multitech gateways so I would use that trunk template to get you going and then modify it to suit your needs.   For receiving calls, you will need to get the gateway to send in a fixed DDI of some description (we usually use the real PTT number of the line).  You can then just set up a PTT-DiD to receive and switch the inbound calls to wherever you want them to go.

Best

S
 

[rrkelly]

ok it semi works -- i used the multi tech howto in the manual and built a ptt did group
http://www.aelintra.com/docs/cgi-bin/view/Main/DocChapter2513

incoming on the pots is answered by the gxw4101 and i get a dial tone  if  try to dial a internal extention i.e. 3000# this is the error message

[Oct 24 15:01:02] NOTICE[5001]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '3000' rejected because extension not found.
since i at least get a err msg that woulld be progress. the question is where is the configure err in the gxw4104 or sail?

i left the multitech cfg intact



rob

[rrkelly]

ok i can route (dial) from an internal extension out thru the gxw4101 and over a pots line.  on the gxw i have dialing set to stage 1
i do not have any incoming to sail from the gxw4101 anymore
i have tried sip registration yes and no also wait for dial tone yes and no.
rob

[bbialy]

i wil have some tests tommorow
one of them will be with grandstream

send pls your config i simulate it and try to tell what to do

support@datech.pl

[jester]

bbialy,

Do you have a general config for the Grandstream that you can maybe post here on the forum?
You would happen to have a Grandstream gxw400x config as wel ?!   

[rrkelly]

bbialy
Thanks i do not have access to the machine right now. Here is what i think i know.
If the Gxw is set to 2 stage dialing  it will seize the ringing inbound pots give me a dial tone and talk to sail at least that is what the msg below says
(i set the name so the quotes are not empty any more)

[Oct 24 15:01:02] NOTICE[5001]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '3000' rejected because extension not found.

I used the multi tech template for the trunk carrier and a 4 ptt did group
with the  Gxw set to 1 stage dialing and a route set i can dial out to any of the 4 lines
from any of the internal extensions. I have tried to set in bound pots from the Gxw to an extension/ring group it appears that the line is not answered by the Gxw as it just keeps ringing and there is no traffic to asterisk that shows in the asterisk CLI.
The unconditional forward to voip appears to be set as it works with two stage dialing
i did not set any sip registration in sail for the 4 Gxw ports. The remaining problem appears to be notifying and connecting sail to an inbound pots call. Any ideas would be welcome
rob

 

[rrkelly]

ok how do i get sail to accept an in bound call from the gxw4104 this is the msg i get

[Oct 27 10:43:20] NOTICE[5001]: chan_sip.c:14035 handle_request_invite: Call from 'gxw4104' to extension '3000' rejected because extension not found.

the gxw4101 is set for single stage dialing ----out bound calling works fine -- inbound gives the above msg for all 4 dids
the trunk config  is
Trunk DID or IP Name: gxw4104    Carrier Name: MultiTech(MultiVoip

[peergxw4104]

type=peer
host=192.168.200.9
qualify=no
canreinvite=no
username=gxw4101
insecure=very
disallow=all
allow=alaw
allow=ulaw

[gxw4104]

type=peer
context=incoming
host=192.168.200.9
insecure=port

ptt did
5074731143     default     PTT_DiD_Group     LOCAL     N/A     N/A     DiD     None     3000     Operator

gxw4101 fxo unconditional forward cfg is

User id ch1-4:3000;     Sip server ch1-4:p1;     Sip destination port ch1-4:5060;

p1 = profile 1 has sip server and proxy set to 192.168.200.1  and sail seems to accept it


thanks rob

[SARK devs]

Hi Rob,

It looks like the Grandstream is sending 3000 in as a DiD to SAIL.  Up until very recently, SAIL would not accept a 4-digit DiD so you have to get the grandstream to send a longer digit stream.  As an example, you might have it send the full external number of the line (all 10 or 11 digits, depending upon your national PSTN).  You would then create a PTT-DiD-Group with the full number and, in turn point the PTT_DiD trunk at the extension you want to hit (in the OPEN or CLOSED inbound route).  Which looks like what you've tried to do, but I think you've sent the number in as 3000 instead of 5074731143.

However, in -670 (which is very new), we changed the rules so that you can create a 4-digit PTT_DiD.  You can also use something called "smartlink" which will look to see if there is an extension which matches the last four digits of the DiD and, if there is, automatically fill out the routing for you.  So if your Grandstream is sending 3000, then simply upgrade to 670 and create a DiD for 3000.  SAIL will automatically point it at exten 3000 (as long as you check the smartlink box in PTT_DiD create panel) and you should be good to go.

Hope that this helps

Kind Regards

S

[rrkelly]

thanks that is what i did(670 upgrade) -- i added a context in the config of the grandstream trunk after i looked at sip.conf   when i set the context to default it worked. i will try it with smart link and full did and see if that also works. It would be nice to add a config template for both the GWX4104 and 8. I would be willing to write up some notes if you think it will help.


type=peer
context=default
host=192.168.200.9
insecure=port

rob

[bbialy]

insecure shouldn't be port,invite ??
i think i saw it somewhere ingrandstream docs
bbialy