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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: zammelx on January 01, 2007, 03:42:25 PM
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Hi there, and happy new year! ;)
i'm giving a try to sail on sme7, i have a linksys/cisco ata device hooked to an analog gsm gateway with an fxo port, and wish to route calls with gsm prefixes through this adapter.
How could i configure sail to use the ata extension as a trunk?
thanks!
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HI,
Unfortunately, we've lost our main dsl link (AGAIN) and we're having to run over our backup which means that you can't currently access our docs pages, which have a whole section on Cisco/linksys (formerly Sipura) ATA's.
However, running your linksys (300x?) device as a trunk is very straightforward. Simply define a trunk and choose Sipura 3000 FXO from the dropdown carrier type. Make sure to fill out the mac address correctly when you create the trunk. If you are running in server-gateway mode and your 300x is in factory default mode then that's it. Just plug it into the network and off you go. You will need to create a route that uses it, but other than that you are done (if you also want to run a conventional phoe off it you will also need to create an extension with the same MAC address and a device type of Sipura 3000 FXS).
If you are running in server-only mode or the 300x isn't set to respond to DHCP option 66 then you'll need to set the 300x to point to your SARK/SAIL box for tftp - you will need our manual pages for that. They should be back on-line in a couple of days.
Several other folks on here run 300x devices as trunk/pstn-gateways so you may be able to get help in the meantime.
Kind Regards
Selintra
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Selintra, thanks for this kind answer on 01/01, and I'm really sorry for your dsl link shortage, it must be very irritating.
Unfortunately no, I'm not running a sipura spa3000 but a Cisco 186 dual fxs ata, sorry I didn't specified it before.
I'm afraid being far from a solution, when I call the ata (ext 5002 configured as generic sip) the gsm gateway answers, but cannot dial numbers over its dial tone, nor the ata will ever hangup once call ends.
There are some proprietary params to adapt the ata config for non standard hooked devices like SigTimer/FeatureTimer/OpFlags etc all features which I have no idea how to set and the doc isn't helpful at all, Cisco as always isn't for newbies. Maybe digging around a little more on asterisk/cisco forums I'll find some stuff about this but I really doubt.
Anyway, thanks again for your prompt support.
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Hi
Sorry - I assumed a linksys sipura device. We've not played with Cisco 186 - they are available here in UK, but not widely. The word "trunk" kind of threw me - I had assumed the ATA had to have at least one FXO port for gateway operations. The 186 has 2 FXS ports.
Most GSM gateway units we have played with, operate as true trunks (using FXS into FXO). Usually we connect them to an X100P or TDM400 (FXO) port, but occasionally we have run them through the FXO port of Sipura 3000 type analogue gateways. Either way, we simply send the call out over the trunk in the normal way and off it goes. Similarly an inbound call coming in off the GSM is handled like any other inbound Trunk call (SIP or Analogue).
From what I can see, you have your GSM device connected as an extension and I can't figure out how this would work because I don't see how DNID can be transmitted to the GSM device (other than possibly the extension number; in your case 5002)
Kind Regards
Selintra
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Most GSM gateway units we have played with, operate as true trunks (using FXS into FXO). Usually we connect them to an X100P or TDM400 (FXO) port, but occasionally we have run them through the FXO port of Sipura 3000 type analogue gateways. Either way, we simply send the call out over the trunk in the normal way and off it goes. Similarly an inbound call coming in off the GSM is handled like any other inbound Trunk call (SIP or Analogue).
that's a scenario i tried too, the gsm gateway has a dip switch to let you choose between fxo/fxs, but wasn't my preferred solution as my x100p actually hooks the box to plain pstn.
Anyway, using the gateway in fxs to fxo (x100p) outgoing calls are routed just like perfect (!), couldn't believe it. Problems come at the time to receive calls, ztmonitor shows no activity on TX channel while RX detects the incoming call, and will never pickup. Maybe a signalling issue? Calls just stop there, they won't reach asterisk.
thanks
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What kind of signalling is it using? These days loopstart (or Digium's equivalent; kewlstart) is pretty ubiquitous although I suppose there might be a few groundstart systems still out there.
Best
Selintra
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What kind of signalling is it using? These days loopstart (or Digium's equivalent; kewlstart) is pretty ubiquitous although I suppose there might be a few groundstart systems still out there.
that's a good question, i absolutely have no idea. btw i noticed, using an analog voltmeter, voltage stays around 130v while off hook and there doesn't seems to be any inversion when it starts ringing, but it peaks and goes beyond 250v.
tried changing signalling to fxsgs in zaptel.conf but that's what I get:
root@gateway:$ ztcfg -vv
Zaptel Configuration
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Channel map:
Channel 01: FXS Groundstart (Default) (Slaves: 01)
1 channels configured.
Changing signalling on channel 1 from FXS Kewlstart to FXS Groundstart
ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?
correct me if i'm wrong, X100P/TDM400 hardware do not provide support for groundstart signalling, but could this explain the troubles with this gsm brick? would i be able to place a call and hangup with ks/ls if the gsm was gs signalling?
noob question: is there any "hack" to let the x100p be less "sensitive" and pick up as soon as it detects activity?
sorry for anything stupid i could have stated here :wink: