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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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Setup MV-370 GSM Gateway with Asterisk

The GSM gateway MV-370 is manufactured by http://www.portech.com.tw/.
It's main interest in comparison with other gateways is that it is GSM to SIP (not FXS), so voice quality is very good.
Another point of interest is the price (around $150 on ebay).
With that gateway properly configured, you are able to receive calls from GSM to Asterisk (including DISA) and to give calls from Asterisk to GSM network.

Usage

A typical usage of such a gateway is to be able to give a call with your normal mobile to any destination at voip cost :
Your mobile <----gsm network----> MV-370 <--lan--> Asterisk <--internet--> VOIP provider <--whatever--> landline

To do such a call, you just call your MV-370 number (it has its own simcard), then you get an invitation tone, then you dial the number which is handled by Asterisk.
If you have some special deals with your mobile operator, like free special number, you can call your MV-370 for free.
You can then call all around the world from your mobile at voip cost :-)

MV-370 Configuration

The first thing to do with the MV-370 is to flash the latest firmware.
Any firmware dated before 2006/09/26 is totally unstable, making the box very difficult to parameter and to work with.
Once you've configured everything in the box, one good advice is to unplug the power and to restart it.
By this way you sould have all the parameters taken into account.

To have the MV-370 to work with Asterisk, you need first to configure the box.
Here are some screen shots showing all the important parameters.
You have to note that in all the configuration process, the MV-370 is considered as extension '103' of the IPBX.
In Bold are the parameters depending on your installation




Here the '#' is important to avoid the two stage dialing when you give a call from Asterisk to GSM.


The mobile number you give in that page are the authorised mobile which can call GSM to Asterisk.
These mobile number must be defined as your GSM provider displays the number.
If you don't know how it is displayed, just give a call to the box and check the number given in the 'Incoming Mob' field of the 'Mobile Status' page.
Any number which is not in that list won't have acces to the LAN side, so to Asterisk.
If you want to allow any number, just set '*' in that field ... but beware of the bill ;-)


Once Asterisk configuration is made, you should get 'Registered' on the Realm1.


It is very important to use only ulaw or alaw as all DTMF is inband.
So if you want to be able to do some DISA when you call from GSM to Asterisk, it has to be one of these 2 codecs.


These settings seem to be ok, just adjust ...

Antenna position

Another important thing is to properly place the provided antenna.
If your gsm reception is good, you should get around 18 or 19 as Signal Quality in the "Mobile Status" page.
With that level of signal quality, your audio quality will be very good.
On the other end, I've experienced that with a signal quality down to 11, audio becomes very jerky.
So, maximum signal quality = maximum audio quality.

Asterisk configuration

Once the MV-370 is set, you have to configure Asterisk.
On that side, you have to setup files as follow :


sip.conf

; GSM VOIP Gateway MV-370
[103]
type=friend
username=103
fromuser=103
regexten=103 ; When they register, create extension 401
secret=xxxxxxx ; Asterisk extension password
context=gateway ; Incoming calls context
dtmfmode=inband ; Very important for DISA to work
call-limit=1 ; Limit to 1 call max
callerid=GSM Gateway <103>
host=dynamic
nat=no ; Gateway is not behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
insecure=very
qualify=yes
disallow=all
allow=ulaw ; prefered codec for DTMF detection
allow=alaw


extensions.conf

; ******* GSM Gateway incoming calls **********
[gateway]
exten => _103,1,Answer()
exten => 103,2,Set(TIMEOUT(digit)=3) ; give enough time to do second stage dialing
exten => 103,3,Set(TIMEOUT(response)=5)
exten => _103,4,DISA(no-password|outgoing) ; here 'outgoing' is the normal context to deal with the dial plan

[outgoing]
...
; example of LAN to GSM call
; call the MV-370 sim card mail box thru GSM
exten => _888,1,SetCallerID("xxxxxxxxxx")
exten => _888,2,Dial(SIP/${EXTEN}@103,60,r)
exten => _888,3,Hangup()



Here you can see that all my dialplan is defined in the [outgoing] section.


As a conclusion, MV-370 is a very nice toy to save lot of GSM bills, but it is a hell to setup as the documentation is really, really poor.

That's it ... hope it works for you :-)



Connection to TRIXBOX 2.0
-----------------------
Well - i have got it working after a while. The problem was that the MV-370 did forward calls from gsm to trixbox, but not the other way. I changed, in the lan to mobile section of the mv-370, the "#" to "#d0a0" to cut of the trailing "0" and the add it again. For me this works since all numbers start like this.. I added a trunk, configuration only in "peer Details" like this :
allow=ulaw&alaw
canreinvite=no
context=from-pstn
disallow=all
dtmfmode=inband
fromuser=61
host=dynamic
insecure=very
nat=no
qualify=yes
secret=61
type=friend
username=61


Created by nicolas bernaerts, Last modification by gsmd on Fri 01 of Jun, 2007 [11:30 UTC]

Comments Filter

MV-372: using 2 SIP channels parallel

by gator on Friday 27 of July, 2007 [10:03:12 UTC]
If 2 user want to make a SIP phone call at the same time
you have to change the extensions.conf manually without the Web interface.
I setup two SIP trunks (channel1 and channel2). One is using port 5060 and
number 2 uses port 5062. In the OutboundRoutes you specify the two trunks in sequence.
Pick an editor and open /etc/asterisk/extensions.conf.
Goto the macro-dialout-trunk section and to the line
exten => s-BUSY,2,Busy(20)
Comment out this line:
; exten => s-BUSY,2,Busy(20)
Save file and refresh configuration in FreePBX.
Now Trixbox tries to take the 2nd channel in MV-372 if the first is in use.

Have fun with this.


MV-372: howto use 2 channels with Trixbox 2.2

by gator on Wednesday 25 of July, 2007 [21:29:54 UTC]
How do I have to setup Trixbox to use the 2 channels simultaneously?
Right now I use 1-stage dialing.
There is 1 outbound rule and I configured 2 GSM trunks with port=5060 and port=5062.
Port setting in GSM trunk seems to be ignored. Where do I have to set the two different ports?
Any help is appreciated. Thank you.

Firmware Update

by darylp on Friday 29 of June, 2007 [22:31:18 UTC]
Using new firmware, unit works for ~7 days before stops answering incoming calls.
Using 2 stage dialing, rather than 1 stage. Maybe 1 stage would work O.K.

Rebooting the MV-372 using the web-interface allows incoming calls to be answered again.

MV-372 works O.K. with Firefox 1.5.0.12 now.

TECHNICAL APPENDIX:
Model Type: VoIP2 GSM:900/1800MHz
Firmware Version: Mon May 28 09:24:27 2007
Codec Version: Mon Jul 24 10:55:05 2006

Firmware update

by mon888 on Tuesday 12 of June, 2007 [08:03:03 UTC]
Just an update for you guys. Portech sent me a firmware with ff details almost a month ago:

Firmware Version: Mon May 7 13:30:57 2007.
Codec Version: Mon Jul 24 10:55:05 2006.

This firmware works flawlessly for single stage dialing to and fro the MV-372 connected to my Trixbox machine.

DTMF set to rfc2833 also works which was not the case under older firmware.

sending SMS

by arun on Monday 04 of June, 2007 [09:49:51 UTC]
Can anyone tell me how to send SMS using my MV 370

Re: MV-372 Firmware

by darylp on Sunday 27 of May, 2007 [07:39:44 UTC]
Hi Richmond,
I have had the MV-372 stop answering incoming calls.
It also is not responding to pings.

Power-cycling the unit solves the problem.

I've installed a power-timer that power-cycles it every Monday at 0:00...

TECHNICAL APPENDIX:
MV-372 VoIP2 GSM:900/1800MHz, Firmware Wed Apr 11 03:54:33 2007, Codec Version: Mon Jul 24 10:55:05 2006

MV-372 Firmware

by mon888 on Saturday 05 of May, 2007 [03:03:17 UTC]
Single stage dialing now works under the the April 23 2007 firmware. However, the unit would suddenly stop answering incoming gsm calls in less than a day of operation. A reboot from the admin page will fix this but the problem will pop up again in a matter of hours.

Re: MV-372 for Trixbox 2.0

by darylp on Thursday 22 of March, 2007 [22:37:00 UTC]
1 stage dialling may not work with the MV-372:

Fang from Portech wrote to me:
"
MV-372 has most functions that MV-370 has,
except when M to L, 1 stage dialing, or we call it "assign mode" is not available.
While in Mobile to Lan routing, an assigned IP is not acceptable.
You could set the routing for 2 stage dialing:
after a dial tone, press the Lan IP, ext. SIP number or even the destination phone number.
Note: for phone number function, related routings shall be setted in SIP server.
"

Your problem seems to be LAN -> Mobile rather than Mobile -> LAN.
However, the above information may help.

It's Amazing ! TRue Value for Money.

by Mukul Jain on Sunday 18 of March, 2007 [11:22:13 UTC]
I've got my MV-370 from http://www.e-xcessories.com/estore/portech-mv370-voip-gateway-asterisk-trixbox-p-2628.html , The voip-info site was amazing to help me to make this working. This is working very well with my TrixBox.

E-xcessories did a good job of delivering this quickly to me, rest was taken care by Voip-info.org , The incoming calls are now coming with Caller-ID, outgoing calls are very very clear. the GSM gateway has been installed in India :) at my home country and I am able to save considerable savings now instead of making IDD calls into India. Also my relatives / colleuges are able to call me using a local india no. Whole setup was very easy, fast and took around 2 days for me to install / tweak and make it well working.

Thanks to this site for the wonderful resource. I highly recommend MV-370.

Re: mv-372 testing results

by Mike Rezvani on Saturday 17 of March, 2007 [19:00:12 UTC]
gool, Firmware Version: Tue Dec 5 13:38:59 2006 is what I downloaded from portech site, but yours is showing January. Do you have Model Name: VoIP GSM:900/1800/1900MHz or Quad? Or did I miss something!

Thanks

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