I am having quite difficulties configuring the Portech MV-372 as a trunk to Trixbox.
Does someone has any complete (step by step) solution how to configure it?
Thank you
 
Portech MV-372 trunk
Submitted by kpevoip on Sat, 06/14/2008 - 6:31am.
Portech MV-372 How-To
Thu, 06/19/2008 - 2:54am
Since no one helped me setup Portech MV372 in Trixbox/Asterisk here is how i did it:
Configuring Portech:
Mobile to Lan:
CID: *
URL: 100 (put your extension or ringgroup)
Lan to Mobile:
URL: * (or you can put your asterisk IP)
Call Num: # (this will pass the number dialed from IP phone, so no 2stage dialing)
SIP Settings:
Realm 1
Display name: 1001
User Name: 1001
Register name: 1001
Register password: 1001 (you change it)
Domain Server: asteriskIP
Proxy Server: asteriskIP
the same applies if you put Sim 2 so the configuration goes to Mobile2 except you change the username to like 1002
Now we configure the Asterisk (with freepbx)
Setup SIP Trunk
Outbound Caller ID: put you mobile number
Maximum Channels: 1
Trunk name: SIM1
Peer Details:
host=192.168.x.x (put your Portech IP)
type=peer
Incoming settings
USER Context: 1001
User Details:
type=friend
secret=1001
username=1001
qualify=yes
nat=yes
canreinvite=no
context=from-internal
host=192.168.x.x (your Portech IP)
No Registration needed
Submit changes (dont forget to apply settings)
You setup another SIP Trunk for Mobile 2 (sim2), settings are the same as we configured SIM1 except in Peer details and User details you have to add a line:
port=5062
as portech's sim's are configured to different port each.
And then the easy part to create Outbound Route:
Type any name you want, put any dial pattern you want i have set mine so only mobile number are dialed from this route like: 049XXXXXX (operators code) but you can change it as u need, and you add your trunks to Trunk Sequence.
 
Thanks
Thu, 06/19/2008 - 1:31pm
Thanks for contributing back to the community. Maybe nobody else with a Portech saw this in the time between the OP and when you gesolved it.
Glad you got it working.
Do you mind if I add this information to the Wiki so it can be easily found?
Thanks....Scott
--
Scott
aka "Skyking"
 
Of course i dont mind just
Fri, 06/20/2008 - 1:35am
Of course i dont mind just go ahead :D
Thanks,
Ilir
 
not working
Wed, 07/30/2008 - 7:16am
Hy!
This configuratin is not working to me. I make this configuration step by step, but in the mv-372's menu, SIP Settings --> Service domain -->Realm 1 :
Status: Not Registered
why? Help my please!
in the asterisk log:
[Jul 30 16:15:13] ERROR[2276] chan_sip.c: Peer '30' is trying to register, but not configured as host=dynamic
[Jul 30 16:15:13] NOTICE[2276] chan_sip.c: Registration from '"SIM1"
 
How to setup Trixbox to receive CID from MV-372?
Fri, 12/05/2008 - 4:26am
...and display on IP phone?
 
Sorry for coming late with
Fri, 12/05/2008 - 7:57am
Sorry for coming late with the reply.
I have come up with another configuration that fits me better and receive CID properly without much hassle
---------
Portech config
Mobile to lan:
Item: 0
CID: *
URL: 192.168.x.x (your asterisk ip)
Lan to mobile:
Item: 0
URL: *
Call num: #
Mobile > Settings:
You adjust your gain as you need
Mobile 1:
SIP From: Tel/Tel (No reg)
CLID Presentation: Invocation
LAN Answer Mode: Income
Same settings for Mobile 2
SIP settings > Service Domain:
You only need to write Domain Server and Proxy Server to your asterisk ip address, leave everything else blank. Same for Mobile 2
Port Settings:
Mobile1 > Sip port: 5060
Mobile2 > Sip port: 5062
Don't forget to Save changes and then reboot
---------
Asterisk trunks:
Outbound called id: type you mobile number
Maximum channels: 1
Outgoing settings:
Trunk name: SIM1
Peer details:
host=youportechip (i havent tried dynamic i believe it will work)
type=peer
port=5060
User Context: type you mobile number
Create another trunk for SIM2 with the same settings except
Trunk name: SIM2
Peer details:
host=youportechip
type=peer
port=5062 (important)
and User context: your second mobile number
Now i assume that you have already Inbound Route with any DID/any CID and allowed Anonymous inbound sip calls (under General Settings Allow Anonymous Inbound SIP Calls)
Then you could check in Peers if Sim1 and SIM2 are registered there. (sip show peers)
And that should be it. Let me know if this works for you.
 
it works!
Sat, 12/06/2008 - 2:22am
"Mobile 1:
SIP From: Tel/Tel (No reg)
CLID Presentation: Invocation
LAN Answer Mode: Income"
This is essential.
Thanks.
 
Tel/Tel does not allow to call external number...
Thu, 02/12/2009 - 3:20am
When use settings "SIP From: Tel/Tel (No reg)" I can't dial external number through Portech. Works only calling extension. How to setup Portech / PBX to call both? Of course authorized users can call both and unauthorized users can call only extensions and IVR numbers.
Thanks
Igor
 
Igor im not sure if i
Thu, 02/12/2009 - 8:17am
Igor im not sure if i understood your problem very well.
As i understand you want to dail numbers from inside asterisk to outside world, if that is the case then you MUST have trunks and outbound routes to make that work.
Or
You want that when someone calls from outside world to your portech(asterisk) be able to dial extensions and use trunks also to make outbound calls, if this is the case then the best scenario would be for you to have IVR so guest users would dial only internal extensions and setup DISA so authorized users can call the IVR and choose DISA and after that they are able to use any trunk that you will make available for them.
Or
I am not understanding nothing what you want to achieve :P
 
"You want that when someone calls from outside..."
Fri, 02/13/2009 - 4:11am
This is your good answer :-). Thanks. With DISA I have some progress, but not enough.
How to setup Portech "Mobile To LAN" or/and PBX "Incoming Route" that authorized mobile user can dial any number and unauthorized mobile user can dial only extensions and IVR? When authorized user call Portech GSM, hear "dial tone" and can dial any number. When unauthorized mobile call Portech GSM, hear "IVR" and can dial only IVR numbers and extensions.
Or how to set Portech "Mobile To LAN" for authorized CID to hear "dial tone" and unauthorized CID hear IVR?
I hope that is now much more clearly what I want.
I know to make this but must use "SIP From: Tel/User (standard)" and then my IP phone display only extension number of Portech (no CID of phone that call Portech).
In manual of Portech "22.How to setup Asterisk to receive Caller ID from MV-370/MV-372.................. 54" is very vaguely.
 
Okay then you have 2
Fri, 02/13/2009 - 7:25am
Okay then you have 2 choices, more secure, less secure:
1. More Secure.
You make inbound route so any CID, any DID have destination set to IVR and then make inbound routes for each number you want to authorize to use DISA and set destination to those authorized numbers to DISA. Ill give you example so you can understand more easily:
AnyCID/AnyDID with destination to IVR
then numbers:
CID: 079545321 to dest: DISA
CID: 029981156 to dest: DISA
in this case only when users call from the numbers you put in inbound route get access to DISA and all other number are transferred to IVR.
2. Less secure.
You set any cid any did to go to IVR and in the IVR you add/increase option with destionation to DISA so everyone calls, they get IVR and only who knows the extension/number will press and they will be transferred to DISA.
IMO i would go with 1st option since only numbers which you specify can use DISA, using option 2 will be much more risky since all those authorized members that you have could give the extension to DISA to anyone and get access to all your outbound routes/trunks.
In both cases i strongly suggest you use PIN code just for added security.
Thanks
 
Incoming to 7777
Fri, 08/07/2009 - 5:09am
Hello,
I looked everywhere but could not get incoming calls to work to go to the IVR.
Have just worked it out,
setup the hardware (MV-378) as per the book except Mobile to LAN item=0 CID=* URL=7777. Also setup Trixbox as per the (very basic) insturctions that come with the unit, except also added.
host=192.168.1.100
port=5060
type=peer
into incoming settings in trunk.
The change that got it to work (I could get it to go to an extention but not the IVR) was to edit the extensions_additional.conf file and copy this line,
exten => 7777,1,Goto(from-pstn,s,1) - which was under [ext-test]
and paste it under
[ext-local]
include => ext-local-custom
Also in settings in trixbox you need to allow anon SIP calls.
Hope this helps someone as no searching helped me.
Thanks,
Callum Henderson
Dial It Limited / Auckland It Limited (New Zealand)
 
gsm trunk help
Fri, 08/07/2009 - 5:14pm
Hi, I have trixbox 2.2 working with LAN extensions, calling out to PSTN through my voip provider ok.
I have a GSM gateway suncomm sc386 just to make LAN to mobil calls. i think is similar to the one you are dealing here.
I got my GSM GW registered in trixbox but when i dial out i get this:
-- Executing Set("SIP/103-0921cde0", "USEROUTCID=") in new stack
-- Executing Set("SIP/103-0921cde0", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/103-0921cde0", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/103-0921cde0", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/103-0921cde0", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/103-0921cde0", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/103-0921cde0", "CallerID set to "103" ") in new stack
-- Executing GotoIf("SIP/103-0921cde0", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,16)
-- Executing DeadAGI("SIP/103-0921cde0", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/103-0921cde0", "OUTNUM=99155858") in new stack
-- Executing Set("SIP/103-0921cde0", "custom=ZAP/g0") in new stack
-- Executing GotoIf("SIP/103-0921cde0", "0?customtrunk") in new stack
-- Executing Dial("SIP/103-0921cde0", "ZAP/g0/99155858|300|") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/103-0921cde0", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/103-0921cde0", "Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
-- Executing Macro("SIP/103-0921cde0", "outisbusy|") in new stack
-- Executing Playback("SIP/103-0921cde0", "all-circuits-busy-now|noanswer") in new stack
-- Playing 'all-circuits-busy-now' (language 'en')
== Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/103-0921cde0' in macro 'outisbusy'
== Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/103-0921cde0'
can anyone help me?
 
Portech MV-372 trunk
Mon, 08/10/2009 - 1:21pm
Hello, I am new here in the forum, but I am with the following problem. I have a MV-372 with Asterisk running almost perfect, with my dial extensions by MV-372 normally, but when I try to call the MV-372, the asterisk serves with a recording (ss-noservice) that the number is not in service, etc ... Detail, this occurs in Trixbox v.2.2.12 in Trixbox 2.6.2.2 in extensions_custom.conf just add the lines below:
[from-sip-external]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,4,Goto(ext-queues,10,1)
Somebody help me?
Tanks.
 
Hello. I've managed to setup
Mon, 11/16/2009 - 7:24am
Hello. I've managed to setup both inbound and outbount calls over MV-372, and works ok, but i minor problem.
When I call from outside to MV-372 -->>asterisk i get a hangup delay. I hangup, but asterisk keeps ringing for about 3-5 seconds.
Anyone with a clue?
Thank you.
 
latency
Tue, 11/17/2009 - 5:41am
702/702 172.16.0.117 D N A 5062 OK (10 ms)
701/701 172.16.0.68 D N A 5060 OK (10 ms)
501 (Unspecified) D N A 0 UNKNOWN
1001/1001 172.16.0.65 D 5060 OK (200 ms)
Latency is to big for internal LAN. Phones have 10ms, MV-327 has 200ms (connected on same switch).
Anyone?
Thank you
 
Did some sip debug...it just
Tue, 11/17/2009 - 6:25am
Did some sip debug...it just takes about 5 seconds from actual termination of the call to send Hangup to asterisk.
But calls get connected over MV-372 to asterisk in about 3-4sec, that seems pretty ok (it must first connect to GSM carrier).
Do i have false settings on MV-372? Or a faulty device?
Ping to MV-372 is ok:
ping 172.16.0.65
PING 172.16.0.65 (172.16.0.65) 56(84) bytes of data.
64 bytes from 172.16.0.65: icmp_seq=1 ttl=255 time=0.518 ms
64 bytes from 172.16.0.65: icmp_seq=2 ttl=255 time=0.485 ms
64 bytes from 172.16.0.65: icmp_seq=3 ttl=255 time=0.497 ms
64 bytes from 172.16.0.65: icmp_seq=4 ttl=255 time=0.499 ms
--- 172.16.0.65 ping statistics ---
4 packets transmitted, 4 received, 0% packet loss, time 3003ms
rtt min/avg/max/mdev = 0.485/0.499/0.518/0.029 ms
Please help me...
Thank you.
 
How do I get to the webpage of the MV378 device
Tue, 02/02/2010 - 8:46am
I connected the MV378 to a switch but couldnt access the webpage settings. How do I get the default IP to use to access the webpage settings?
 
How do I get to the webpage of the MV378 device
Wed, 03/24/2010 - 12:16am
I don't know about through a switch but I also had trouble bringing up the webpage until I connected a cross over cable into the WAN port of the MV372.
 
Member Since:
2007-09-27