[asterisk-users] Portech MV-378 with Asterisk

Sat Jan 17 00:44:38 CST 2009

I want to dial out using the sim card.  What I did, I have used the SIP
channel ex:

Channel: SIP/thenumber at mv378

It shows the called is being made in the dialplan, but the number I have
entered does not dial, it just goes straight to the specified dialplan
extensions.

Then what I did, in the Lan to Mobile Table, I put * in url and the number I
wanted to dial in call num, then the call was made to that number using the
sim card properly.

I was wondering if I cannot supply the number to be dialed using an asterisk
call file, or do I have to put that number in the Lan to Mobile table.

Any help would be appreciated.

Thanks

On Sat, Jan 17, 2009 at 12:39 AM, Pascal Bruno <tipascal at gmail.com> wrote:

> Marco,
>
> The configs work fine for me.  I can receive calls with no problem.  Now,
> were you able to dial using the sim card?  I cant figure out how I can do it
> since asterisk doesnt have a channel to place call through the portech
> gateway.
>
>
>
>
>
> On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno <tipascal at gmail.com> wrote:
>
>> Thank you!, I will try that in a few hours and let you know what happens.
>>
>>
>>
>> On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini <marcotasto at libero.it>wrote:
>>
>>>
>>>
>>> Pascal Bruno wrote:
>>>
>>> Thanks for your reply!
>>>
>>> Can you tell me what you have in your Portech configuration settings
>>> (Mobile to Lan Settings; Sip Proxy settings etc...)  My sip.conf file is
>>> pretty similar to yours but still cant register.
>>>
>>>
>>>
>>> On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini <marcotasto at libero.it>wrote:
>>>
>>>> Emmanuel Pascal Bruno wrote:
>>>>
>>>>  Has anyone been able to configure portech's mv-378 gateway with
>>>> asterisk?
>>>>
>>>> I did the configuration as per the manual but it does not work.
>>>>
>>>> My server sees the portech gateway, but when the gateway is trying to
>>>> register to my server it fails.  It says peer is not suppose to register.
>>>>
>>>> The gateway and the asterisk box are on two different location (two
>>>> network, 2 differrent IP address).
>>>>
>>>> I would appreciate any kind of tutorial or advice on how to make it
>>>> work.
>>>>
>>>> Thanks
>>>>
>>>> ------------------------------
>>>>
>>>> _______________________________________________
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>>>>
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>>>>
>>>>
>>>> Hi,
>>>> I've an installation working with Portech MV-370. I'm supposing it's
>>>> quite similar to what you have. If it could be useful to you, this is my
>>>> sip.conf configuration file.
>>>>
>>>> [GSMGtw1]
>>>> type=friend
>>>> context=from-gsm
>>>> host=dynamic                    ; we have a DHCP assigned address
>>>> secret=reallyverysecret
>>>> nat=no                          ; there is not NAT between phone and
>>>> Asterisk
>>>> canreinvite=no
>>>> dtmfmode=INFO
>>>> insecure=invite                 ; required to overcome authentication
>>>> problems in incoming calls
>>>> call-limit=1                       ; permit only 1 outgoing call at a
>>>> time
>>>> disallow=all
>>>> allow=ulaw
>>>> allow=alaw
>>>> allow=gsm
>>>> qualify=500
>>>>
>>>> I remember that I've found a bug on the firmware that prevents to the
>>>> unit to register correctly on my asterisk box unless I'm using the raw IP
>>>> address instead of the name of the asterisk box. I remember something wrong
>>>> in cryptography chiper/dechiper based on realm... So, if you have problems,
>>>> let's try to specify the asterisk raw IP address in the Portech.
>>>>
>>>> Best regards,
>>>> Marco Signorini.
>>>>
>>>>
>>>>
>>> Hi,
>>>
>>> I don't know if the problem could be in the Mobile to Lan or Lan to
>>> Mobile settings because these  settings are related on how calls coming
>>> from/to mobile are routed.  I didn't use the Portech routing features at all
>>> because I need a simple GSM gateway to/from the asterisk box.
>>> For this reason:
>>> 1. The only rule I've on Mobile to Lan is CID=*; URL=mob at 192.168.0.5where "mob" is the extension I've generated in the asterisk box under the
>>> context where the Portech operates;
>>> 2. The only rule I've on Lan to Mobile is URL=*; Call Num=#
>>>
>>> I think the most relevant parameters for your problem are under the
>>> "Service Domain" menu option (assuming that the firmware you have is similar
>>> to what I've). On this menu I've compiled the 1st Realm (as I've only one
>>> account) like that:
>>>
>>> UserName: GSMGtw1
>>> RegisterName: GSMGtw1
>>> RegisterPassword: reallyverysecret
>>> Domain Server: 192.168.0.5
>>> Proxy Server: 192.168.0.5
>>>
>>> Pay attention that, having specified the Domain Server with the raw IP
>>> address, asterisk needs to be able to authenticate peers associated to that.
>>> For this reason I've set:
>>>
>>> domain=192.168.0.5
>>>
>>> on sip.conf [general] section (remember to issue a sip reload from
>>> asterisk cli).
>>>
>>> Hope this helps!
>>>
>>>
>>> Best regards.
>>> Marco Signorini
>>>
>>>
>>>
>>> ========================
>>> Marco Signorini
>>> INGEGNI Tech S.r.l.
>>> http://www.ingegnitech.com
>>>
>>> _______________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>
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