[asterisk-users] Portech MV-378 with Asterisk
Pascal Bruno
tipascal at gmail.com
Fri Jan 16 23:39:04 CST 2009
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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>>>
>>> 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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