I want to dial out using the sim card. What I did, I have used the SIP channel ex:
Channel: SIP/thenum...@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 <[email protected]> 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 <[email protected]> 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 >> <[email protected]>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 >>> <[email protected]>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 >>>> >>>> ------------------------------ >>>> >>>> _______________________________________________ >>>> -- 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 >>>> >>>> >>>> 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=*; [email protected] >>> "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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