Hi, On 10/27/2010 05:55 AM, GBR Icasiano, Ryan A. wrote: > anyone??? > > regards, > > RYAN ICASIANO > > Hi, > > I changed my sip.conf and added call-limit. At first I thought it works ok, > since i tried calling a cellphone that is currently busy(phone answers 1st > softphone, then another softphone calls the same number, it now returns > INUSE). But then, i tried calling a different number while the first phone is > busy, but it returns INUSE. It seems that the status being returned was from > the peer itself(both phones uses the same peer) and not from the > device(mobile phone) which i believe is more logical. > > I also tried using DIALSTATUS(which of course you need to DIAL first), but > then I only hear a busy tone and the dialstatus will return a noanswer. Do I > have to configure it first in order to capture the busy status of a device? > Have you done something similar to this? > > I'm using ver. 1.6. Thanks in advance.
I'm not sure I understand your setup. Are you using SIP for trunking, or for extensions? Are you calling a normal mobile phone, or a SIP client on a mobile phone? Sebastian > > regards, > > RYAN ICASIANO > ________________________________________ > From: [email protected] > [[email protected]] On Behalf Of GBR Icasiano, Ryan A. > [[email protected]] > Sent: Tuesday, October 26, 2010 10:41 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Mobile Phones and Asterisk > > Hi, > > Is the dev_state can also be used to track a mobile phone's status via SIP? > I tried it on several phones(nokia, samsung) but it returns NOANSWER but i > can hear a beep beep beep sound indicating that it is currently busy. > > regards, > > RYAN ICASIANO > > __________________________ > From: [email protected] > [[email protected]] On Behalf Of Sebastian > [[email protected]] > Sent: Tuesday, October 26, 2010 7:50 PM > To: [email protected] > Subject: Re: [asterisk-users] Mobile Phones and Asterisk > > On 10/26/2010 12:30 PM, ayodele abejide wrote: >> Hello Jonathan, >> >> The solution would work only if the ISP has one public address, but in >> my solution they have a pool of public address, any other possible solution? > > With dynamic dns, you either install a piece of software on your server > (dynamic dns client) or you use the facility provided by your router > (some firewall/router/access point combo's have them). This software > updates automatically the record with dyndns every time your IP address > changes. > > Sebastian > > >> >> ABEJIDE, Ayodele A. (CCNA) >> +2348039269311 >> >> >> >> >> ------------------------------------------------------------------------ >> From: [email protected] >> To: [email protected] >> Date: Tue, 26 Oct 2010 11:01:09 +0000 >> Subject: Re: [asterisk-users] Mobile Phones and Asterisk >> >> thanks i would check it up >> >> ABEJIDE, Ayodele A. (CCNA) >> +2348039269311 >> >> >> >> >> ------------------------------------------------------------------------ >> Date: Tue, 26 Oct 2010 12:52:30 +0200 >> From: [email protected] >> To: [email protected] >> Subject: Re: [asterisk-users] Mobile Phones and Asterisk >> >> Try http://www.dyndns.com/ that should solve your problem with dynamic IPs. >> >> Regards, >> Jonathan >> >> On Tue, Oct 26, 2010 at 12:40 PM, ayodele abejide >> <[email protected]<mailto:[email protected]>> wrote: >> >> Dear Asterisk-Users, >> >> I have this Asterisk Box I run in my house, I need to terminate and >> originate remote calls through the box via internet (SIP), the >> problem is in Nigeria most ISPs would not provide you with Public >> Addresses, all they provide is dynamic Natted addresses which change >> each time one connects, I have thought of all possible solutions and >> cannot come up with one, can anyone please help. >> >> Thanks in anticipation >> >> ABEJIDE, Ayodele A. (CCNA) >> +2348039269311 >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> >> -- >> Personal webpage - www.jonbaraq.eu<http://www.jonbaraq.eu> >> >> -- _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE >> or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> -- _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE >> or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
