Hi, I can actually place a successful call using that configuration. The telco i'm currently working requires the prefix.
What I'm trying to do is to capture the status of the mobile phone, if it is currently engaged in a call or not. I achieved this successfully by emulating it via a softphone, when I call a softphone and it is currently engaged in a call, asterisk returns BUSY in DIALSTATUS and will automatically fallback to the next step in the dialplan. But this is not the case when applying it to the mobile phone. When the target phone is currently engaged in a call, and I called the mobile phone, I can hear a "busy tone"(which is alright, since the target phone is actually busy), but it will wait until it timed out as defined in the DIAL cmd, and the "var DIALSTATUS" returns NOANSWER, instead of BUSY, as if the mobile phone is available and it was not answered at all. It may also have to do on how the tones are being handled, or it can also be that the mobile phone and the media gateway are the one talking to each other, and asterisk cannot get the status of the phone itself. regards, RYAN ICASIANO ________________________________________ From: [email protected] [[email protected]] On Behalf Of Sebastian [[email protected]] Sent: Thursday, October 28, 2010 5:27 PM To: [email protected] Subject: Re: [asterisk-users] Mobile Phones and Asterisk Hi, On 10/28/2010 01:06 AM, GBR Icasiano, Ryan A. wrote: > Hi, > > Thanks for your reply. I'm calling a normal phone using the DIAL cmd. Here is > my sample dial command: > > exten =>s,4,Dial(SIP/xxx${extensi...@media_gateway,10,t) > > but when I use: > > exten =>s,5,NoOp(SIP/xxx${extensi...@media_gateway has state ${DIALSTATUS}) I'm not quite sure what you are trying to do. So you called the phone for 10 seconds, the phone didn't answer - and the variable "DIALSTATUS" told you exactly that. Is the problem the fact that the line is not ringing out? Is that what is wrong? And why do you have some "xxx" in front of ${extension}? You shouldn't need them. Just pass ${extension} - which is the number you dialled on the phone. Sebastian > > I hear a busy tone, after the 10 sec. timeout it returns NOANSWER, as defined > in my DIAL func. > > I also tried getting the DEVICE_STATE > > exten =>s,3,NoOp(SIP/xxx${extensi...@media_gateway has state > ${DEVICE_STATE(SIP/xxx${extensi...@media_gateway)}) > > and same thing happens as stated on the scenario below. > > Thanks again! > > regards, > > RYAN ICASIANO > ________________________________________ > From: [email protected] > [[email protected]] On Behalf Of Sebastian > [[email protected]] > Sent: Wednesday, October 27, 2010 5:00 PM > To: [email protected] > Subject: Re: [asterisk-users] Mobile Phones and Asterisk > > 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 > -- _____________________________________________________________________ -- 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
