On 10/29/2010 04:40 AM, jon pounder wrote: > On 10/28/2010 11:18 PM, GBR Icasiano, Ryan A. wrote: > > Here is what I do today and it works fine: > > - asterisk/trixbox > - Dext/android phone > - Bell Canada cell provider > - call comes in, to an extension with voicemail > - rings a bunch of sip devices (real phones, and the android via > linphone if it happens to be near wifi and registered (set to only use > wifi not 3g to register) > - if not answered call is forwarded back out a pots line and dials the > cell number (cell is not subscribed to provider voicemail)
This is an advantage over my situation. Here (UK) - if you don't configure voicemail on your mobile - the mobile operator just plays a message along the lines "The phone number xxxx is not available right now. Please try again later" (or something similar). Which screws things up - as Asterisk can't tell that the mobile is not available. To Asterisk, that message is the same as somebody answering the line. Same in France and Spain - as far as I've seen. Sebastian > - still no answer that pots line is hung up and call drops back into the > original extension's vm. (I have not run into a problem with answer > detection, only that people don't stay on the line long enough for me to > answer on the second set of ringing, but if they are that impatient the > call was probably not important anyway) > > outgoing calls if registered I have a choice once I dial of linphone or > dialer to make the call. > > checking vm is just *98<ext> from linphone as the dialing app, or dial > in and navigate to vm. > > linphone is a little less polished gui but seems to work the best for me > to reliably register when it should. > (tried about 5 different sip clients) > > > > >> Hi, >> >> Thanks for your very informative response. This is really helpful. I >> wouldn't be pushing it though since it isn't possible as of now. >> >> Kudos! >> >> RYAN ICASIANO >> ________________________________________ >> From: [email protected] >> [[email protected]] On Behalf Of Sebastian >> [[email protected]] >> Sent: Friday, October 29, 2010 5:50 AM >> To: [email protected] >> Subject: Re: [asterisk-users] Mobile Phones and Asterisk >> >> Hi, >> >> On 10/28/2010 11:20 AM, GBR Icasiano, Ryan A. wrote: >> >>> 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. >>> >> Maybe others who know better will jump in - but I seriously doubt you >> will be able to do this. From my limited knowledge, I believe mobile >> phone networks use different signalling then regular terrestrial based >> providers. I don't really think that the engaged tone sent back by the >> mobile operator will be decoded correctly by Asterisk. >> >> Not to mention that, I don't what happens where you are - but in UK for >> example - you don't even get an engaged tone from a mobile phone. You >> just get either sent to the user's voice mail, or you are played a >> message from the mobile phone operator which essentially tells you that >> the user is engaged or unavailable. Operators in many other European >> countries do the same. So from the point of what you are trying to >> achieve - this is useless in Asterisk. >> >> I would have liked to do the same thing - as I have line divert in >> Asterisk to my mobile phone - and I would have liked for Asterisk to >> just skip along to my Asterisk voice mail when my mobile is either out >> of coverage, or when I'm in a conversation on it. But no such luck. I >> believe the mobile operators wouldn't like the idea anyway - as they get >> to charge you extra for playing all those messages or sending you to >> their voicemail. >> >> I believe in parts of the North American continent things are similar, >> but even worse. As the caller gets charged as soon as the mobile phone >> starts ringing - apparently simply the act of accessing the mobile >> operator's network is chargeable - never mind if you get to speak to >> anybody or not. >> >> Then again, maybe things are different where you are - and maybe there >> is a way to get Asterisk to recognise the busy tone from your mobile >> operator. Maybe somebody here will jump in with a suggestion. It seems >> that it has to do with "busy signalling" in Asterisk. A softphone I >> believe will accomplish this out of band - with some commands over SIP. >> While PSTN (normal phone lines) and mobiles I believe tend to signal >> this with inband tones (part of the sound coming down the line). >> >> You might also want to check your regional settings in Asterisk. >> >> >> Sebastian >> >> 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 >> >> > > -- _____________________________________________________________________ -- 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
