Hi, one way to solve the problem with Mailbox or that Message that get's played when busy/not available (same happens with Orange in Austria and other providers) you can implement something similar to what Elastix/FreePBX has. "Confirm call" - this will let the caller think it's still ringing while you will have to confirm the call after picking it up by dialing 1#. I use this when traveling through more then one country. Since I don't want to always change the GSM Number that is dialed when not in the office I simply send the call to ALL GSM Numbers with this option activated. Whichever I answer and press 1# gets the call.
Cris On 2 Nov, 2010, at 04:30 , GBR Icasiano, Ryan A. wrote: > Yup, that's exactly what is happening. If there is only a way to override the > response(busy tone) by a ringing tone from asterisk, then the caller will not > hang up since after the "busy" status interpreted by asterisk as NOANSWER, > there will be a fallback which it will either transfer to another extension > or go directly to the callee's voicemail. > > regards, > > RYAN ICASIANO > ________________________________________ > From: [email protected] > [[email protected]] On Behalf Of Sebastian > [[email protected]] > Sent: Sunday, October 31, 2010 9:24 AM > To: [email protected] > Subject: Re: [asterisk-users] Mobile Phones and Asterisk > > 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 > > > This message has been scanned for malware by Websense. www.websense.com > > -- > _____________________________________________________________________ > -- 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
