On 10/30/2010 09:24 PM, Sebastian wrote:
>
> 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.
>
I think it does that here as well, but after a much longer delay than
asterisk sits around waiting - like close to a minute I think.
It definitely varies by carrier as well - Rogers here can't even get
their heads around delivering a txt message from an email to sms
gateway, let alone handle something like the above.
> 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: asterisk-users-boun...@lists.digium.com
>>> [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
>>> [s...@open-t.co.uk]
>>> Sent: Friday, October 29, 2010 5:50 AM
>>> To: asterisk-users@lists.digium.com
>>> 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: asterisk-users-boun...@lists.digium.com
>>>> [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
>>>> [s...@open-t.co.uk]
>>>> Sent: Thursday, October 28, 2010 5:27 PM
>>>> To: asterisk-users@lists.digium.com
>>>> 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: asterisk-users-boun...@lists.digium.com
>>>>> [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
>>>>> [s...@open-t.co.uk]
>>>>> Sent: Wednesday, October 27, 2010 5:00 PM
>>>>> To: asterisk-users@lists.digium.com
>>>>> 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: asterisk-users-boun...@lists.digium.com
>>>>>> [asterisk-users-boun...@lists.digium.com] On Behalf Of GBR Icasiano,
>>>>>> Ryan A. [raicasi...@globalbridgeresources.com]
>>>>>> 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: asterisk-users-boun...@lists.digium.com
>>>>>> [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
>>>>>> [s...@open-t.co.uk]
>>>>>> Sent: Tuesday, October 26, 2010 7:50 PM
>>>>>> To: asterisk-users@lists.digium.com
>>>>>> 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: ayodeleabej...@hotmail.com
>>>>>>> To: asterisk-users@lists.digium.com
>>>>>>> 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: jonathan....@gmail.com
>>>>>>> To: asterisk-users@lists.digium.com
>>>>>>> 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
>>>>>>> <ayodeleabej...@hotmail.com<mailto:ayodeleabej...@hotmail.com>>
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> _____________________________________________________________________
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>>>>>>> http://www.api-digital.com --
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>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Personal webpage - www.jonbaraq.eu<http://www.jonbaraq.eu>
>>>>>>>
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>>>>>>>
>>>>>>>
>>>>>> --
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>>>>>>
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>>>>>>
>>>>>>
>>>>> --
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>>>>>
>>>>>
>>>> --
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>>>>
>>>>
>>> --
>>> _____________________________________________________________________
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>>
>>
>
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