Mobile Phones and Asterisk

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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

18 thoughts on - Mobile Phones and Asterisk

  • 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.

    regards,

    RYAN ICASIANO
    ________________________________________
    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A. [raicasiano@globalbridgeresources.com]
    Sent: Tuesday, October 26, 2010 10:41 AM

    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

  • 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

  • thanks i would check it up

    ABEJIDE, Ayodele A. (CCNA)
    +2348039269311

    href=”mailto:jonathan.gsc@gmail.com”>jonathan.gsc@gmail.com
    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com

    Try http://www.dyndns.com/ that should solve your problem with dynamic IPs.

    Regards,
    Jonathan

    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

  • 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?

    ABEJIDE, Ayodele A. (CCNA)
    +2348039269311

    href=”mailto:ayodeleabejide@hotmail.com”>ayodeleabejide@hotmail.com
    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com

    thanks i would check it up

    ABEJIDE, Ayodele A. (CCNA)
    +2348039269311

    href=”mailto:jonathan.gsc@gmail.com”>jonathan.gsc@gmail.com
    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com

    Try http://www.dyndns.com/ that should solve your problem with dynamic IPs.

    Regards,
    Jonathan

    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

  • 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

    href=”mailto:ayodeleabejide@hotmail.com”>ayodeleabejide@hotmail.com
    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com
    href=”mailto:jonathan.gsc@gmail.com”>jonathan.gsc@gmail.com
    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com

  • 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.

    regards,

    RYAN ICASIANO
    ________________________________________
    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A. [raicasiano@globalbridgeresources.com]
    Sent: Tuesday, October 26, 2010 10:41 AM

    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

    __________________________
    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Sebastian [shop@open-t.co.uk]
    Sent: Tuesday, October 26, 2010 7:50 PM
    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com

    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

    href=”mailto:ayodeleabejide@hotmail.com”>ayodeleabejide@hotmail.com
    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com
    href=”mailto:jonathan.gsc@gmail.com”>jonathan.gsc@gmail.com
    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com

  • Hi,

    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

    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A. [raicasiano@globalbridgeresources.com]
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  • 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${extension}@media_gateway,10,t)

    but when I use:

    exten =>s,5,NoOp(SIP/xxx${extension}@media_gateway has state ${DIALSTATUS})

    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${extension}@media_gateway has state ${DEVICE_STATE(SIP/xxx${extension}@media_gateway)})

    and same thing happens as stated on the scenario below.

    Thanks again!

    regards,

    RYAN ICASIANO
    ________________________________________
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    Sent: Wednesday, October 27, 2010 5:00 PM
    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com

    Hi,

    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

    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A. [raicasiano@globalbridgeresources.com]
    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Sebastian [shop@open-t.co.uk]
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  • Hi,

    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

    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Sebastian [shop@open-t.co.uk]
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  • 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
    ________________________________________
    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Sebastian [shop@open-t.co.uk]
    Sent: Thursday, October 28, 2010 5:27 PM
    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com

    Hi,

    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

    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Sebastian [shop@open-t.co.uk]
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  • Hi,

    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.
    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Sebastian [shop@open-t.co.uk]
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  • 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
    ________________________________________
    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Sebastian [shop@open-t.co.uk]
    Sent: Friday, October 29, 2010 5:50 AM
    href=”mailto:asterisk-users@lists.digium.com”>asterisk-users@lists.digium.com

    Hi,

    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.
    href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Sebastian [shop@open-t.co.uk]
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  • Here is what I do today and it works fine:

    – asterisk/trixbox
    – Dext/android phone
    – Bell Canada cell provider
    – 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)
    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 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)

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  • 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

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  • 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.

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  • 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
    ________________________________________
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    Sent: Sunday, October 31, 2010 9:24 AM
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    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

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  • 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

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