Re: [asterisk-users] Mobile Phones and Asterisk
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: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian [s...@open-t.co.uk] Sent: Sunday, October 31, 2010 9:24 AM To: asterisk-users@lists.digium.com 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 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 *98ext 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
Re: [asterisk-users] Mobile Phones and Asterisk
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: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian [s...@open-t.co.uk] Sent: Sunday, October 31, 2010 9:24 AM To: asterisk-users@lists.digium.com 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 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 *98ext 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
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 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 *98ext 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
Re: [asterisk-users] Mobile Phones and Asterisk
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 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 *98ext 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
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 +
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.commailto: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
Re: [asterisk-users] Mobile Phones and Asterisk
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: 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
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
Re: [asterisk-users] Mobile Phones and Asterisk
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
Re: [asterisk-users] Mobile Phones and Asterisk
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) - 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 *98ext 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
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 + 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.commailto: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 -- _ -- 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.euhttp://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
Re: [asterisk-users] Mobile Phones and Asterisk
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 ${DIALSTATUS})
I hear a busy tone, after the 10 sec. timeout it returns NOANSWER, as defined
in my DIAL func.
I also tried getting the DEVICE_STATE
exten =s,3,NoOp(SIP/xxx${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 +
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.commailto: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
--
_
-- Bandwidth and Colocation Provided
Re: [asterisk-users] 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 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 -- _ -- 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
Re: [asterisk-users] Mobile Phones and Asterisk
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
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 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 -- _ -- 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
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 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 -- _ -- 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
Re: [asterisk-users] Mobile Phones and Asterisk
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 From: ayodeleabej...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 26 Oct 2010 11:01:09 + 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 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 -- _ -- 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
[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 -- _ -- 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
