[asterisk-users] SIP client on a mobile?
Steve Totaro wrote: Just use a SIP client on your phone. Many providers have multiple failover paths for inbound calls. This thread morphed from a nice home phone system into something completely different. Yup. For my situation, DISA is pointless except for road warriors who call all over the world, from anywhere, they can call into the corp system, get dialtone and skip the whole process of expense reports for work related calls. It makes things less complex, not more. Using DISA also means getting a corp caller id, not a mobile. Yes, spoofing provides that. Maybe if you explain your situation and how your plan works, but for me, personally, DISA would be a an added cost and complication. The only purpose I can think of for myself could be accomplished by spoofing caller id. How is that done from a mobile? Sofar that has been my main reason for using DISA - cost is not a real issue. SIP client. Spoof card, yes it is DISA, but you don't have to do anything but use the card. Steve, even if I could get SIP clients for our phones, doesn't this mean using a data connection rather than just voice? That would make it a lot pricier than the current setup with DISA (which is largely free). /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] How to know how many user is connected
I'm not a php expert, but seems your php script is incomplete/ you are
sending to socket (fputs) but note receiving anything(fgets) :
See this page
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP will
help you.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, August 24, 2011 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know how many user is connected
Hi List,
I want to know how many manager is connected into asterisk server. I have
made simple file but I don't have any idea how to get information back from
Asterisk CLI
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: root\r\n);
fputs($socket, Secret: energy\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: manager show connected\r\n);
$done=1;
}
?
Now how to get information into this PHP file
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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Re: [asterisk-users] How to know how many user is connected
hi:
please refer this:
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP
and check the manager.conf, make sure the accounts in managers.conf matchs the
managers displayed.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: gohar.ah...@vopium.com
To: asterisk-users@lists.digium.com
Date: Thu, 25 Aug 2011 11:26:53 +0500
Subject: Re: [asterisk-users] How to know how many user is connected
I’m not a php expert, but seems your php script is incomplete/ you are sending
to socket (fputs) but note receiving anything(fgets) :See this page will help
you. From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, August 24, 2011 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know how many user is connected Hi List,
I want to know how many manager is connected into asterisk server. I have made
simple file but I don't have any idea how to get information back from Asterisk
CLI
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: root\r\n);
fputs($socket, Secret: energy\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: manager show connected\r\n);
$done=1;
}
?
Now how to get information into this PHP file
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
_
-- 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:
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Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
Steve Totaro wrote: VoIP mostly aside, a couple more thoughts. I am not sure I understand your reasoning for DISA or how it is cheaper. The only reason we use DISA is to spoof the caller id. The OP also wanted to save costs, which is also possible (as someone already confirmed). DISA does save some cost for me too, but it is immaterial. The call from the mobile to the asterisk box is free or flat fee due to calling groups offered by our provider. The outgoing call is charged at regular fixnet prices, much cheaper than mobile ditto. You can buy a card that accepts SIMs as FXO and FXS. For your reasoning, a card of such nature is required. Populate it with different SIMs or whatever that are in calling groups or whatever you were trying to say. You've lost me, I have no idea what you're talking about. Just use callback back and some logic to reduce your costs. Call back will allow you to use the corp identity, and LCR will cut costs over DISA. The system calls you back after you make a call. Then the call is placed. There is a very brief outbound cell phone call, followed by a an inbound call from the server that you initiated with call back. OK, I see. I haven't looked at that, but it sounds more complicated than using DISA, and I'm not convinced it would be any cheaper. (it's important that the scheme be easy to use from the mobile end). Inbound to a cell is generally less expensive that oubound on a cell, sometimes completely free. Yes, inbound to a mobile is free as long as you're not roaming. However, with our calling group setup, it doesn't matter who (fix or mobile) originates the call, the cost is the same. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] How to know how many user is connected
Hi
You can use simple cli command
Manager show connected
On Thursday, August 25, 2011, James zhu zhulizh...@live.com wrote:
hi:
please refer this:
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP
and check the manager.conf, make sure the accounts in managers.conf matchs
the managers displayed.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: gohar.ah...@vopium.com
To: asterisk-users@lists.digium.com
Date: Thu, 25 Aug 2011 11:26:53 +0500
Subject: Re: [asterisk-users] How to know how many user is connected
I’m not a php expert, but seems your php script is incomplete/ you are
sending to socket (fputs) but note receiving anything(fgets) :
See this page
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP will help
you.
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, August 24, 2011 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know how many user is connected
Hi List,
I want to know how many manager is connected into asterisk server. I have
made simple file but I don't have any idea how to get information back from
Asterisk CLI
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: root\r\n);
fputs($socket, Secret: energy\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: manager show connected\r\n);
$done=1;
}
?
Now how to get information into this PHP file
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
-- _
-- 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] SIP client on a mobile?
On Thu, Aug 25, 2011 at 2:27 AM, Per Jessen p...@computer.org wrote: Steve Totaro wrote: Just use a SIP client on your phone. Many providers have multiple failover paths for inbound calls. This thread morphed from a nice home phone system into something completely different. Yup. For my situation, DISA is pointless except for road warriors who call all over the world, from anywhere, they can call into the corp system, get dialtone and skip the whole process of expense reports for work related calls. It makes things less complex, not more. Using DISA also means getting a corp caller id, not a mobile. Yes, spoofing provides that. Maybe if you explain your situation and how your plan works, but for me, personally, DISA would be a an added cost and complication. The only purpose I can think of for myself could be accomplished by spoofing caller id. How is that done from a mobile? Sofar that has been my main reason for using DISA - cost is not a real issue. SIP client. Spoof card, yes it is DISA, but you don't have to do anything but use the card. Steve, even if I could get SIP clients for our phones, doesn't this mean using a data connection rather than just voice? That would make it a lot pricier than the current setup with DISA (which is largely free). /Per Jessen, Zürich A Wifi connection? I guess that wifi is not like it is here. I can get on highspeed wifi anywhere I go in the DC Metro area for free. Just driving around, there is always an open access point. When driving around, I pick up thousands of APs in a couple miles and don't have any protection at all. I would suspect that most road warriors have high speed data needs? Not sure what business you are in, but having fast internet (relatively speaking) is a must to do work. I am not saying to use the data supplied from phone, if that is what you are thinking. If your phones don't have SIP, then use callback. You call your company, go through whatever you seutp in the dialplan, and the phone system calls you back as well as calling the other party. You edited out much of the context of the conversation to support your side. I don't play games like that... SIP client on the phone was an option. Was the original question about using DISA to save money? Yes it was. Now you are stating that it is largely free. Callback is a great solution when outbound cell phone calls quite a bit more than your cutrate VoIP provider. As I said, many countries do not charge for inbound calls. I am still clueless what your point is/was but if it is almost free then, stick with it. Still clueless why you posted if it almost free. Thanks, Steve Totaro -- _ -- 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] 1.8.5 Voicemail duration incorrect
Hi, Am Mittwoch, den 24.08.2011, 13:18 -0400 schrieb Robert Huddleston: Anyone else seen this? I saw a jira but was in feedback status.. I just checked with a voicemail of 60 seconds. It was reported in .txt-file with a duration of 19 seconds. So there is a bug. Do You have a link to the Jira issue? Karsten -- _ -- 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] SIP client on a mobile?
On Thu, Aug 25, 2011 at 3:06 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Thu, Aug 25, 2011 at 2:27 AM, Per Jessen p...@computer.org wrote: Steve Totaro wrote: Just use a SIP client on your phone. Many providers have multiple failover paths for inbound calls. This thread morphed from a nice home phone system into something completely different. Yup. For my situation, DISA is pointless except for road warriors who call all over the world, from anywhere, they can call into the corp system, get dialtone and skip the whole process of expense reports for work related calls. It makes things less complex, not more. Using DISA also means getting a corp caller id, not a mobile. Yes, spoofing provides that. Maybe if you explain your situation and how your plan works, but for me, personally, DISA would be a an added cost and complication. The only purpose I can think of for myself could be accomplished by spoofing caller id. How is that done from a mobile? Sofar that has been my main reason for using DISA - cost is not a real issue. SIP client. Spoof card, yes it is DISA, but you don't have to do anything but use the card. Steve, even if I could get SIP clients for our phones, doesn't this mean using a data connection rather than just voice? That would make it a lot pricier than the current setup with DISA (which is largely free). /Per Jessen, Zürich A Wifi connection? I guess that wifi is not like it is here. I can get on highspeed wifi anywhere I go in the DC Metro area for free. Just driving around, there is always an open access point. When driving around, I pick up thousands of APs in a couple miles and don't have any protection at all. I would suspect that most road warriors have high speed data needs? Not sure what business you are in, but having fast internet (relatively speaking) is a must to do work. I am not saying to use the data supplied from phone, if that is what you are thinking. If your phones don't have SIP, then use callback. You call your company, go through whatever you seutp in the dialplan, and the phone system calls you back as well as calling the other party. You edited out much of the context of the conversation to support your side. I don't play games like that... SIP client on the phone was an option. Was the original question about using DISA to save money? Yes it was. Now you are stating that it is largely free. Callback is a great solution when outbound cell phone calls quite a bit more than your cutrate VoIP provider. As I said, many countries do not charge for inbound calls. I am still clueless what your point is/was but if it is almost free then, stick with it. Still clueless why you posted if it almost free. Thanks, Steve Totaro I am not sure why people try to prove me wrong, but they do. On rare occasions, I am wrong, I am also big enough to admit it. To answer your question, and get on the same terms, VoIP (or data as you prefer) would probably be cheaper. Isn't that the whole reason behind VoIP? You say voice, does that mean your provider's voice service? Depending on the cost of inbound and out abound calls on a cell are the key here. Is it next to nothing to call a foreign country from your cell? Is it much more expensive than rates at the office. Generally, I think outbound calls from an office are much lower than cell phone charges. I was paying a $40k plus weekly for long distance calls from Iraq to mostly Fiji, Uganda, Peru. That was with VoicePulse, all 703 DIDs around the world. Voicepulse gave me great rates because $40k a week is not chump change. I wonder what the cost of cell phone calls would amount to? My international rates for outbound cell phone calls are beyond a rip-off. Thanks, Steve Totaro -- _ -- 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] Looking for ideas for nice **Asterisk** home phone system
So in other worlds you had nothing to contribute to this thread. On Thu, Aug 25, 2011 at 2:44 AM, Per Jessen p...@computer.org wrote: Steve Totaro wrote: VoIP mostly aside, a couple more thoughts. I am not sure I understand your reasoning for DISA or how it is cheaper. The only reason we use DISA is to spoof the caller id. The OP also wanted to save costs, which is also possible (as someone already confirmed). DISA does save some cost for me too, but it is immaterial. The call from the mobile to the asterisk box is free or flat fee due to calling groups offered by our provider. The outgoing call is charged at regular fixnet prices, much cheaper than mobile ditto. You can buy a card that accepts SIMs as FXO and FXS. For your reasoning, a card of such nature is required. Populate it with different SIMs or whatever that are in calling groups or whatever you were trying to say. You've lost me, I have no idea what you're talking about. Just use callback back and some logic to reduce your costs. Call back will allow you to use the corp identity, and LCR will cut costs over DISA. The system calls you back after you make a call. Then the call is placed. There is a very brief outbound cell phone call, followed by a an inbound call from the server that you initiated with call back. OK, I see. I haven't looked at that, but it sounds more complicated than using DISA, and I'm not convinced it would be any cheaper. (it's important that the scheme be easy to use from the mobile end). Inbound to a cell is generally less expensive that oubound on a cell, sometimes completely free. Yes, inbound to a mobile is free as long as you're not roaming. However, with our calling group setup, it doesn't matter who (fix or mobile) originates the call, the cost is the same. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] Looking for ideas for nice **Asterisk** home phone system
Steve Totaro wrote: So in other worlds you had nothing to contribute to this thread. I did - you didn't understand my reasoning, I explained it. If you had nothing to contribute to this thread, perhaps you should have stayed away too. /Per Jessen, Zürich -- _ -- 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] How to know how many user is connected
Hi List,
Thanks for guide me. Yes I know that CLI command , My motive is to get
information into Php that's why I am finding the solution.
Ahmad Sir, You are right I forget to get information back from CLI to Php
file. Thanks for provide the help link.I will revert back after testing my
code with your guidance
On Thu, Aug 25, 2011 at 12:21 PM, DHAVAL INDRODIYA dhaval.it01...@gmail.com
wrote:
Hi
You can use simple cli command
Manager show connected
On Thursday, August 25, 2011, James zhu zhulizh...@live.com wrote:
hi:
please refer this:
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP
and check the manager.conf, make sure the accounts in managers.conf
matchs the managers displayed.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: gohar.ah...@vopium.com
To: asterisk-users@lists.digium.com
Date: Thu, 25 Aug 2011 11:26:53 +0500
Subject: Re: [asterisk-users] How to know how many user is connected
I’m not a php expert, but seems your php script is incomplete/ you are
sending to socket (fputs) but note receiving anything(fgets) :
See this page
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP will
help you.
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, August 24, 2011 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know how many user is connected
Hi List,
I want to know how many manager is connected into asterisk server. I have
made simple file but I don't have any idea how to get information back from
Asterisk CLI
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: root\r\n);
fputs($socket, Secret: energy\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: manager show connected\r\n);
$done=1;
}
?
Now how to get information into this PHP file
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
-- _
-- 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
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
_
-- 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] Looking for ideas for nice **Asterisk** home phone system
Steve, On Thu, 2011-08-25 at 00:39 -0400, Steve Totaro wrote: ... For fax, I use Hylafax and for text, I use Kannel. These are WAY more powerful than Asterisk apps. With Kannel, I used the Bluetooth GSM modem to send SMS from my cell. Kannel is awesome as is HylaFAX I used the Asteirsk System() app to call lynx with a special URL. The URL contains all the authentication, recipient, and SMS body. Calling that URL via System(), as I said, I like lynx, causes an SMS to be sent. Kannel is extremely customizable. I once had ten cell phones for for SMS modems. My findings with t-mobile were that each phone could send an SMS once a second. With ten, using chan_bluetooth, I could send ten SMS per second using ten phones. Kannel is very well developed. Chan_mobile is incredible. The same is true with HylaFAX. Thanks, Steve T I'm looking at using Kannel for a project here. Would you mind if I contacted you off list with some getting started questions? Skyler -- _ -- 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] SIP client on a mobile?
Steve Totaro wrote: Steve, even if I could get SIP clients for our phones, doesn't this mean using a data connection rather than just voice? That would make it a lot pricier than the current setup with DISA (which is largely free). /Per Jessen, Zürich A Wifi connection? I guess that wifi is not like it is here. I can get on highspeed wifi anywhere I go in the DC Metro area for free. In the cities, WiFi is typically only available in restaurants and cafes (Starbucks, McDonalds etc). In the country, no wifi. Well, the odd open access point, but using it is illegal, so that's a no-go. I would suspect that most road warriors have high speed data needs? Not sure what business you are in, but having fast internet (relatively speaking) is a must to do work. I am not saying to use the data supplied from phone, if that is what you are thinking. For my company, the mobile is primarily for voice - people don't spend that much time on the road, but when they do, they still want to appear as if they're in the office. If your phones don't have SIP, then use callback. You call your company, go through whatever you seutp in the dialplan, and the phone system calls you back as well as calling the other party. You edited out much of the context of the conversation to support your side. I don't play games like that... Sorry, that wasn't my intention, I just snip out the bits that aren't relevant to a reply. SIP client on the phone was an option. Was the original question about using DISA to save money? Yes it was. Now you are stating that it is largely free. I think the OPs question was about saving money, to which I suggested using DISA - it my setup it's largely free. Callback is a great solution when outbound cell phone calls quite a bit more than your cutrate VoIP provider. As I said, many countries do not charge for inbound calls. Right. I am still clueless what your point is/was but if it is almost free then, stick with it. Still clueless why you posted if it almost free. I did not post the original question, I just responded to it. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] Looking for ideas for nice **Asterisk** home phone system
On Thu, Aug 25, 2011 at 4:36 AM, Skyler skchopper...@gmail.com wrote: Steve, On Thu, 2011-08-25 at 00:39 -0400, Steve Totaro wrote: ... For fax, I use Hylafax and for text, I use Kannel. These are WAY more powerful than Asterisk apps. With Kannel, I used the Bluetooth GSM modem to send SMS from my cell. Kannel is awesome as is HylaFAX I used the Asteirsk System() app to call lynx with a special URL. The URL contains all the authentication, recipient, and SMS body. Calling that URL via System(), as I said, I like lynx, causes an SMS to be sent. Kannel is extremely customizable. I once had ten cell phones for for SMS modems. My findings with t-mobile were that each phone could send an SMS once a second. With ten, using chan_bluetooth, I could send ten SMS per second using ten phones. Kannel is very well developed. Chan_mobile is incredible. The same is true with HylaFAX. Thanks, Steve T I'm looking at using Kannel for a project here. Would you mind if I contacted you off list with some getting started questions? Skyler Skyler, I would be glad to help within reason. Since it is not Asterisk and I use app System() and Lynx as the glue, it wouldn't fit asterisk user's list anyways. I use fast AGI for most of the SMS variables. Helping within reason is good for my karma, too much and I need to be compensated. At the very least, thanked publically ; Like the old Italian saying, I give my friends just enough so that they need me, but not too much so that they dont I have quite a bit of experience with Kannel and the code. Hit me up and let's see what help I can provide. Thanks, Steve T -- _ -- 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] Looking for ideas for nice **Asterisk** home phone system
On Thu, Aug 25, 2011 at 12:39:14AM -0400, Steve Totaro wrote: I used the Asteirsk System() app to call lynx with a special URL. The URL contains all the authentication, recipient, and SMS body. Calling that URL via System(), as I said, I like lynx, causes an SMS to be sent. Kannel is extremely customizable. Slightly off-topic: why not use CURL()? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] security: SIP header spoofing CHANNEL(recvip)?
I am currently suffering various SIP attacks. I am using the following
extension to record the caller's IP address:
exten = h,n,set(CDR(srcip)=${CHANNEL(recvip)})
However, in recent attacks, this IP address is not correct, and I
believe that they are spoofing it. I am using asterisk 1.6.2.15.
Does the CHANNEL(recvip) variable record IP show in the SIP header
instead of the real, UDP source IP? If the CHANNEL(recvip) variable
records the IP address set in the SIP header, and not the real IP
address, how can I obtain the REAL IP address of the caller?
--
_
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Re: [asterisk-users] How to know how many user is connected
Hi List,
Thanks now I am able to get all values from asterisk CLI but I want 2 more
things .
1) I want to disconnect *connected manager* into Asterisk. Is it possible ?
2) I want to maintain this socket connection until we disconnect it from web
page.
On Thu, Aug 25, 2011 at 1:57 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
Thanks for guide me. Yes I know that CLI command , My motive is to get
information into Php that's why I am finding the solution.
Ahmad Sir, You are right I forget to get information back from CLI to Php
file. Thanks for provide the help link.I will revert back after testing my
code with your guidance
On Thu, Aug 25, 2011 at 12:21 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi
You can use simple cli command
Manager show connected
On Thursday, August 25, 2011, James zhu zhulizh...@live.com wrote:
hi:
please refer this:
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP
and check the manager.conf, make sure the accounts in managers.conf
matchs the managers displayed.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: gohar.ah...@vopium.com
To: asterisk-users@lists.digium.com
Date: Thu, 25 Aug 2011 11:26:53 +0500
Subject: Re: [asterisk-users] How to know how many user is connected
I’m not a php expert, but seems your php script is incomplete/ you are
sending to socket (fputs) but note receiving anything(fgets) :
See this page
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP will
help you.
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, August 24, 2011 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know how many user is connected
Hi List,
I want to know how many manager is connected into asterisk server. I
have made simple file but I don't have any idea how to get information back
from Asterisk CLI
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: root\r\n);
fputs($socket, Secret: energy\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: manager show connected\r\n);
$done=1;
}
?
Now how to get information into this PHP file
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
-- _
-- 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
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
_
-- 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] How to know how many user is connected
What I understood: you need to disconnect the AMI socket.
1) I want to disconnect connected manager into Asterisk. Is it possible ?
ß Close the $socket after you get the response.
What I understood: you need to maintain the socket until some button is
pressed to stop AMI
2) I want to maintain this socket connection until we disconnect it from web
page. ß Close the $socket on particular action from web-page. This ones
tricky btw maintain a while loop and break loop on a condition toggled by
web-page)
See php section for other examples.
http://www.voip-info.org/wiki/view/Asterisk+manager+Examples
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, August 25, 2011 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to know how many user is connected
Hi List,
Thanks now I am able to get all values from asterisk CLI but I want 2 more
things .
1) I want to disconnect connected manager into Asterisk. Is it possible ?
2) I want to maintain this socket connection until we disconnect it from web
page.
On Thu, Aug 25, 2011 at 1:57 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
Thanks for guide me. Yes I know that CLI command , My motive is to get
information into Php that's why I am finding the solution.
Ahmad Sir, You are right I forget to get information back from CLI to Php
file. Thanks for provide the help link.I will revert back after testing my
code with your guidance
On Thu, Aug 25, 2011 at 12:21 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi
You can use simple cli command
Manager show connected
On Thursday, August 25, 2011, James zhu zhulizh...@live.com wrote:
hi:
please refer this:
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP
and check the manager.conf, make sure the accounts in managers.conf matchs
the managers displayed.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: gohar.ah...@vopium.com
To: asterisk-users@lists.digium.com
Date: Thu, 25 Aug 2011 11:26:53 +0500
Subject: Re: [asterisk-users] How to know how many user is connected
Im not a php expert, but seems your php script is incomplete/ you are
sending to socket (fputs) but note receiving anything(fgets) :
See this page
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP will help
you.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, August 24, 2011 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know how many user is connected
Hi List,
I want to know how many manager is connected into asterisk server. I have
made simple file but I don't have any idea how to get information back from
Asterisk CLI
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: root\r\n);
fputs($socket, Secret: energy\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: manager show connected\r\n);
$done=1;
}
?
Now how to get information into this PHP file
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
-- _
-- 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
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
_
-- 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] How to know how many user is connected
Just realized that maybe youre talking about disconnecting any other
AMI/manger connected user from another manager connection
hhmmm
I dont
think so. Check AMI commands from asterisk wiki. If not, you may need system
command in your AMI connection to close some other socket.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gohar Ahmed
Sent: Thursday, August 25, 2011 4:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to know how many user is connected
What I understood: you need to disconnect the AMI socket.
1) I want to disconnect connected manager into Asterisk. Is it possible ?
ß Close the $socket after you get the response.
What I understood: you need to maintain the socket until some button is
pressed to stop AMI
2) I want to maintain this socket connection until we disconnect it from web
page. ß Close the $socket on particular action from web-page. This ones
tricky btw maintain a while loop and break loop on a condition toggled by
web-page)
See php section for other examples.
http://www.voip-info.org/wiki/view/Asterisk+manager+Examples
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, August 25, 2011 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to know how many user is connected
Hi List,
Thanks now I am able to get all values from asterisk CLI but I want 2 more
things .
1) I want to disconnect connected manager into Asterisk. Is it possible ?
2) I want to maintain this socket connection until we disconnect it from web
page.
On Thu, Aug 25, 2011 at 1:57 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
Thanks for guide me. Yes I know that CLI command , My motive is to get
information into Php that's why I am finding the solution.
Ahmad Sir, You are right I forget to get information back from CLI to Php
file. Thanks for provide the help link.I will revert back after testing my
code with your guidance
On Thu, Aug 25, 2011 at 12:21 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi
You can use simple cli command
Manager show connected
On Thursday, August 25, 2011, James zhu zhulizh...@live.com wrote:
hi:
please refer this:
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP
and check the manager.conf, make sure the accounts in managers.conf matchs
the managers displayed.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: gohar.ah...@vopium.com
To: asterisk-users@lists.digium.com
Date: Thu, 25 Aug 2011 11:26:53 +0500
Subject: Re: [asterisk-users] How to know how many user is connected
Im not a php expert, but seems your php script is incomplete/ you are
sending to socket (fputs) but note receiving anything(fgets) :
See this page
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP will help
you.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, August 24, 2011 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know how many user is connected
Hi List,
I want to know how many manager is connected into asterisk server. I have
made simple file but I don't have any idea how to get information back from
Asterisk CLI
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: root\r\n);
fputs($socket, Secret: energy\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: manager show connected\r\n);
$done=1;
}
?
Now how to get information into this PHP file
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
-- _
-- 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
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
-
Thanks and regards
Re: [asterisk-users] How to know how many user is connected
Hi Ahmed,
Just realized that maybe you’re talking about disconnecting any other
AMI/manger connected user from another manager connection…hhmmm… I don’t
think so. Check AMI commands from asterisk wiki. If not, you may need system
command in your AMI connection to close some other socket
*
Yes I was looking for this :)
*Please tell me how to close other socket from current sockets.
one more thing in my case it may be possible that
root 127.0.0.1 may be more then one then how to close them individually? *
*
On Thu, Aug 25, 2011 at 5:09 PM, Gohar Ahmed gohar.ah...@vopium.com wrote:
Just realized that maybe you’re talking about disconnecting any other
AMI/manger connected user from another manager connection…hhmmm… I don’t
think so. Check AMI commands from asterisk wiki. If not, you may need system
command in your AMI connection to close some other socket.
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gohar Ahmed
*Sent:* Thursday, August 25, 2011 4:25 PM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] How to know how many user is connected
** **
What I understood: you need to disconnect the AMI socket.
1) I want to disconnect *connected manager* into Asterisk. Is it possible
?ß Close the $socket after you get the response.
** **
What I understood: you need to maintain the socket until some button is
pressed to stop AMI
2) I want to maintain this socket connection until we disconnect it from
web page. ß Close the $socket on particular action from web-page. This
one’s tricky btw maintain a while loop and break loop on a condition toggled
by web-page)
See php section for other examples.
http://www.voip-info.org/wiki/view/Asterisk+manager+Examples
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
*Sent:* Thursday, August 25, 2011 4:02 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] How to know how many user is connected
** **
Hi List,
Thanks now I am able to get all values from asterisk CLI but I want 2 more
things .
1) I want to disconnect *connected manager* into Asterisk. Is it possible
?
2) I want to maintain this socket connection until we disconnect it from
web page.
On Thu, Aug 25, 2011 at 1:57 PM, virendra bhati virbh...@gmail.com
wrote:
Hi List,
Thanks for guide me. Yes I know that CLI command , My motive is to get
information into Php that's why I am finding the solution.
Ahmad Sir, You are right I forget to get information back from CLI to Php
file. Thanks for provide the help link.I will revert back after testing my
code with your guidance
On Thu, Aug 25, 2011 at 12:21 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi
You can use simple cli command
Manager show connected
On Thursday, August 25, 2011, James zhu zhulizh...@live.com wrote:
hi:
please refer this:
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP
and check the manager.conf, make sure the accounts in managers.conf
matchs the managers displayed.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: gohar.ah...@vopium.com
To: asterisk-users@lists.digium.com
Date: Thu, 25 Aug 2011 11:26:53 +0500
Subject: Re: [asterisk-users] How to know how many user is connected
I’m not a php expert, but seems your php script is incomplete/ you are
sending to socket (fputs) but note receiving anything(fgets) :
See this page
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP will
help you.
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, August 24, 2011 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know how many user is connected
Hi List,
I want to know how many manager is connected into asterisk server. I have
made simple file but I don't have any idea how to get information back from
Asterisk CLI
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: root\r\n);
fputs($socket, Secret: energy\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: manager show connected\r\n);
$done=1;
}
?
Now how to get information into this PHP file
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
[asterisk-users] Bind SIP over TCP port in asterisk 1.4.42.
Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Thank you very much. Jonson. -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
On Thu, Aug 25, 2011 at 9:26 AM, Catalin S. jonsonpla...@gmail.com wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Thank you very much. Jonson. -- I looked TCP + Transport are listed in http://svn.asterisk.org/svn/asterisk/branches/1.4/channels/chan_sip.c but not in http://svn.asterisk.org/svn/asterisk/branches/1.4/configs/sip.conf.sample try transport=TCP Beware, some systems use SIP(not encrypted) over TCP on port 5061, which is not really wrong, just not what the standards say. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
Hi, On Thu, 2011-08-25 at 16:26 +0300, Catalin S. wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Not 100% with 1.4 but with 1.6 you don't need to set tcpbindaddr because tcpenable will listen on same IP as udp. No transport either I believe. If you want, set udpbindaddr and tcp will listen on this IP too. tcpenable=yes is all you should need. S. -- _ -- 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] Possibly odd sip.conf security requirements. Possible?
Hi, Is the following possible in some way? I want to have several SIP providers able to send me calls, each provider may send calls into many possible DDIs. Each provider has a cluster of servers, but is unable to authenticate with me, so the following would be a sort of pseudo-code sip.conf example. [general] context = barred ; Unknown/other source of calls [provider 1] type = peer context = provider1-context ; deal with provider's calls 1 deny = 0.0.0.0/0.0.0.0 permit = 12.13.14.0/24 ; This provider has servers in this range [provider 2] type = peer context = provider2-context ; deal with provider's calls 2 deny = 0.0.0.0/0.0.0.0 permit = 22.23.24.0/24 ; This provider has servers in this range [provider 3] type = peer context = provider3-context ; deal with provider's calls 3 deny = 0.0.0.0/0.0.0.0 permit = 32.33.34.0/24 ; This provider has servers in this range Normally a call into SIP has one of 3 paths: 1) Unauthenticated, so use the default 2) Identifiable username 3) Identifiable IP address In the above example, we have a BLOCK of IP addresses instead of a single address. Can this be made to work? Thanks for any pointers. Regards, Steve -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
hello, I tried still not working. :( something is wrong. On Thu, Aug 25, 2011 at 4:37 PM, Skyler skchopper...@gmail.com wrote: Hi, On Thu, 2011-08-25 at 16:26 +0300, Catalin S. wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Not 100% with 1.4 but with 1.6 you don't need to set tcpbindaddr because tcpenable will listen on same IP as udp. No transport either I believe. If you want, set udpbindaddr and tcp will listen on this IP too. tcpenable=yes is all you should need. S. -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
Hello, I tried but still not works. Can you make some test at your side? Something is wrong. Thank you. On Thu, Aug 25, 2011 at 4:35 PM, Andrew Latham lath...@gmail.com wrote: On Thu, Aug 25, 2011 at 9:26 AM, Catalin S. jonsonpla...@gmail.com wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Thank you very much. Jonson. -- I looked TCP + Transport are listed in http://svn.asterisk.org/svn/asterisk/branches/1.4/channels/chan_sip.c but not in http://svn.asterisk.org/svn/asterisk/branches/1.4/configs/sip.conf.sample try transport=TCP Beware, some systems use SIP(not encrypted) over TCP on port 5061, which is not really wrong, just not what the standards say. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
On 11-08-25 09:26 AM, Catalin S. wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Asterisk 1.4 does not have support SIP over TCP. It was added in Asterisk 1.6.0. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
I have the below in my sip.conf bindaddr = PublicIP tcpenable = yes tcpbindaddr = PublicIP On Thu, 2011-08-25 at 17:22 +0300, Catalin S. wrote: hello, I tried still not working. :( something is wrong. On Thu, Aug 25, 2011 at 4:37 PM, Skyler skchopper...@gmail.com wrote: Hi, On Thu, 2011-08-25 at 16:26 +0300, Catalin S. wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Not 100% with 1.4 but with 1.6 you don't need to set tcpbindaddr because tcpenable will listen on same IP as udp. No transport either I believe. If you want, set udpbindaddr and tcp will listen on this IP too. tcpenable=yes is all you should need. S. -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
Hello Paul, I choose 1.4.42 version because of iksemel for gtalk module. I need gtalk feature for my asterisk and when i upgrade it at 1.6 this module doesn't work. Can you tell me if is some trick to make 1.4.42 to work with tcp option? Maybe some patches... etc. Thank you. On Thu, Aug 25, 2011 at 5:25 PM, Paul Belanger pabelan...@digium.comwrote: On 11-08-25 09:26 AM, Catalin S. wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Asterisk 1.4 does not have support SIP over TCP. It was added in Asterisk 1.6.0. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- __**__**_ -- 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-**usershttp://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] Bind SIP over TCP port in asterisk 1.4.42.
On 11-08-25 10:34 AM, Catalin S. wrote: Hello Paul, I choose 1.4.42 version because of iksemel for gtalk module. I need gtalk feature for my asterisk and when i upgrade it at 1.6 this module doesn't work. Can you tell me if is some trick to make 1.4.42 to work with tcp option? Maybe some patches... etc. Well, both 1.4 and 1.6 branches are unsupported so you should move to asterisk 1.8 and test. There is no 'trick' for adding TCP over SIP into Asterisk 1.4, that is not realistic. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
Thank you Paul for answers. Please tell me if i upgrade to 1.8 is gtalk module still working? On Thu, Aug 25, 2011 at 5:42 PM, Paul Belanger pabelan...@digium.comwrote: On 11-08-25 10:34 AM, Catalin S. wrote: Hello Paul, I choose 1.4.42 version because of iksemel for gtalk module. I need gtalk feature for my asterisk and when i upgrade it at 1.6 this module doesn't work. Can you tell me if is some trick to make 1.4.42 to work with tcp option? Maybe some patches... etc. Well, both 1.4 and 1.6 branches are unsupported so you should move to asterisk 1.8 and test. There is no 'trick' for adding TCP over SIP into Asterisk 1.4, that is not realistic. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- __**__**_ -- 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-**usershttp://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] Bind SIP over TCP port in asterisk 1.4.42.
On 11-08-25 10:58 AM, Catalin S. wrote: Thank you Paul for answers. Please tell me if i upgrade to 1.8 is gtalk module still working? There was some reports last week that is was broken, because google was making some protocol changes. However, I believe they reverted them. Hopefully somebody else on the -users list can comment. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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] security: SIP header spoofing CHANNEL(recvip)?
I was wondering if these could be spoofed recently when reading the docs.
Have you tried peerip rather than recvip?
Does that give the same result?
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Recarey
Sent: 25 August 2011 11:34
To: Asterisk Users Mailing List
Subject: [asterisk-users] security: SIP header spoofing CHANNEL(recvip)?
I am currently suffering various SIP attacks. I am using the following
extension to record the caller's IP address:
exten = h,n,set(CDR(srcip)=${CHANNEL(recvip)})
However, in recent attacks, this IP address is not correct, and I
believe that they are spoofing it. I am using asterisk 1.6.2.15.
Does the CHANNEL(recvip) variable record IP show in the SIP header
instead of the real, UDP source IP? If the CHANNEL(recvip) variable
records the IP address set in the SIP header, and not the real IP
address, how can I obtain the REAL IP address of the caller?
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Re: [asterisk-users] Asterisk Integration with Android device
Thanks for all your comments. Actually I have 3G connection but even then the signal in my mobile automatically changes from 3g to 2G; it is automatically going to Edge signal. Anyways let me try with some other softphone like media5. Regards, Gopal On Thu, Aug 25, 2011 at 9:21 AM, James Perkins ja...@clove.net.au wrote: ** Try media5 fone. I couldn't get 3cx to work on my iphone and tried about 7 different softfones. Media5 is the best by a long shot. Android is still in better and haven't tried it but if its anything like their iphone app it will be worth a look. There is a signup for the better at the website. let us know how you go. James - Original Message - *From:* bakko asannu...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, August 24, 2011 11:48 PM *Subject:* Re: [asterisk-users] Asterisk Integration with Android device I think don't work with 2G network. Regards - Original Message - *From:* Gopal krishnan gopalakrishnan...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, August 24, 2011 4:01 PM *Subject:* [asterisk-users] Asterisk Integration with Android device Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be appreciated. Regards, Gopal -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX MOS Score measuring solution
Greetings ! Has anyone used any solution for getting the MOS Score on IAX channels using codes like g729. I have found a few but all are measuring sip and/or a-ulaw. Regards Stelios -- _ -- 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] Asterisk Integration with Android device
Hi find the answer inline On Thu, Aug 25, 2011 at 21:24, Gopal krishnan gopalakrishnan...@gmail.comwrote: Thanks for all your comments. Actually I have 3G connection but even then the signal in my mobile automatically changes from 3g to 2G; it is automatically going to Edge signal. Anyways let me try with some other softphone like media5. Regards, Gopal On Thu, Aug 25, 2011 at 9:21 AM, James Perkins ja...@clove.net.au wrote: ** Try media5 fone. I couldn't get 3cx to work on my iphone and tried about 7 different softfones. Media5 is the best by a long shot. Android is still in better and haven't tried it but if its anything like their iphone app it will be worth a look. There is a signup for the better at the website. let us know how you go. James - Original Message - *From:* bakko asannu...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, August 24, 2011 11:48 PM *Subject:* Re: [asterisk-users] Asterisk Integration with Android device I think don't work with 2G network. Regards - Original Message - *From:* Gopal krishnan gopalakrishnan...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, August 24, 2011 4:01 PM *Subject:* [asterisk-users] Asterisk Integration with Android device Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be appreciated. Regards, Gopal -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Try sipdroid for android -- Amit Anand +91 9818559898 -- _ -- 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] 1.8.5 Voicemail duration incorrect
https://issues.asterisk.org/jira/browse/ASTERISK-16981 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten Wemheuer Sent: Thursday, August 25, 2011 3:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.8.5 Voicemail duration incorrect Hi, Am Mittwoch, den 24.08.2011, 13:18 -0400 schrieb Robert Huddleston: Anyone else seen this? I saw a jira but was in feedback status.. I just checked with a voicemail of 60 seconds. It was reported in .txt-file with a duration of 19 seconds. So there is a bug. Do You have a link to the Jira issue? Karsten -- _ -- 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] IAX MOS Score measuring solution
On Thu, 25 Aug 2011 19:56:51 +0300
Stelios Koroneos skoron...@digital-opsis.com wrote:
Greetings !
Has anyone used any solution for getting the MOS Score on IAX channels
using codes like g729.
I have found a few but all are measuring sip and/or a-ulaw.
Regards
Stelios
You can extract data like rtt, jitter and packet loss from the
dialplan with something like:
${CHANNEL(rtpqos|audio|all)}
Based on these u can calculate R and MOS using the formulas on this
page:
http://www.nessoft.com/kb/50
Lefteris Zafiris
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[asterisk-users] Thunderbird extension using AMI to dial
Hi I've just added direct support for AMI to a forthcoming version of TBDialOut, a Thunderbird extension for dialling direct from Thunderbird's address book. If anyone fancies testing it I'd be grateful for any feedback. If you feel like casting a critical eye over the code, or doing some translating, even better. AMI support is available in TBDialOut 1.7.0pre1, which can be found either at http://www.oak-wood.co.uk/tbdialout/ or from the 'Development channel' at the bottom of the page at https://addons.mozilla.org/en-US/thunderbird/addon/tbdialout/ Thanks for your help Chris -- _ -- 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
