[asterisk-users] How to get Call-ID SIP header outside chan_sip scope ...
Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the source of information for the entire infraestructure; and, 2- A call control application that will be responsible for call timing control and pre-paid support; Here we are prioritizing internal modifications and loadable modules (like modules, applications, etc) against external AGI components to acchieve the best performance possible for the entire solution. One problem we have here is to find out the best option (even one that results on some internal Asterisk files changing) that allow us to propagate the SIP header Call-ID to both modules described above. The best shot we have until now is to use the callid field from the sip_pvt structure of SIP channel, what will lead us to two considerable code changes: 1- Propagate the channel to method realtime_var_get of our proprietary ARA driver; and 2- Duplication of necessary structs to a header (.h) file so the modules can navigate on private structure sip_pvt. The first change isn't big deal. But the need of validation of the second modification, every time we make a merge with updated codes is concerning me a lot. Does anyone have a better approach to get this done ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get Call-ID SIP header outside chan_sip scope ...
Hello there! I really hate when this happens, but... It seems channel variable SIPCALLID will have the info I need, so the changes will be reduced to propagate the channel to ARA driver method realtime_var_get. If someone have any additional info, or can indicate some problem on using this variable, please let me know. Thanks and best regards, Mauro. Mauro Sergio Ferreira Brasil escreveu: Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the source of information for the entire infraestructure; and, 2- A call control application that will be responsible for call timing control and pre-paid support; Here we are prioritizing internal modifications and loadable modules (like modules, applications, etc) against external AGI components to acchieve the best performance possible for the entire solution. One problem we have here is to find out the best option (even one that results on some internal Asterisk files changing) that allow us to propagate the SIP header Call-ID to both modules described above. The best shot we have until now is to use the callid field from the sip_pvt structure of SIP channel, what will lead us to two considerable code changes: 1- Propagate the channel to method realtime_var_get of our proprietary ARA driver; and 2- Duplication of necessary structs to a header (.h) file so the modules can navigate on private structure sip_pvt. The first change isn't big deal. But the need of validation of the second modification, every time we make a merge with updated codes is concerning me a lot. Does anyone have a better approach to get this done ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does L(x:y:z) Dial option work on Asterisk version 1.4 ?
Sorry guys.
My bad!
As you can see, the command on prior message is incorret.
I've changed to:
Dial(SIP/${EXTEN}|20|RtTL(30:6:2))
and it's working now.
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello there!
I'm testing Dial call limit option on Asterisk version 1.4.26, but
it's not working.
The issued command is: Dial(SIP/${EXTEN}|20|RtT|L(30:6:2)).
Am I missing something ?
Does it only work with Asterisk version 1.6.X ?
Thanks and best regards,
--
__At.,
_
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br mailto:@tqi.com.br
: www.tqi.com.br http://www.tqi.com.br
( + 55 (34)3291-1700
( + 55 (34)9971-2572
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[asterisk-users] [UOL - Manutenões Desktop] Controlling call duration ...
Hello there! The only available way to control call duration is using the RTCC patch (discussed here https://issues.asterisk.org/view.php?id=6335; and mainteined here http://ast.varna.net/;) ? The purpouse is to have a way to monitor (probably on a per-minute basis) and hangup costly calls (and/or multiple calls initiated by same SIP user). Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does L(x:y:z) Dial option work on Asterisk version 1.4 ?
Hello there!
I'm testing Dial call limit option on Asterisk version 1.4.26, but
it's not working.
The issued command is: Dial(SIP/${EXTEN}|20|RtT|L(30:6:2)).
Am I missing something ?
Does it only work with Asterisk version 1.6.X ?
Thanks and best regards,
--
__At.,
_
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br mailto:@tqi.com.br
: www.tqi.com.br http://www.tqi.com.br
( + 55 (34)3291-1700
( + 55 (34)9971-2572
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Re: [asterisk-users] Multiple user registration ...
Thanks a lot Faheem for you help.
I totaly understand now the approach you've used.
It's very interesting and inventive for sure.
I didn't know that I could append IP:Port info on user when using the
Dial command and that this will make calling to two different devices
registered using same user work.
With this little but extemelly important peace of information you gave
me the answer to our questions here.
Thanks again, and best regards,
Mauro.
Faheem escreveu:
The purpose of Perl script is to store user registrations records only
and nothing else regarding call dialing.
The script will main records like this.
User1:
IP1: 192.168.0.100 Por1: 5060
IP2: 69.30.21.10 Port2: 5060
User2:
IP1: 192.168.10.1 Por1: 5060
IP2: 192.168.10.1 Por2: 5061
User3:
IP1: 192.168.10.121 Por1: 5060
IP2: 192.168.10.123 Por2: 5061
and so on
No it all depends on you to store these information on files or database.
Assume you have stored IP/Ports in the database.
Database=cloneline
Table = users(username,ip1,port1,ip2,port2)
For dialing:
Assume username=user1 and extension =123456
exten= 123456,1,NoOp()
exten= 123456,n,MYSQL(Connect connid 'localhost' cdr dbpass cloneline)
exten= 123456,n,NoOP(Connection ID:${connid})
exten= 123456,n,MYSQL(Query resultid ${connid} SELECT\ ip1\, port1\,
ip2\, port2\, status\ from\ users\ where\ username=user1 )
exten= 123456,n,MYSQL(Fetch fetchid ${resultid} ip1 port1 ip2 port2)
exten= 123456,n,Dial(SIP/us...@${ip1}:${port1}SIP/us...@${ip2}:${port2})
for dialing user3
username=user3 and extension =112233
exten= 112233,1,NoOp()
exten= 112233,n,MYSQL(Connect connid 'localhost' cdr dbpass cloneline)
exten= 112233,n,NoOP(Connection ID:${connid})
exten= 112233,n,MYSQL(Query resultid ${connid} SELECT\ ip1\, port1\,
ip2\, port2\, status\ from\ users\ where\ username=user3 )
exten= 112233,n,MYSQL(Fetch fetchid ${resultid} ip1 port1 ip2 port2)
exten= 112233,n,Dial(SIP/us...@${ip1}:${port1}SIP/us...@${ip2}:${port2})
Hope every thing would be clear...
Muhammad Faheem
Software Engineer
AxVoice Inc.
307,Y Commercial,
DHA Lahore, Pakistan
+92-333-4793314
http://www.axvoice.com
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--
__At.,
_
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br mailto:@tqi.com.br
: www.tqi.com.br http://www.tqi.com.br
( + 55 (34)3291-1700
( + 55 (34)9971-2572
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Re: [asterisk-users] Multiple user registration ...
Thank you very much for all your help, Muhammad! (please let me know if I should call you Faheem, instead). I'll make some tests with this script on my premises as soon as possible. Having a look on it, I couldn't realize how it really works in conjunction with Asterisk. I mean, it seems that the line cloning is acchieved by the creation/update of a file (with a name that matches the SIP user name) inside folder /var/lib/asterisk/users. The point is that I couldn't find any similar folder on my test server, and a search on Google by this folder didn't returned any usefull results. Am I missing something here ? Suppose I want to acchieve this feature by database update. I've noticed here that it will be a problem considering that field name at sip_buddies, that is my Realtime table for SIP users, have a UNIQUE_KEY constraint. Moreover, I don't know what will happen on Realtime (probably an error or undesired behavior) that seems to be expecting just one record user record information. Have you tried database approach ? Thanks again and best regards, Mauro. Faheem escreveu: Mauro, Yes, you will receive simultaneous ring on all devices which are registered with the same SIP User Account. If a SIP user is registered on multiple devices i.e. only one SIP account is used and only one extension is used here in my implementation, then he will ring on all registered SIP enabled devices/softphones. Also I've tested it with following combinations of SIP enabled devices/Softphones. 1) Both ports of SPA2100 are registered with one SIP account(Same IP address but different ports) 2) The same SIP user is registered with one port of SAP2100 and the same user is registered with Xten (multiple IP addresses) 3) The same SIP User is registered with two different SIP Dialers. Here in these three cases I've sucessfully able to receive concurrent ring on the registered devices/softphones. Also CDR are working correctly. The perl script works perfectly with my customization, you need to modify it according to your requirements. Muhammad Faheem Software Engineer AxVoice Inc. 307,Y Commercial, DHA Lahore, Pakistan +92-333-4793314 http://www.axvoice.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple user registration ...
Outch... my bad. I saw Muhammad Faheem at the end of your email, and... well... thought the first was your name... sorry about that. Thank's a lot again, but I'm still curious about how Asterisk integrates with your secondary persistence. I mean... until now I saw only the codes regarding the persistence or multiple registration info, but I can't still realize how Asterisk perform the invitation to all these devices... Could you please explain how it works with your solution ? Thanks and best regards, Mauro. Faheem escreveu: Yes, Its my Name! Well, my DB server and asterisk servers are on different locations. For optimization I've used Files instead of Database queries. Secondly the /var/lib/asterisk/user folder is a simple folder if it does not exists on your asterisk machine then simple create it on the specified location or simply change the folder path in the perl script. Before File handling I've used Databases for maintaing active registered users with multiple IP/Ports. The attatched perl script uses database for maintain active registration. The structure of cloneline table should be. DB: Cloneline table:users(Username,IP1,Port1,Ip2,Port2) all varchars(30) Please adjust the table fields appropriately. Hope this code block will solve you problems. Muhammad Faheem Software Engineer AxVoice Inc. 307,Y Commercial, DHA Lahore, Pakistan +92-333-4793314 http://www.axvoice.com --- On *Fri, 8/28/09, Mauro Sergio Ferreira Brasil /mauro.bra...@tqi.com.br/* wrote: From: Mauro Sergio Ferreira Brasil mauro.bra...@tqi.com.br Subject: Re: [asterisk-users] Multiple user registration ... To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, August 28, 2009, 5:38 PM Thank you very much for all your help, Muhammad! (please let me know if I should call you Faheem, instead). I'll make some tests with this script on my premises as soon as possible. Having a look on it, I couldn't realize how it really works in conjunction with Asterisk. I mean, it seems that the line cloning is acchieved by the creation/update of a file (with a name that matches the SIP user name) inside folder /var/lib/asterisk/users. The point is that I couldn't find any similar folder on my test server, and a search on Google by this folder didn't returned any usefull results. Am I missing something here ? Suppose I want to acchieve this feature by database update. I've noticed here that it will be a problem considering that field name at sip_buddies, that is my Realtime table for SIP users, have a UNIQUE_KEY constraint. Moreover, I don't know what will happen on Realtime (probably an error or undesired behavior) that seems to be expecting just one record user record information. Have you tried database approach ? Thanks again and best regards, Mauro. Faheem escreveu: Mauro, Yes, you will receive simultaneous ring on all devices which are registered with the same SIP User Account. If a SIP user is registered on multiple devices i.e. only one SIP account is used and only one extension is used here in my implementation, then he will ring on all registered SIP enabled devices/softphones. Also I've tested it with following combinations of SIP enabled devices/Softphones. 1) Both ports of SPA2100 are registered with one SIP account(Same IP address but different ports) 2) The same SIP user is registered with one port of SAP2100 and the same user is registered with Xten (multiple IP addresses) 3) The same SIP User is registered with two different SIP Dialers. Here in these three cases I've sucessfully able to receive concurrent ring on the registered devices/softphones. Also CDR are working correctly. The perl script works perfectly with my customization, you need to modify it according to your requirements. Muhammad Faheem Software Engineer AxVoice Inc. 307,Y Commercial, DHA Lahore, Pakistan +92-333-4793314 http://www.axvoice.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At
Re: [asterisk-users] Multiple user registration ...
Hi Muhammad, and thanks a lot for the answer. On this moment I'm making some tests in order to collect enough information to participate of a meeting at the end of this day regarding the use of Asterisk. I won't have time to validate your contribution before this meeting and this info would be very handfull. So... could you please just clarify me if this approach you've used allows multiple SIP clients (softphone, ATA, VoIP-Celular) registrate with Asterisk using the same SIP user (like SIP/101, for example) on such way that if someone call this number all clients gets simultaneously called? Thanks and best regards, Mauro. Faheem escreveu: Dear Mauro, Your requirement seems Clone line feature for asterisk. The same question I've asked here in this group, a months later but could't get well. But actually implemented it now! It is done using AMI. Here is its basic psudo code. # ami-event.pl Connect to AMI Read the AMI Events Parse the events If it is registration Event then store the Username/IP/Ports/Technology in Database # dial plan run agi script to get all strings eg. first Device: SIP/u...@192.168.0.123:5061 second Device: SIP/u...@10.0.0.150:6060 The complete script is attached. Muhammad Faheem Software Engineer AxVoice Inc. 307,Y Commercial, DHA Lahore, Pakistan +92-333-4793314 http://www.axvoice.com http://advcomm.net/ --- On *Wed, 8/26/09, Mauro Sergio Ferreira Brasil /mauro.bra...@tqi.com.br/* wrote: From: Mauro Sergio Ferreira Brasil mauro.bra...@tqi.com.br Subject: [asterisk-users] Multiple user registration ... To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, August 26, 2009, 7:07 PM Hello there! We are planning to use Asterisk on our VoIP platform, and we are spending some brains on a way to provide the following facility: let some SIP user (extension) registrate with more than one client (ATA, SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate calls from any of this devices that are registrated with the same user - no problems on tests too -, but also receive INVITE requests on all devices if someone calls this user - yeah... here the thing gets creepy. The demand is quite simple: let a user registrate with multiple devices using the same SIP user on such way that if someone call him, all these registered devices will ring and the first to take the call will be the lucky one. The demand, as I've said, is quite simple and logical (translated to our living world), but the reality is a very different history. On our tests, always is the last registered application/device that receives the call indication. And only the last one. We are making some tests trying to kind of deceive Asterisk on second, third, and additional, registrations so it receives from Realtime fake extensions numbers on such a way that we can use all these fake extensions to build a queue dinamicaly (through ARA) and provide the desired ring on all functionality. I think this will lead us to lots of SIP sinalization and multi user registration problems, but that was the best shot we had here until now. I would like to know if anyone had the same demand and, maybe, have found any viable solution to it. Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br /mc/compose?to=mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio
Re: [asterisk-users] Realtime with rtcachefriends=no problems...
Thanks Atis, its working pretty fine now. Best regards, Mauro. Atis Lezdins escreveu: On Wed, Aug 26, 2009 at 12:11 AM, Mauro Sergio Ferreira Brasilmauro.bra...@tqi.com.br wrote: Hello there! Problem found. For some reason, the update statement below is generated with an invalid atribution of empty value '' to field port that is an integer. Because of that, this record keeps with prior fullcontact information that was updated by another client (which uses a different port) what leads to wrong client rtp packets routing... wow... that was weird... :-) [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query: UPDATE sip_buddies SET fullcontact = '', ipaddr = '', port = '', regseconds = '0', username = '', regserver = '' WHERE name = '101' [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect integer value: '' for column 'port' at row 1 First of all... my appologies by the false alarm. But now I need your help to identify why is this update statement being generated wrongly. Does someone have any idea ? Asterisk Realtime Architecutre currently treats all fields as strings. I wish too that it would take into account actual field type retrieved from DESCRIBE statement and add the quotes only if it's string. You can safely do ALTER TABLE sip_buddies CHANGE COLUMN port port VARCHAR(5); Regards, Atis -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple user registration ...
Hello there! We are planning to use Asterisk on our VoIP platform, and we are spending some brains on a way to provide the following facility: let some SIP user (extension) registrate with more than one client (ATA, SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate calls from any of this devices that are registrated with the same user - no problems on tests too -, but also receive INVITE requests on all devices if someone calls this user - yeah... here the thing gets creepy. The demand is quite simple: let a user registrate with multiple devices using the same SIP user on such way that if someone call him, all these registered devices will ring and the first to take the call will be the lucky one. The demand, as I've said, is quite simple and logical (translated to our living world), but the reality is a very different history. On our tests, always is the last registered application/device that receives the call indication. And only the last one. We are making some tests trying to kind of deceive Asterisk on second, third, and additional, registrations so it receives from Realtime fake extensions numbers on such a way that we can use all these fake extensions to build a queue dinamicaly (through ARA) and provide the desired ring on all functionality. I think this will lead us to lots of SIP sinalization and multi user registration problems, but that was the best shot we had here until now. I would like to know if anyone had the same demand and, maybe, have found any viable solution to it. Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple user registration ...
Hi Elliot, and thanks for the reply. I'm not completely sure you've considered that the SIP users registered on all devices are the same. Have you ? I mean... How will I use Dial command with a sequence of same devices, like: Dial(SIP/101SIP/101SIP/101), for example ? That's why we are testing the possibility to create virtual devices on subsequent registrations, so we can at the end make something like: Dial(SIP/101SIP/101-001SIP/101-002) if someone dials to SIP/101. Note: SIP/101-001 and SIP/101-002 don't really exist. They will be provided by our ARA driver to allow the multiple device ringing. Thanks and best regards, Mauro. Elliot Otchet escreveu: Is your goal here to have multiple devices ring when an extension is dialed and the first one to answer take the call? If so, see the Dial command Dial(Technology/resourceTechnology/resourceTechnology/resource...[|timeout][|options][|URL]). When multiple technology/resource entries are listed, the first one to answer will take the call. That accomplishes your goal, if I understand you correctly. The nice part about doing it this way (with each device independently registered) is that you gain a substantial amount of granularity in controlling where calls go and you don't have to find creative ways (read: unsupported) to trick Asterisk or endpoints. If you're developing your own GUI to have people set up their devices, you can easily create a wizard that walks them through setting up each device and associating them together through either channel variables or other tables in a database. I use this methodology in 1.4 and it works quite reliably. For a good reference, check out http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial or from your Asterisk console try: 'core show application dialenter' It's not perfect because you can have devices that do funny things with a SIP INVITE, but in most cases it works very well. Regards, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple user registration ...
Hi Barry, and thanks for the reply! This was the first question I've made on meeting yesterday to decide about this facility. Having me here today making this question should give you an idea of the level of acceptance of my suggestion :-). Anyway, the idea is really try to make it work with only one SIP user. I totally agree with you that this is an unnatural behavior, but I have to agree as well with our commercial staff because their vision was naturaly translated from our telephony world (we don't have a different ID - telephone number - to each phone we have home, right ?). So, I thank you for your handy Dial approach, which will be easier than the queue approach I was considering before. Given that I'll acchieve the virtual devices running. Considering my annoying insistence on work with just one SIP user, do you have any helpfull thoughts to share that can help me out ? Best regards, Mauro. Barry L. Kline escreveu: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Instead of trying to make Asterisk do this unnatural act, why not register each device with a separate id, then use the dial function to call all of them? e.g.exten = 122,1,Dial(SIP/1SIP/2SIP/3) You could use some creating scripting to decide which devices to ring based on the dialed extension. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKlU65CFu3bIiwtTARAu0DAJ4szfX1dp/BNZojIKhgIL/tIhkjvQCeLXCf A+Dys6+LgrNhL/zQpU8Vuwk= =1Y6q -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple user registration ...
Thanks again Elliot for everything. Considering our needs to develop a proprietary ARA driver, I think it's possible to use it and make Asterisk believe that additional registrations of same SIP device are in fact different device registrations. BUT, and yes it's a big BUT, we will end with an Asterisk version a litle hacked and even if we get this working on some version now, it doesn't give us any guarantee that it will in future. Anyway, I've put this question here just to be sure no one has already made such a thing before, and how odd is it. I'll take care and don't use - on sip device names... thanks for that too. Best regards, Mauro. Elliot Otchet escreveu: Your first example illustrates why having multiple devices registered as the same entity is a bad idea. It is impossible to differentiate between each device when you have multiple registering as the same entity. My users also really like setting up rules per device/per caller. When you treat a group of devices as one, you make it really hard to do that. On your theoretical virtual devices in Asterisk - you either have a device or you don't. The device will need to register in order to receive a call, so if you're expecting to do some magic on the registration to have a user who registers with the credentials of user 101 and be assigned to user 101-001, you'll be disappointed in the results. Also, you'll want to steer away from using hyphens in your sip device names. Hyphens are used in the SIP channel driver for a special purpose and using them in your device names may cause problems. See http://www.digium.com/handbook-draft.pdf page 19 for more info. If you're looking for a good separator, try using the underscore (_) character instead. All that being said, if you want to register multiple devices with a single set of credentials, you might want to check out a SIP Proxy instead of Asterisk's SIP B2BUA. Some can handle multiple registrations with a single set of credentials quite nicely. Regards, Elliot -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio Ferreira Brasil Sent: Wednesday, August 26, 2009 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple user registration ... Hi Elliot, and thanks for the reply. I'm not completely sure you've considered that the SIP users registered on all devices are the same. Have you ? I mean... How will I use Dial command with a sequence of same devices, like: Dial(SIP/101SIP/101SIP/101), for example ? That's why we are testing the possibility to create virtual devices on subsequent registrations, so we can at the end make something like: Dial(SIP/101SIP/101-001SIP/101-002) if someone dials to SIP/101. Note: SIP/101-001 and SIP/101-002 don't really exist. They will be provided by our ARA driver to allow the multiple device ringing. Thanks and best regards, Mauro. Elliot Otchet escreveu: Is your goal here to have multiple devices ring when an extension is dialed and the first one to answer take the call? If so, see the Dial command Dial(Technology/resourceTechnology/resourceTechnology/resource...[|timeout][|options][|URL]). When multiple technology/resource entries are listed, the first one to answer will take the call. That accomplishes your goal, if I understand you correctly. The nice part about doing it this way (with each device independently registered) is that you gain a substantial amount of granularity in controlling where calls go and you don't have to find creative ways (read: unsupported) to trick Asterisk or endpoints. If you're developing your own GUI to have people set up their devices, you can easily create a wizard that walks them through setting up each device and associating them together through either channel variables or other tables in a database. I use this methodology in 1.4 and it works quite reliably. For a good reference, check out http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial or from your Asterisk console try: 'core show application dialenter' It's not perfect because you can have devices that do funny things with a SIP INVITE, but in most cases it works very well. Regards, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971
Re: [asterisk-users] Multiple user registration ...
Thanks again Barry for the help and attention. Thanks for wishing me lucky as well... If we insist on this road I'll need it for sure :-). I can't agree more with your position, and I'll try to be sure our commercial demands can't be acchieved with normal approaches before adventuring on such path. Best regards, Mauro. Barry L. Kline escreveu: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Well, our phones at home are probably analog and can be connected in parallel. Unfortunately, VoIP phones are a different matter and need to be identified individually. I guess I don't get the problem your commercial side is having with this concept. You can produce the same result doing things within the constraints of SIP using the features built into Asterisk. Doing what you want may be possible with a bunch of contortions, but it's going to be an unnatural act fraught with tons of unexpected behavior. If you do get it working the way you describe you'll likely be doing so because of a side-effect behavior in a GIVEN version of Asterisk. The moment you change versions, the side effect may or may not be the same and you may find yourself in the same trouble. I can't offer anything more to help you except to wish you the best of luck. You're going to need it. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKlWonCFu3bIiwtTARAu1WAJ0eS2Eh6n6Tici9eDA82UIesuozNACaA9yi jT8u2aZfUHcSXGvJnc1FDEI= =VQhJ -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime with rtcachefriends=no problems...
Hello there! I was testing Asterisk for the last two weeks using the Realtime driver for MySQL, and leaving rtcachefriends=yes configured to enable MWI. Today I started making additional tests with rtcachefriends=no because we will probably need to use Asterisk without this cache. For some strange reason, calls stop to get routed between the SIP clients. I've registered successfuly with two sip clients as usual, but the call indication that I have on originator client (call in progress) don't match with the target client that indicates nothing at all. Using Wireshark I could see lots of ICMP errors being returned from the target machine with Destination Unreachable/Port Unreachable indications. And this happens on both ways, client 1 calling client 2 and vice-versa. I switched back to rtcachefriends=yes and all worked fine again. (note: always I change rtcachefriends to no, I change qualify parameter of all SIP users to no as well - to avoid warnings on CLI). Does anyone had this problem ? What Am I missing here ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime with rtcachefriends=no problems...
Hello there! Problem found. For some reason, the update statement below is generated with an invalid atribution of empty value '' to field port that is an integer. Because of that, this record keeps with prior fullcontact information that was updated by another client (which uses a different port) what leads to wrong client rtp packets routing... wow... that was weird... :-) [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query: UPDATE sip_buddies SET fullcontact = '', ipaddr = '', port = '', regseconds = '0', username = '', regserver = '' WHERE name = '101' [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect integer value: '' for column 'port' at row 1 First of all... my appologies by the false alarm. But now I need your help to identify why is this update statement being generated wrongly. Does someone have any idea ? Thanks and best regards, Mauro. Mauro Sergio Ferreira Brasil escreveu: Hello there! I was testing Asterisk for the last two weeks using the Realtime driver for MySQL, and leaving rtcachefriends=yes configured to enable MWI. Today I started making additional tests with rtcachefriends=no because we will probably need to use Asterisk without this cache. For some strange reason, calls stop to get routed between the SIP clients. I've registered successfuly with two sip clients as usual, but the call indication that I have on originator client (call in progress) don't match with the target client that indicates nothing at all. Using Wireshark I could see lots of ICMP errors being returned from the target machine with Destination Unreachable/Port Unreachable indications. And this happens on both ways, client 1 calling client 2 and vice-versa. I switched back to rtcachefriends=yes and all worked fine again. (note: always I change rtcachefriends to no, I change qualify parameter of all SIP users to no as well - to avoid warnings on CLI). Does anyone had this problem ? What Am I missing here ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call routing between two Asterisk boxes using SIP not working ...
Hello there! I need some help to configure two Asterix boxes to route calls using SIP. I followed the instructions present at this site: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html;, but I couldn't get it working so far. The only difference, besides the names that I've used, is that I'm using realtime to retrieve all information. Both boxes registrate on the other perfectly. The problem happens when one call gets routed. It seems that realtime on destination box is trying to find locally a SIP user 1001 that is the originator of the call and is a user of the original box. It finally ends with a: chan_sip.c:14780 handle_request_invite: Failed to authenticate user 1001 sip:1...@10.10.100.158;tag=as1e79b629 on destination box. Wireshark present on destination box indicates all the following steps: 1- Wengo client registered with user 1001 starts the call to number 2001 with Box 1 (at 10.10.100.158); 2- Box 1 makes the challenge; 3- Wengo replies the challenge; 4- Box 1 send an successfull ack to Wengo client and sends the INVITE to Box 2 (at 10.10.100.156) that holds user 2001; 5- Box 2 makes the challenge; 6- Box 1 replies the challenge; 7- Box 2 sends a 403 Forbidden; Has anyone had this problem ? Can anyone help me out on that ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call routing between two Asterisk boxes using SIP not working ...
Hi guys! The problem was solved by the use of same password for registration users of both boxes. Is there no way to indicate different password for registration user of Box1 and registration user of Box2 ? Thanks and best regards, Mauro. Mauro Sergio Ferreira Brasil escreveu: Hello there! I need some help to configure two Asterix boxes to route calls using SIP. I followed the instructions present at this site: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html;, but I couldn't get it working so far. The only difference, besides the names that I've used, is that I'm using realtime to retrieve all information. Both boxes registrate on the other perfectly. The problem happens when one call gets routed. It seems that realtime on destination box is trying to find locally a SIP user 1001 that is the originator of the call and is a user of the original box. It finally ends with a: chan_sip.c:14780 handle_request_invite: Failed to authenticate user 1001 sip:1...@10.10.100.158;tag=as1e79b629 on destination box. Wireshark present on destination box indicates all the following steps: 1- Wengo client registered with user 1001 starts the call to number 2001 with Box 1 (at 10.10.100.158); 2- Box 1 makes the challenge; 3- Wengo replies the challenge; 4- Box 1 send an successfull ack to Wengo client and sends the INVITE to Box 2 (at 10.10.100.156) that holds user 2001; 5- Box 2 makes the challenge; 6- Box 1 replies the challenge; 7- Box 2 sends a 403 Forbidden; Has anyone had this problem ? Can anyone help me out on that ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Platform decision ...
Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ;): The main points listed on Asterisk's CONS that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT traversal; * Asterisk doesn't use SpanDSP; * Use of no longer maintained Berkeley DB1 engine as its internal database; * Asterisk doesn't allow CSRC entries in RTP; * Asterisk doesn't have an universal jitterbuffer for use with any channel type; * Asterisk doesn't use POSIX realtime extensions (having dependency with Zaptel timing); We were considering Asterisk as the chosen platform, but after reading this I got a little worried. The comparison considers 1.4 old version of Asterisk. So, can someone give me an update on what have changed for this items considering new 1.6 version ? Maybe someone can point me a site with an updated comparison. As long as I could see by now SpanDSP is present on new version of Asterisk, so this item isn't a difference any more. Right ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Platform decision ...
Man... I need to be very frank with you... I don't know any more. We started analysing what can be done to get Asterisk working on a way we want it to work, that is: totally dynamic dial plan generated by an external server (responsible for business logic and legacy interface), and retrieved through an new configuration driver (something like res_config_legacy.c). This point is clear to us now that is reachable without much effort. We considered, at first, a infraestructure with a redirect-server/load-balancer (played by OpenSIPS) directing the voip calls to final Asterisk instances. The problem is that after getting the first issue solved (about the driver acessing the legacy interface explained above), I started a research about Asterisk scalability and I didn't liked of what I found. Consulting some friends of mine that work with Voip (but that unfortunatelly don't need the PBX features) the impression was worst. One of them told me that on the only part of their infraestructure where Asterisk is used they want at all costs to remove it. Making things short, I need to have sure that Asterisk can handle a considerable number of concurrent calls, or I need an indication of another PBX that is scalable to be placed on Asterisk's place and that can be changed to retrieve the dialplan (or what it uses on call routing) from another server. Does anyone have any idea ? Thanks and best regards, Mauro. C. Savinovich escreveu: It all depends what are you going to use Asterisk for. Sounds like it is for conferencing. Would you care to elaborate? CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio Ferreira Brasil Sent: Tuesday, August 18, 2009 10:23 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Platform decision ... Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ;): The main points listed on Asterisk's CONS that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT traversal; * Asterisk doesn't use SpanDSP; * Use of no longer maintained Berkeley DB1 engine as its internal database; * Asterisk doesn't allow CSRC entries in RTP; * Asterisk doesn't have an universal jitterbuffer for use with any channel type; * Asterisk doesn't use POSIX realtime extensions (having dependency with Zaptel timing); We were considering Asterisk as the chosen platform, but after reading this I got a little worried. The comparison considers 1.4 old version of Asterisk. So, can someone give me an update on what have changed for this items considering new 1.6 version ? Maybe someone can point me a site with an updated comparison. As long as I could see by now SpanDSP is present on new version of Asterisk, so this item isn't a difference any more. Right ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
