[asterisk-users] Asterisk Configuration GUI Question
Hi All, There are a lot of existing projects for configuring Asterisk via GUI, so instead of trudging through them all, I'm hopeing to get some guidance. My architecture is ITSP based, we supply hosted PBX's to business customers. A few systems are dedicated PBX's but the majority are virtualized instances. We have been very successful managing the systems for our customers, not a lot of request for user portals or anything like that, so our PBX management consist of command line editing of Asterisk flat files and minimal sql database routines. We have built a few custom user portals for some of our customers using LAMP and have deployed a couple of other user web utilities, CDR search, Operator panel, Queue stats. I would like to implement a more standardized user portal for basic functions like call forward, voicemail password reset, user info change, queue member add/remove, ect I know there are many projects that could do just that, but most of what I'm finding are GUI's that take over the system and have conventions for many more configurable elements than I really need. Most are overkill for what I'm looking for. Because the majority of my PBX's are hosted virtual systems, overhead must be light. I would like to have a centralized management portal that pushes configs out to the PBX's but I'm not apposed to running a GUI on each PBX instance as long as it is light. I would like to be able to customize the interface, brand with my business logos, add or remove configuration elements. I kind of like the Digium Asterisk GUI but I'm just not real familiar with it, just test driving it a bit. What I do like about it is the flat file manipulation, no database needed. Any guidance is much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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] Asterisk Configuration with Sphinx speech engine
hi, i am using asterisk 1.6 and i want to integrate sphinx speech engine with my asterisk, so that i can use the generic speech API provided by asterisk 1.6... Plz help me, how can i do that... any help will be highly appreciated... waiting for your positive response... Thanks Regards,Rizwan HasnaniFinal Year Student - NUCES-FASTEmail id: k060...@nu.edu.pkcell#: 0345-3235008 _ Windows 7: Unclutter your desktop. Learn more. http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7secslideid=1media=aero-shake-7secondlistid=1stop=1ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_7secdemo:122009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Configuration with Sphinx speech engine
Hello I also tried it in begining but cant give time to it. So no success. you can try this link http://www.voip-info.org/wiki/view/Sphinx http://cmusphinx.sourceforge.net/html/cmusphinx.php http://cmusphinx.sourceforge.net/html/cmusphinx.php hope this helps On Tue, Dec 1, 2009 at 12:16 PM, Rizwan Hasnani rizwanhasn...@hotmail.comwrote: hi, i am using asterisk 1.6 and i want to integrate sphinx speech engine with my asterisk, so that i can use the generic speech API provided by asterisk 1.6... Plz help me, how can i do that... any help will be highly appreciated... waiting for your positive response... Thanks Regards, Rizwan Hasnani Final Year Student - NUCES-FAST Email id: k060...@nu.edu.pk Cell#: 0345-3235008 -- Windows 7: Unclutter your desktop. Learn more.http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7secslideid=1media=aero-shake-7secondlistid=1stop=1ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_7secdemo:122009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk configuration
Hello,
I have installed iaxmodem(1.1.1) and hylafax(4.4.4) but my Asterisk(1.2)
configuration seems to be wrong (calls are destroyed by asterisk).
Does someone know how to configure Asterisk (iax.conf, sip.conf and
extension.conf)???
This is my configuration of iaxmodem:
device /dev/ttyIAX0
owner uucp:uucp
mode 660
port 4570 #each line should have it's own port number!
refresh 300
server 127.0.0.1
peername IAXmodem #this is the local extension number in FreePBX (create
it!) secret 12345 #password for the
extension
cidname Fax1
cidnumber
codec ulaw
My iax.conf is :
[iaxmodem]
type=friend
host=127.0.0.1
secret=x
context=fax-out
permit=127.0.0.1
disallow=all
allow=ulaw
My sip.conf is:
[123456]
type=friend
insecure=very
nat=yes
username=123456
fromuser=12345
secret=
host=127.0.0.1
qualify=yes
context=fax-in
My extension.conf is:
[fax-in]
Exten = _900.,1Dial(IAX2/iaxmodem)
[fax-out]
Exten =_900.,1,Dial(SIP/12345/${EXTEN})
I hope someone could help me!
Thanks!
Nadjia Boumediene, Legos
[EMAIL PROTECTED]
+33172292995
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[asterisk-users] Asterisk configuration for T1 CAS lines
Hi, I am trying to use Asterisk PBX with T1 CAS. The setup that I am looking for is as below Analog phones == Asterisk T1 CAS === Integrated Access Device IP Network for VoIP. The Asterisk has a T1 card and I want a CAS config between Asterisk and T1 port of IAD. The Asterisk has got a FXS card to which the analog phones are connected. I would like to know whether T1 CAS configuration is possible with Asterisk. If possible, any pointers to configuration would be really helpful. Thanks and Regards, Jana ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Configuration File Parser
Hi all, I would need to parse asterisk configuration files with PHP code. Does anyone know if one already exist? Thanks in advance Yann JOUANIN ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Configuration File Parser
On Mon, Aug 13, 2007 at 10:34:56AM +0200, [EMAIL PROTECTED] wrote: I would need to parse asterisk configuration files with PHP code. Does anyone know if one already exist? Parse? In what way? What information do you want to extract? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk configuration directly with Mandi (Speechphone)
Has anyone set up Speechphone (Mandi) directly with Asterisk and not used an ATA? If so, could you share how you did it? TIA ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Configuration Complete Newbie question
Hello Am starting on my Asterisk journey - am getting a single span Digium card to connect Asterisk to our Alcatel 4400 EPABX and install about 100 VoIP instruments. The Asterisk VoIP extensions and Alcatel digital extensions have to talk to each other. Am I right in understanding that IN ASTERRISK : I have to create a config with either all Asterisk and Alcatel extensions - which config files? extensions.conf for both with the two types of extensions in different contexts? Would that be the correct way? IN ALCATEL : List of Asterisk extensions and the PRI card to which the calls have to be delivered. Is that broadly correct? Thanks very much Best wishes Iyer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Configuration Complete Newbie question
On 10/6/06, K Y Iyer [EMAIL PROTECTED] wrote: Is that broadly correct? Yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Configuration Complete Newbie question
Title: RE: [asterisk-users] Asterisk Configuration Complete Newbie question Thanks very much - let me see how far I can take it now. Best wishes Iyer -Original Message- From: [EMAIL PROTECTED] on behalf of Lacy Moore - Aspendora Sent: Fri 10/6/2006 03:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Configuration Complete Newbie question On 10/6/06, K Y Iyer [EMAIL PROTECTED] wrote: Is that broadly correct? Yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Configuration
Hi All I am new member to asterisk mailing list. I have complied the asterisk and it is running fine. I have configured two extensions in extensions.conf exten = 228,1,Dial exten = 234,1,Dial and configured the xlite soft phone. when I am calling from 234 to 228 it is unable to establish the call. I am able to here all automated playback IVR. ex.500, 600 can any one help to configure the inbound / outbound calls and how to add sip users. -Linga Reddy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Configuration
Hello Linga, you could download and install the ESCAUX net.PBX Free Edition and create whatever device or call flow you want with the web interfaces. Afterwards, in the /etc/asterisk/ directory, take a look at the generated configuration files (gen_sip.conf, gen_extensions.conf, profiles.conf, ...). This might help you to learn and understand how asterisk works. Download here: http://www.escaux.com Cheers, Jordi -- Jordi Nelissen ESCAUX Business IP Telephony www.escaux.com R.Linga Reddy wrote: Hi All I am new member to asterisk mailing list. I have complied the asterisk and it is running fine. I have configured two extensions in extensions.conf exten = 228,1,Dial exten = 234,1,Dial and configured the xlite soft phone. when I am calling from 234 to 228 it is unable to establish the call. I am able to here all automated playback IVR. ex.500, 600 can any one help to configure the inbound / outbound calls and how to add sip users. -Linga Reddy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Configuration
Hi All I am new member to asterisk mailing list. I have complied the asterisk and it is running fine. I have configured two extensions in extensions.conf exten = 228,1,Dial exten = 234,1,Dial and configured the xlite soft phone. when I am calling from 234 to 228 it is unable to establish the call. I am able to here all automated playback IVR. ex.500, 600 can any one help to configure the inbound / outbound calls and how to add sip users. -Linga Reddy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Configuration
Hi: First at all: You SIP phones are right register on sip.conf file? Cris From: R.Linga Reddy [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Configuration Date: Wed, 09 Aug 2006 19:40:50 +0530 Hi All I am new member to asterisk mailing list. I have complied the asterisk and it is running fine. I have configured two extensions in extensions.conf exten = 228,1,Dial exten = 234,1,Dial and configured the xlite soft phone. when I am calling from 234 to 228 it is unable to establish the call. I am able to here all automated playback IVR. ex.500, 600 can any one help to configure the inbound / outbound calls and how to add sip users. -Linga Reddy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Configuration
You need to tell asterisk what to dial. Check the dial command syntax and probably the sip.conf file.On 8/9/06, R.Linga Reddy [EMAIL PROTECTED] wrote:HiAllI am new member to asterisk mailing list. I have complied the asterisk and it is running fine.I have configuredtwo extensions in extensions.confexten = 228,1,Dialexten = 234,1,Dialand configured the xlite soft phone. when I am calling from 234 to 228 it is unable to establish the call.I am able to here all automated playback IVR. ex.500, 600can any one help to configure the inbound / outbound calls and how toadd sip users.-Linga Reddy ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk configuration for h323 calls
Hello,I'm new to Asterisk. I want to do the folloing job.I want to run Asterisk as a voip gateway to forward h323 calls to another gateway. my-gateway - Asterisk -- your-gateway h323 h323Is it possible to do this? If so, can anyone give me an idea how to do it? How many configuration files relates to this job? Can you give a sample configuration? Thank yo u in advance.Roda Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk configuration using Database..!
Hi all, I want to configure Asterisk using Mysql Database. But on compilation of asterisk-addons I am getting some errors. I have pasted the errors in the pastebin. Please checkout this link. http://pastebin.com/499106 Also please do let me know which are packages required for compiling the asterisk-addons? Does the asterisk version also make a difference in configuration of Asterisk using database? Which version of Asterisk supports configuration using Mysql? Thanks and Regards, Bharat Sarvan The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Configuration
I have installed Redhat Linux 9 and Asterisk 1.2.1 on new computer. I need to know initial configuration of Asterisk i.e How to register a sip user?. What files do I have to edit? I am new about the Asterisk please help me Faheem Ahmed ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Configuration
Here is where you will find the answer to all of your questions: http://www.asterisk.org/ http://www.voip-info.org/wiki-Asterisk Jonathan On Sat, 2005-12-24 at 01:34 +0500, Faheem Ahmed wrote: I have installed Redhat Linux 9 and Asterisk 1.2.1 on new computer. I need to know initial configuration of Asterisk i.e How to register a sip user?. What files do I have to edit? I am new about the Asterisk please help me Faheem Ahmed ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk configuration from database with res_config
I want to let Asterisk read its configuration from a mysql database. I configured everything according to the wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+res_config. However it doesn't work. I am using 1.0.8 asterisk version and here are my config files: Extconfig.conf: [settings] ;uncomment to load queues.conf via the db engine. ;queues.conf = odbc,mysql1,ast_config ;extensions.conf = odbc,mysql1,ast_config sip.conf = odbc,mysql1,ast_config res_odbc.conf: ;;; odbc setup file [mysql1] dsn = MySQL-asterisk username = blaat password = blaat pre-connect = yes [mysql2] dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes odbc to mysql is working fine, I tested it. here is my odbc.ini from /etc/ [MySQL-asterisk] Description = MySQL Asterisk database Trace = Off TraceFile = stderr Driver = MySQL SERVER = localhost USER= blaat PASSWORD= blaat PORT= 3306 DATABASE= asterisk I used the load_res_config.pl to put the sip.conf into the database in ast_config. Via phpMyadmin I can see the data in there correctly. When booting or reloading Asterisk I don't see anything indicating it is connecting to odbc. I tried removing the sip.conf from /etc/asterisk, leaving an empty sip.conf, and only leaving the general section of sip.conf there. Nothing works. Cheers, Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk configuration from database withres_config
is your Asterisk compiled from cvs head? Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Frank Aartman Sent: Thursday, August 18, 2005 3:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk configuration from database withres_config I want to let Asterisk read its configuration from a mysql database. I configured everything according to the wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+res_config. However it doesn't work. I am using 1.0.8 asterisk version and here are my config files: Extconfig.conf: [settings] ;uncomment to load queues.conf via the db engine. ;queues.conf = odbc,mysql1,ast_config ;extensions.conf = odbc,mysql1,ast_config sip.conf = odbc,mysql1,ast_config res_odbc.conf: ;;; odbc setup file [mysql1] dsn = MySQL-asterisk username = blaat password = blaat pre-connect = yes [mysql2] dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes odbc to mysql is working fine, I tested it. here is my odbc.ini from /etc/ [MySQL-asterisk] Description = MySQL Asterisk database Trace = Off TraceFile = stderr Driver = MySQL SERVER = localhost USER= blaat PASSWORD= blaat PORT= 3306 DATABASE= asterisk I used the load_res_config.pl to put the sip.conf into the database in ast_config. Via phpMyadmin I can see the data in there correctly. When booting or reloading Asterisk I don't see anything indicating it is connecting to odbc. I tried removing the sip.conf from /etc/asterisk, leaving an empty sip.conf, and only leaving the general section of sip.conf there. Nothing works. Cheers, Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users attachment: winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk configuration from database
Nope, I got the stable 1.08 release from cvs. Frank From: Wei Kun [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk configuration from database withres_config To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii is your Asterisk compiled from cvs head? Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Frank Aartman Sent: Thursday, August 18, 2005 3:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk configuration from database withres_config I want to let Asterisk read its configuration from a mysql database. I configured everything according to the wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+res_config. However it doesn't work. I am using 1.0.8 asterisk version and here are my config files: Extconfig.conf: [settings] ;uncomment to load queues.conf via the db engine. ;queues.conf = odbc,mysql1,ast_config ;extensions.conf = odbc,mysql1,ast_config sip.conf = odbc,mysql1,ast_config res_odbc.conf: ;;; odbc setup file [mysql1] dsn = MySQL-asterisk username = blaat password = blaat pre-connect = yes [mysql2] dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes odbc to mysql is working fine, I tested it. here is my odbc.ini from /etc/ [MySQL-asterisk] Description = MySQL Asterisk database Trace = Off TraceFile = stderr Driver = MySQL SERVER = localhost USER= blaat PASSWORD= blaat PORT= 3306 DATABASE= asterisk I used the load_res_config.pl to put the sip.conf into the database in ast_config. Via phpMyadmin I can see the data in there correctly. When booting or reloading Asterisk I don't see anything indicating it is connecting to odbc. I tried removing the sip.conf from /etc/asterisk, leaving an empty sip.conf, and only leaving the general section of sip.conf there. Nothing works. Cheers, Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk configuration from database with res_config
That wiki page is old, ugly and out of date. There are many like it and if I only knew how to delete wiki pages, I would clean it up some. The easiest way, Frank, to do what you want is to download CVS-HEAD and use ARA to store your config files. Also download addons from HEAD and you can use the native mysql realtime driver. http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime -Matthew Frank Aartman wrote: I want to let Asterisk read its configuration from a mysql database. I configured everything according to the wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+res_config. However it doesn't work. I am using 1.0.8 asterisk version and here are my config files: Extconfig.conf: [settings] ;uncomment to load queues.conf via the db engine. ;queues.conf = odbc,mysql1,ast_config ;extensions.conf = odbc,mysql1,ast_config sip.conf = odbc,mysql1,ast_config res_odbc.conf: ;;; odbc setup file [mysql1] dsn = MySQL-asterisk username = blaat password = blaat pre-connect = yes [mysql2] dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes odbc to mysql is working fine, I tested it. here is my odbc.ini from /etc/ [MySQL-asterisk] Description = MySQL Asterisk database Trace = Off TraceFile = stderr Driver = MySQL SERVER = localhost USER= blaat PASSWORD= blaat PORT= 3306 DATABASE= asterisk I used the load_res_config.pl to put the sip.conf into the database in ast_config. Via phpMyadmin I can see the data in there correctly. When booting or reloading Asterisk I don't see anything indicating it is connecting to odbc. I tried removing the sip.conf from /etc/asterisk, leaving an empty sip.conf, and only leaving the general section of sip.conf there. Nothing works. Cheers, Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Configuration
Dear users, I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. Regards, Afzaal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Configuration
On Monday 25 July 2005 14:07, Afzaal Mirza wrote: I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. http://www.voip-info.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Configuration
Hi Afzall, i am also still a beginner on *. A made best experience with the * wiki on http://www.voip-info.org/wiki-Asterisk. Maybe you start with the introduction part. Afzaal Mirza wrote: Dear users, I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. Regards, Afzaal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Configuration
I believe that this can help you on your question. http://www.automated.it/guidetoasterisk.htm#_Toc49248757 Regards, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christoph Eicke Sent: Monday, July 25, 2005 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Configuration On Monday 25 July 2005 14:07, Afzaal Mirza wrote: I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. http://www.voip-info.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Configuration
Get [EMAIL PROTECTED] here: http://asteriskathome.sourceforge.net/ This should be the EASIEST first time install out there. Once you get familiar/comfortable, consider building your own following steps at http://www.automated.it/guidetoasterisk.htm -Original Message- From: Afzaal Mirza [mailto:[EMAIL PROTECTED] Sent: Monday, July 25, 2005 7:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Configuration Dear users, I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. Regards, Afzaal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Configuration
Afzaal Mirza wrote: Dear users, I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. Regards, Afzaal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, You can find basic information from my blog http://linuxpower.blogspot.com I'm making a visual guide using flash and next week I'll post on my blog, if you have a question , ask me. Asterisk basic configuration http://linuxpower.blogspot.com/2005/07/asterisk-basic-configurations.html Cheers, ~Madhawa Blog http://linuxpower.blogspot.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, Configuration of SDP in SIP messages
Hello everybody, I'm newto Asterisk and I'm trying to configure the SIP side. I use Asterisk under the following configuration: SIP Proxy INTERNET | NAT FIREWALL | Asterisk SIP Phone A I'm trying to put a call from SIP Phone A through Asterisk to the SIP Proxy. I'm able to deliver messages to SIP Proxy. However, I have noticed that the port used to deliver the audio changes randomly. I would like to fix that to a specific range of ports so that I can tell to NAT Firewall to port forward these particalar ports to Asterisk. I have searched on documentation and the only thing that I found was how to change the SIP port but not the media port. Has anybody any ideas on how to solve that problem or where to look for a solution? Regards, Alex.Help STOP spam with the new MSN 8 and get 2 months FREE* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk, Configuration of SDP in SIP messages
This is done in the rtp.conf file. You specify the port range with a start and end number. By default the range is 1 through 2. Leif. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alexander Simeonidis Sent: Thursday, May 13, 2004 10:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk, Configuration of SDP in SIP messages Hello everybody, I'm new to Asterisk and I'm trying to configure the SIP side. I use Asterisk under the following configuration: SIP Proxy INTERNET | NAT FIREWALL | Asterisk SIP Phone A I'm trying to put a call from SIP Phone A through Asterisk to the SIP Proxy. I'm able to deliver messages to SIP Proxy. However, I have noticed that the port used to deliver the audio changes randomly. I would like to fix that to a specific range of ports so that I can tell to NAT Firewall to port forward these particalar ports to Asterisk. I have searched on documentation and the only thing that I found was how to change the SIP port but not the media port. Has anybody any ideas on how to solve that problem or where to look for a solution? Regards, Alex. Help STOP spam with the new MSN 8 http://g.msn.com/8HMAEN/2731??PS=47575 and get 2 months FREE* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, Configuration of SDP in SIP messages
Alex, The media ports are configured in rtp.conf. Also, note that Asterisk sends RTP packets out the same ports it expects them to return on. This has the effect of creating a NAT mapping for that 5-tuple, as well as opening a hole in your firewall (naturally, YMMV depending on exactly what you're running for a firewall). One interesting consequence of the way Asterisk works is that if you don't have anything behind the NAT/Firewall that's generating RTP packets (ie, no audio) no hole gets made and incoming packets will get rejected. I recently ran into an interesting problem with two SIP phones trying to talk through Asterisk behind a (non-NAT) firewall. The problem was both phones were sending RTP to the Asterisk box but the firewall was blocking both RTP streams because Asterisk never sent any RTP out those ports. And the reason Asterisk hadn't sent RTP out those ports was because it was waiting for RTP from each of the two SIP phones. This was the classic chicken-and-egg scenario. I resolved it by opening up the firewall for the range of ports I had configured Asterisk to use for RTP. A better solution would be fore Asterisk to always send a starter RTP packet so that it can ensure that the firewall opens up. -brian Alexander Simeonidis wrote: Hello everybody, I'm new to Asterisk and I'm trying to configure the SIP side. I use Asterisk under the following configuration: SIP Proxy INTERNET | NAT FIREWALL | Asterisk SIP Phone A I'm trying to put a call from SIP Phone A through Asterisk to the SIP Proxy. I'm able to deliver messages to SIP Proxy. However, I have noticed that the port used to deliver the audio changes randomly. I would like to fix that to a specific range of ports so that I can tell to NAT Firewall to port forward these particalar ports to Asterisk. I have searched on documentation and the only thing that I found was how to change the SIP port but not the media port. Has anybody any ideas on how to solve that problem or where to look for a solution? Regards, Alex. Help STOP spam with the new MSN 8 http://g.msn.com/8HMAEN/2731??PS=47575 and get 2 months FREE*___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk configuration inside a DMZ w/SIP
Brian D'Arcy wrote: Hello all, Im having a nightmare of a time trying to get stable results with SIP clients on Asterisk. I cant seem to find a configuration that works! In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall. We've discovered that certain versions of the sonic wall products do strange things with SIP. For example the TC170 with standard firmware works fine (Public Asterisk, Polycom IP600 behind the Sonic wall). Upgrade that box to the enhanced version and suddenly transfer and hold stop working. It's not just SIP, either. SNTP on the IP600 through the Sonic Wall gear changes the time by 10 hours. These things have been reported to Sonic Wall, but no word on a patch. -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk configuration inside a DMZ w/SIP
Hi Russ, Thanks for your feedback! I hadn't received any other responses from anyone, so I was starting to worry that I was one of the few having these erratic issues. I might ping Sonicwall, being a good customer and all, maybe I can get some information out of them. I've always liked using the sonicwall for ease of use and administration (and reliability), since I'm overworked as it is, but if I have to get rid of it to make this work, I'm not against it. On a side note, I tried IAX2 last night for the first time using IAXPHONE. HOLY CRAP I'M IMPRESSED!!! Everything just *works*, period. I might just use softphones until IAX hardphones are released and say screw SIP. If anyone else is having SIP nightmares and you have a flexible deployment schedule, I highly recommend giving IAX a shot!! Thanks again for the comments, Russ. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre, P.E. Sent: Saturday, April 24, 2004 5:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk configuration inside a DMZ w/SIP Brian D'Arcy wrote: Hello all, I'm having a nightmare of a time trying to get stable results with SIP clients on Asterisk. I can't seem to find a configuration that works! In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall. We've discovered that certain versions of the sonic wall products do strange things with SIP. For example the TC170 with standard firmware works fine (Public Asterisk, Polycom IP600 behind the Sonic wall). Upgrade that box to the enhanced version and suddenly transfer and hold stop working. It's not just SIP, either. SNTP on the IP600 through the Sonic Wall gear changes the time by 10 hours. These things have been reported to Sonic Wall, but no word on a patch. -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk configuration inside a DMZ w/SIP
On Saturday 24 April 2004 18:39, Brian D'Arcy wrote: Hi Russ, On a side note, I tried IAX2 last night for the first time using IAXPHONE. HOLY CRAP I'M IMPRESSED!!! Everything just *works*, period. I might just use softphones until IAX hardphones are released and say screw SIP. I'll second that :) I've been messing with KPhone simply because I use KDE and KPhone matches the rest of the eye candy, but KPhone wouldn't let me call anything outside, and even after wresting with various NAT options, I drew a blank. So I just downloaded the IaxComm binary from http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-lin-20040228.tar, ran it, configured host/user/password - and that was it- outgoing calls worked a charm, even with NAT :) $iax2++ :))) gdh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk configuration inside a DMZ w/SIP
Hello all, Im having a nightmare of a time trying to get stable results with SIP clients on Asterisk. I cant seem to find a configuration that works! In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall. Originally, I had configured Asterisk to run on the NAT side so that those within the office could connect easily, and those outside the office could connect via VPN. However the VPN route is proving to be a little too latent for quality calls. Even still, some people were able to receive audio, and others not. After much reading about Asterisk and the problems inherent to NAT, I decided OK, Ill just toss it on the DMZ with a public address, and let the clients themselves worry about addressing their NAT issues @ home, or wherever they might be. So here I am, with Asterisk running on the DMZ with a public IP address, totally unfirewalled to the outside world and now I find that not only can I not connect (from the nat side of the same SIP aware firewall hosting the asterisk server), but clients on public IPs, using no NAT at all, are either unable to connect, or are able to log in, but calls to any extension (whether they be sip extensions, voicemail, conference etc..) come up 408 timed out. In every case, the message in the * CLI is reported as: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 30841 (Response) This to me would imply that for whatever reason, the packets from the Asterisk server are being blocked by the local firewall when it attempts to send them back to me. This I can understand, because maybe Im having NAT issues myself, however I get the *same* messages broadcast into the CLI when users on the public IP addresses attempt to connect in (unfirewalled). Ive checked and triple checked to make sure that the DMZ port is not firewalled in any way, so Im a bit stumped. After this rambling, I suppose the real question Im asking here is, what is the most stable, preferred networking setup people tend to use when they are expecting to have SIP clients connecting both internally, and externally? Incase everyone wants to know about my SIP configurations, Im using disallow=all, and allow=ulaw ONLY. Ive toyed with the nat=1/nat=yes settings, however they seem to have no real effect on the behavior of the clients. Ive been testing strictly with X-Lite, as it came recommended by a few folks in #Asterisk on irc.freenode.net. [General] section from SIP.conf and an example SIP client entry: [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind SIP channel to ;externip = 216.9.32.42 ;localmask=255.255.254.0 ;localnet=192.168.0.0 context = default ; Default context for incoming calls ;srvlookup = yes [bdarcy] type=friend username=bdarcy secret=blah host=dynamic qualify=400 mailbox=3209 callerid=Brian D'Arcy 3209 nat=1 disallow=all allow=ulaw If anyone can provide any feedback on what works for you, or whats recommended, it would be highly appreciated. Thanks in advance. Brian D'Arcy
[Asterisk-Users] Asterisk Configuration + MySQL
Hi, Is there any patches to make asterisk to read all the conf from mysql instead of files? Regards, Soragan
