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
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
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
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
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
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
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
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
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
