[Asterisk-Users] Re: Asterisk-Users Digest, Vol 15, Issue 28
Hello All Anybody had used ooH323 for asterisk i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2 audio is very good, better than SIP and IAX, but i have problem. how to router call from openh323 to outside PSTN. my h323.conf setting ; Objective System's H323 Configuration example for tvcti; ooh323c driver configuration;; [general] section defines global parameters;; This is followed by profiles which can be of three types - user/peer/friend; Name of the user profile should match with the h323id of the user device.; For peer/friend profiles, host ip address must be provided as "dynamic" is; not supported as of now.;; Syntax for specifying a H323 device in extensions.conf is; For Registered peers/friends profiles:; H323/name where name is the name of the peer/friend profile.;; For unregistered H.323 phones:; H323/ip[:port] OR if gk is used H323/alias where alias can be any H323; alias;; For dialing into another tvcti peer at a specific exten; H323/exten/peer OR H323/[EMAIL PROTECTED];; Domain name resolution is not yet supported.; ; When a H.323 user calls into tvcti, his H323ID is matched with the profile; name and context is determined to route the call;; The channel driver will register all global aliases and aliases defined in ; peer profiles with the gatekeeper, if one exists. So, that when someone; outside our pbx (non-user) calls an extension, gatekeeper will route that ; call to our tvcti box, from where it will be routed as per dial plan. [general];Define the asetrisk server h323 endpoint ;The port tvcti should listen for incoming H323 connections.;Default - 1720;port=1720 ;The dotted IP address tvcti should listen on for incoming H323;connections;Default - tries to find out local ip address on it's ownbindaddr=192.168.22.224 ;Whether tvcti should use fast-start and tunneling for H323 connections.;Default - yes;faststart=no;h245tunneling=nofaststart=yesh245tunneling=yes ;H323-ID to be used for tvcti server;Default - tvcti PBX;h323id=ObjSystvcti ;e164=100 h323id=9;e164=100;e164=0,1,2,3,4,5,6,7,8,9,*,# ;CallerID to use for calls;Default - Same as h323idcallerid=tvcti ;Whether this tvcti server will use gatekeeper.;Default - DISABLE;gatekeeper = DISCOVER;gatekeeper = a.b.c.dgatekeeper = 192.168.22.224 ;Location for H323 log file;Default - /var/log/tvcti/h323_log;logfile=/var/log/tvcti/h323_log ;Following values apply to all users/peers/friends defined below, unless;overridden within their client definition ;Sets default context all clients will be placed in.;Default - defaultcontext=from-h323 ;Sets rtptimeout for all clients, unless overridden;Default - 60 secondsrtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;Type of Service;Default - none (lowdelay, thoughput, reliability, mincost, none)tos=lowdelay ;amaflags = defaultamaflags = billing ;The account code used by default for all clients.accountcode=h3230101 ;The codecs to be used for all clients.Only ulaw and gsm supported as of now.;Default - ulaw; ONLY ulaw, gsm, g729 and g7231 supported as of nowdisallow=all ;Note order of disallow/allow is important.allow=gsmallow=ulawallow=alaw ; dtmf mode to be used by default for all clients. Only rfc2833 supported as; of now.;Default - rfc 2833;dtmfmode=rfc2833dtmfmode=inband ; User/peer/friend definitions:; User config options Peer config options; -- ---; context ; disallow disallow; allow allow; accountcode accountcode; amaflags amaflags; dtmfmode dtmfmode; rtptimeout ip; port; h323id; email; url; e164; rtptimeout ; [152]type=peercontext=from-h323;ip=a.b.c.d ; UPDATE with appropriate ip address;port=1720 ; UPDATE with appropriate port;e164=0,1,2,3,4,5,6,7,8,9,*,#e164=9911allow=ulawallow=alaw allow=gsm [myfriend11]type=friendcontext=from-h323;ip=10.0.0.82 ; UPDATE with appropriate ip address;port=1820 ; UPDATE with appropriate portdisallow=allallow=ulawe164=9rtptimeout=60dtmfmode=rfc2833 [myfriend1]type=friendcontext=from-h323;ip=10.0.0.82 ; UPDATE with appropriate ip address;port=1820 ; UPDATE with appropriate portdisallow=allallow=ulaw;e164=;h323id=tvcti_tvgk1h323id=0,1,2,3,4,5,6,7,8,9rtptimeout=60;dtmfmode=rfc2833dtmfmode=rfc2833 [abc1]type=friendcontext=from-h323disallow=allallow=ulawallow=alawh323id=9911 rtptimeout=60dtmfmode=rfc2833 how to setup prefix for gateway GNUGK?? Please help me___ --Bandwidth and Colocation sponsored by Easynews.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 PBX in Debian
Hi Please use proper quoting... See below On Sat, Oct 08, 2005 at 12:23:21AM -0400, [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, October 07, 2005 8:17 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk PBX in Debian On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote: Besides I found that using packages with asterisk on debian can do odd things to your custom sound files if you do a remove. Regarding the sounds files: I don't think that the way Asterisk installer handles them is very optimal either. Your message got me thinking, though. I believe that Debian is right installing all sounds to /usr/share/asterisk/sounds . But /var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be kept for custom sounds that are never touched by the package. I figure that file.c:build_filename could be changed to do the following: if exists /var/lib/asterisk/sounds/filename return /var/lib/asterisk/sounds/filename else if exists /usr/share/sounds/asterisk/filename return /usr/share/sounds/asterisk/filename What do you think? I figure I'll try to push this into Debian first. (If this is indeed a good idea) Using /var works, but setting it in asterisk could be a pain when it comes to voicemail prompts. Plus, extensions.conf would need to grow and become a little cluttered. Unless of course, one could do something to specify a new root voicemail path, and if the file is not found it plays from the default. You missed the point: you still keep the same configuration. filename should first be looked for in the custom directory and only afterwards in /usr/share/sounds/asterisk/sounds . voicemail/ is currently (in Debian) a symlink from the sounds directory to /var/lib/asterisk/voicemail -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
I don't know, after looking at their roadmap I don't get it. It must be the asterisk commit policies that are driving this. They have some good ideas, but they are going about this the wrong way if their goal is to create a successful fork of asterisk. Chris ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On 8 Oct 2005, at 09:49, snacktime wrote: I don't know, after looking at their roadmap I don't get it. It must be the asterisk commit policies that are driving this. They have some good ideas, but they are going about this the wrong way if their goal is to create a successful fork of asterisk. If I remember right, OpenPBX folks feel that Digium diverged from the 'right path' when they released ABE. OpenPBX is a response to that. Check on the archives around the ABE announcement to get a flavor. I'm passing _no_ judgement, except to say that some useful projects have been started for eccentric reasons. I started our SNMP stack as the result of someone saying 'Java can't do that' and I just had to prove him wrong :-) Tim. ___ --Bandwidth and Colocation sponsored by Easynews.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] IAX Help
Dear All;
Hope you are fine. I am developing an application for IAX using C#, and I have a problem sending frames to the server, I dont know how exactly I can send the frames. I have saw this site http://splurge.peoples-wireless.com/iax/ but I couldnt understand how can combine the frames. This is the code I have writtenPublic void send(string message){UdpClient udpClient = new UdpClient(); try{ udpClient.Connect(82.201.196.147, 4569); Byte[] sendBytes = Encoding.ASCII.GetBytes(message); udpClient.Send(sendBytes, sendBytes.Length); }}
Then I call send method and give it the frame to send like for example (0x0d) but I always get
this from the server
1696.476346 82.201.205.17 -
82.201.196.147 IAX2 Mini packet, source call# 17969, timestamp 8297ms, unknown (0x00)
OR
82.201.205.17 - 82.201.196.147
IAX2 Mini packet, source call# 24930, timestamp 25444ms, empty frame
If anyone can help sending an example of
what I can put in the message variable that the server can understand and how
can I combine the frames described in the link below will be great.
Thank you.
AdvancedTechnologySolutionsInternational
atsint.com
Youssef
Sayed Youssef
System Integration Manager
Tel
+202
- 6078917 ext: 807
GSM
+2010
16 35 600
Fax
+202
- 6079178
Email
[EMAIL PROTECTED]
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[Asterisk-Users] IAX Help
Dear All;
Hope you are fine. I am developing an application for IAX using C#, and I have a problem sending frames to the server, I dont know how exactly I can send the frames. I have saw this site http://splurge.peoples-wireless.com/iax/ but I couldnt understand how can combine the frames. This is the code I have writtenPublic void send(string message){UdpClient udpClient = new UdpClient(); try{ udpClient.Connect(82.201.196.147, 4569); Byte[] sendBytes = Encoding.ASCII.GetBytes(message); udpClient.Send(sendBytes, sendBytes.Length); }}
Then I call send method and give it the frame to send like for example (0x0d) but I always get
this from the server
1696.476346 82.201.205.17
- 82.201.196.147 IAX2 Mini packet, source call# 17969, timestamp 8297ms,
unknown (0x00)
OR
82.201.205.17 -
82.201.196.147 IAX2 Mini packet, source call# 24930, timestamp 25444ms, empty frame
If anyone can help sending an example of
what I can put in the message variable that the server can understand and how
can I combine the frames described in the link below will be great.
Thank you.
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[Asterisk-Users] Extension bracket matching broken in CVS
Morning all, we just download the today CVS and face a problem: in a context we want to use brackets for matching extensions like exten = _48[1-478]X.,1,Goto(validate,1) for instance. When dialing a number like 4832285 we receive == Auto fallthrough, channel 'Local/[EMAIL PROTECTED],2' status is 'UNKNOWN' Filters being our context in which we have: [filters] exten = _48[1-478]X.,1,Goto(validate,1) Changing the above context in: [filters] exten = _481X.,1,Goto(validate,1) exten = _482X.,1,Goto(validate,1) exten = _484X.,1,Goto(validate,1) exten = _487X.,1,Goto(validate,1) exten = _488X.,1,Goto(validate,1) Make it work. Does someone else faces this problem? -- Daniel ___ --Bandwidth and Colocation sponsored by Easynews.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] Extension bracket matching broken in CVS (solved)
Sorry for noise, problem is solved. It was an priority error. Administrator TOOTAI a écrit : Morning all, we just download the today CVS and face a problem: in a context we want to use brackets for matching extensions like exten = _48[1-478]X.,1,Goto(validate,1) for instance. When dialing a number like 4832285 we receive == Auto fallthrough, channel 'Local/[EMAIL PROTECTED],2' status is 'UNKNOWN' Filters being our context in which we have: [filters] exten = _48[1-478]X.,1,Goto(validate,1) Changing the above context in: [filters] exten = _481X.,1,Goto(validate,1) exten = _482X.,1,Goto(validate,1) exten = _484X.,1,Goto(validate,1) exten = _487X.,1,Goto(validate,1) exten = _488X.,1,Goto(validate,1) Make it work. Does someone else faces this problem? -- Daniel ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
Mike M wrote: On Fri, Oct 07, 2005 at 09:45:53PM -0400, Paul wrote: Also consider that there are situations where 100% open source is never allowed. Check out visa/mastercard processor certification for a good example. Digium dual licensing availability means I could actually stand a chance of using asterisk as the basis for systems used by military and law enforcement in applications that require extremely high security. There is a popular vendor of closed source products whose security has been compromised often. The security of OpenSSH is well established. Reading this list iwe learn that the open source version of Asterisk is currently being used by military personnel. Asterisk offers ways for users to implement eavesdropping applications which undermines the goal of attaining extremely high security. Open source is for sharing if that's feasible and closed source is not. Dual licensing is for both. My point was not to argue that closed source enhances security. I was just pointing out that there are situations where the customer will not accept open source. Credit card processing would be a good example. You could design *-based systems for both the client(merchant) and server(processor) functions but last I knew visa/mc would not certify open source solutions. ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On Sat, Oct 08, 2005 at 07:41:48AM -0400, Paul wrote: Credit card processing would be a good example. You could design *-based systems for both the client(merchant) and server(processor) functions but last I knew visa/mc would not certify open source solutions. Note that you can use whatever license you want for an application that connects to Asterisk via AGI or the manager interface, regardless of whether or not visa/mc would accept free software. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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] IAX Help
On Sat, Oct 08, 2005 at 11:56:40AM +0200, Youssef Sayed wrote: Dear All; Hope you are fine. I am developing an application for IAX using C#, Any reason for not using iaxclient? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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] IAX Help
Hello.
This is d Asterisk users list.
~Madhawa
Youssef Sayed wrote:
*Dear All;*
* *
* Hope you are fine. I am developing an application for IAX using
C#, and I have a problem sending frames to the server, I don’t know how exactly
I can send the frames. I have saw this site
http://splurge.peoples-wireless.com/iax/ but I couldn’t understand how can
combine the frames. This is the code I have written*
Public void send(string message)
{
UdpClient udpClient = new UdpClient();
try{
udpClient.Connect(82.201.196.147, 4569);
Byte[] sendBytes = Encoding.ASCII.GetBytes(message);
udpClient.Send(sendBytes, sendBytes.Length);
}
}
*Then I call **send** **method and give it the frame to send like for
example (**0x0d**) but I always get this from the server*
* *
*1696.476346 82.201.205.17 - 82.201.196.147 IAX2 Mini packet, source
call# 17969, timestamp 8297ms, unknown (0x00)*
* *
*OR*
* *
*82.201.205.17 - 82.201.196.147 IAX2 Mini packet, source call# 24930,
timestamp 25444ms, empty frame*
* *
*If anyone can help sending an example of what I can put in the
message variable that the server can understand and how can I combine
the frames described in the link below will be great.*
* *
Thank you.
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Re: [Asterisk-Users] Where to get the latest SIP Firmware for Polycom Phones?
thanks for that, i knew already but it misses the actual version Jesse Keating wrote: On Fri, 2005-10-07 at 11:17 +0200, Kib Eki wrote: Hello, can anybody tell me where to get the latetest SIP Firmware 1.6.2 for the Polycom phones? http://www.freedomphones.net/polycom/files/ ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
Tzafrir Cohen wrote: On Sat, Oct 08, 2005 at 07:41:48AM -0400, Paul wrote: Credit card processing would be a good example. You could design *-based systems for both the client(merchant) and server(processor) functions but last I knew visa/mc would not certify open source solutions. Note that you can use whatever license you want for an application that connects to Asterisk via AGI or the manager interface, regardless of whether or not visa/mc would accept free software. My guess is that they would object to anything readable by humans. I would be writing c for things easily handled by shell scripts. I find that amusing. I have a lot of experience with disassembly. I have even reverse-engineered machine language code that ran on custom processors which means you have to reverse-engineer the instruction set as part of the task. Closed source might delay the cracker but it also delays pre-crack and post-crack countermeasures. ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote: Tzafrir Cohen wrote: On Sat, Oct 08, 2005 at 07:41:48AM -0400, Paul wrote: Credit card processing would be a good example. You could design *-based systems for both the client(merchant) and server(processor) functions but last I knew visa/mc would not certify open source solutions. Note that you can use whatever license you want for an application that connects to Asterisk via AGI or the manager interface, regardless of whether or not visa/mc would accept free software. My guess is that they would object to anything readable by humans. I would be writing c for things easily handled by shell scripts. PERL included? ;-) I find that amusing. I have a lot of experience with disassembly. I have even reverse-engineered machine language code that ran on custom processors which means you have to reverse-engineer the instruction set as part of the task. Closed source might delay the cracker but it also delays pre-crack and post-crack countermeasures. AGI can't be written in assembly! Anyone up to the task? ;-) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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] How do you verify remote registrations
If you have configured Asterisk to remote to a SIP provider, how do you verify that the registration has been successful? sip show peers Or, do 'sip show registry' depending on who is registering with who. From your words, I'd guess this is more of what you're looking for. ___ --Bandwidth and Colocation sponsored by Easynews.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] No incoming calls from chan_capi 0.6
Hello, I'm trying to use the version 0.6 of chan_capi-cm for outgoing calls it works perfectly but for incoming calls it doesn't work. I tried to set an extension to dial my from-pstn context and it works. So I think there's a problem with my capi.conf or something... Here's a debug when calling -- CONNECT_IND (PLCI=0x101,DID=9100,CID=46720,CIP=0x4,CONTROLLER=0x1) Urgent handler == numeris1: Incoming call '046720' - '9100' Urgent handler -- numeris1: info element PI 84 83 Urgent handler -- numeris1: info element CALLED PARTY NUMBER Urgent handler -- numeris1: info element CHANNEL IDENTIFICATION 89 Urgent handler -- numeris1: info element Sending Complete Urgent handler == numeris1: CAPI Hangingup Urgent handler == numeris1: Interface cleanup PLCI=0x101 Urgent handler -- CONNECT_IND (PLCI=0x101,DID=9100,CID=46720,CIP=0x4,CONTROLLER=0x1) Urgent handler == numeris1: Incoming call '046720' - '9100' Urgent handler -- numeris1: info element PI 84 83 Urgent handler -- numeris1: info element CALLED PARTY NUMBER Urgent handler -- numeris1: info element CHANNEL IDENTIFICATION 89 Urgent handler -- numeris1: info element Sending Complete Urgent handler == numeris1: CAPI Hangingup Urgent handler == numeris1: Interface cleanup PLCI=0x101 Urgent handler And here is my capi.conf [numeris1] ;ntmode=yes isdnmode=msn incomingmsn=* controller=1 group=1 ;prefix=0 softdtmf=on relaxdtmf=on accountcode= context=entrant holdtype=local immediate=yes echocancelold=yes devices=2 Cedric Fontaine ___ --Bandwidth and Colocation sponsored by Easynews.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] No incoming calls from chan_capi 0.6
On Sat, 8 Oct 2005, Cedric Fontaine wrote: Hello, I'm trying to use the version 0.6 of chan_capi-cm for outgoing calls it works perfectly but for incoming calls it doesn't work. I tried to set an extension to dial my from-pstn context and it works. So I think there's a problem with my capi.conf or something... Please increase verbosity (set verbose 5) and switch on debugging (capi debug). Then you should see what's happening. Armin Here's a debug when calling -- CONNECT_IND (PLCI=0x101,DID=9100,CID=46720,CIP=0x4,CONTROLLER=0x1) Urgent handler == numeris1: Incoming call '046720' - '9100' Urgent handler -- numeris1: info element PI 84 83 Urgent handler -- numeris1: info element CALLED PARTY NUMBER Urgent handler -- numeris1: info element CHANNEL IDENTIFICATION 89 Urgent handler -- numeris1: info element Sending Complete Urgent handler == numeris1: CAPI Hangingup Urgent handler == numeris1: Interface cleanup PLCI=0x101 Urgent handler -- CONNECT_IND (PLCI=0x101,DID=9100,CID=46720,CIP=0x4,CONTROLLER=0x1) Urgent handler == numeris1: Incoming call '046720' - '9100' Urgent handler -- numeris1: info element PI 84 83 Urgent handler -- numeris1: info element CALLED PARTY NUMBER Urgent handler -- numeris1: info element CHANNEL IDENTIFICATION 89 Urgent handler -- numeris1: info element Sending Complete Urgent handler == numeris1: CAPI Hangingup Urgent handler == numeris1: Interface cleanup PLCI=0x101 Urgent handler And here is my capi.conf [numeris1] ;ntmode=yes isdnmode=msn incomingmsn=* controller=1 group=1 ;prefix=0 softdtmf=on relaxdtmf=on accountcode= context=entrant holdtype=local immediate=yes echocancelold=yes devices=2 Cedric Fontaine ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Outbound Mediatrix 1204.
I have been able to configure my Asterisk BOX to receive calls from
Mediatrix 1204.
I'm having problems sending calls out via my Mediatrix unit.
The SIP Invite is sent to the Mediatrix but the Mediatrix unit sends
back a Status : 480 Temporarily Unavailable.
This is my configuration on Asterisk;
exten = _78996.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _78996,1,Congestion
On the Mediarix end I have defined my SIP proxy.
I had to enable automatic call direction on my 1204 to send the call to
my Asterisk. If I leave this enabled and make an outgoing call via the
Asterisk box the call come back!
I have seen that several people have this feature working and would be
very grateful if you could share your configuration with me or point me
in the correct direction.
I played with the 1204 about a year ago. The 1204 firmware at that time
required me jump through hops to get it to work. As I recall, we had to
define a 'callerid' number in asterisk for each port of the 1204 that we
wanted to direct traffic to, and then on the 1204 define something that
essentially said... if the callerid from asterisk was '', then route
the call via pstn port 1. If the callerid was '', then route to pstn
port 2, etc.
I don't have a clue whether that approach is still required with their
newer firmware.
At that time, the 1204 worked very well. But, the irregularities of their
firmware was far more then enough to make me return the unit to the
reseller.
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Re: [Asterisk-Users] Where to get the latest SIP Firmware for Polycom Phones?
Hello, I 'll ask to my reseller Harry --- [EMAIL PROTECTED] a écrit : thanks for that, i knew already but it misses the actual version Jesse Keating wrote: On Fri, 2005-10-07 at 11:17 +0200, Kib Eki wrote: Hello, can anybody tell me where to get the latetest SIP Firmware 1.6.2 for the Polycom phones? http://www.freedomphones.net/polycom/files/ ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.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] No incoming calls from chan_capi 0.6
Armin Schindler wrote: Please increase verbosity (set verbose 5) and switch on debugging (capi debug). Then you should see what's happening. *CLI set verbose 6 Verbosity was 3 and is now 6 *CLI capi debug CAPI Debugging Enabled CONNECT_IND ID=001 #0x06ed LEN=0042 Controller/PLCI/NCCI= 0x101 CIPValue= 0x4 CalledPartyNumber = 819100 CallingPartyNumber = 83467201717 CalledPartySubaddress = default CallingPartySubaddress = default BC = 90 90 a3 LLC = default HLC = default AdditionalInfo = default -- CONNECT_IND (PLCI=0x101,DID=9100,CID=467201717,CIP=0x4,CONTROLLER=0x1) numeris1: msn='*' DNID='9100' MSN == numeris1: Incoming call '0467201717' - '9100' INFO_IND ID=001 #0x06ee LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x1e InfoElement = 84 83 INFO_RESP ID=001 #0x06ee LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element PI 84 83 numeris1: Origination is non ISDN INFO_IND ID=001 #0x06ef LEN=0020 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = 819100 INFO_RESP ID=001 #0x06ef LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element CALLED PARTY NUMBER numeris1: INFO_IND DID digits not used in this state. INFO_IND ID=001 #0x06f0 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 INFO_RESP ID=001 #0x06f0 LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element CHANNEL IDENTIFICATION 89 INFO_IND ID=001 #0x06f1 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0xa1 InfoElement = a1 INFO_RESP ID=001 #0x06f1 LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element Sending Complete CONNECT_RESP ID=001 #0x06f1 LEN=0032 Controller/PLCI/NCCI= 0x101 Reject = 0x1 BProtocol B1protocol = 0x0 B2protocol = 0x0 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default ConnectedNumber = default ConnectedSubaddress = default LLC = default AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default DISCONNECT_IND ID=001 #0x06f2 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x0 DISCONNECT_RESP ID=001 #0x06f2 LEN=0012 Controller/PLCI/NCCI= 0x101 == numeris1: CAPI Hangingup == numeris1: Interface cleanup PLCI=0x101 CONNECT_IND ID=001 #0x06f3 LEN=0042 Controller/PLCI/NCCI= 0x101 CIPValue= 0x4 CalledPartyNumber = 819100 CallingPartyNumber = 83467201717 CalledPartySubaddress = default CallingPartySubaddress = default BC = 90 90 a3 LLC = default HLC = default AdditionalInfo = default -- CONNECT_IND (PLCI=0x101,DID=9100,CID=467201717,CIP=0x4,CONTROLLER=0x1) numeris1: msn='*' DNID='9100' MSN == numeris1: Incoming call '0467201717' - '9100' INFO_IND ID=001 #0x06f4 LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x1e InfoElement = 84 83 INFO_RESP ID=001 #0x06f4 LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element PI 84 83 numeris1: Origination is non ISDN INFO_IND ID=001 #0x06f5 LEN=0020 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = 819100 INFO_RESP ID=001 #0x06f5 LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element CALLED PARTY NUMBER numeris1: INFO_IND DID digits not used in this state. INFO_IND ID=001 #0x06f6 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 INFO_RESP ID=001 #0x06f6 LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element CHANNEL IDENTIFICATION 89 INFO_IND ID=001 #0x06f7 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0xa1 InfoElement = a1 INFO_RESP ID=001 #0x06f7 LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element Sending Complete CONNECT_RESP
Re: [Asterisk-Users] Teliax users, g729 question
I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. show translations verifies that the registration took place. When I place a call, having allow=g729 as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a call with g726 as the only allow line, then the call completes as desired. I realize I could just use that (g726), but it seems odd that I cannot connect using g729 when both of us support it. I have checked, double checked, triple checked my account with teliax, and made sure that the g729 box is checked for both sip and iax. I have also contacted support, they have responded, but not with anything that would be considered helpful. My question then to you all is this: Are you connecting to Teliax via g729? if so, how... what are you doing that I might be missing? Your guidance will be most appreciated. I just tried g729 with teliax this morning. It worked fine in both directions using three different did's from them. I did have one test call where audio was one way though. During those test calls I watched the CLI and the calls definitely were g729 without a doubt. I used the exact same teliax server you show above. I'm running cvs-head from yesterday. If you do a iax2 debug, you should be able to spot which system is not compat with g729. That should lead you into diagnosing the problem a little deeper. ___ --Bandwidth and Colocation sponsored by Easynews.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] No incoming calls from chan_capi 0.6
On Sat, 8 Oct 2005, Cedric Fontaine wrote: Armin Schindler wrote: Please increase verbosity (set verbose 5) and switch on debugging (capi debug). Then you should see what's happening. *CLI set verbose 6 Verbosity was 3 and is now 6 *CLI capi debug CAPI Debugging Enabled CONNECT_IND ID=001 #0x06ed LEN=0042 Controller/PLCI/NCCI= 0x101 CIPValue= 0x4 CalledPartyNumber = 819100 CallingPartyNumber = 83467201717 CalledPartySubaddress = default CallingPartySubaddress = default BC = 90 90 a3 LLC = default HLC = default AdditionalInfo = default -- CONNECT_IND (PLCI=0x101,DID=9100,CID=467201717,CIP=0x4,CONTROLLER=0x1) numeris1: msn='*' DNID='9100' MSN == numeris1: Incoming call '0467201717' - '9100' INFO_IND ID=001 #0x06ee LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x1e InfoElement = 84 83 INFO_RESP ID=001 #0x06ee LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element PI 84 83 numeris1: Origination is non ISDN INFO_IND ID=001 #0x06ef LEN=0020 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = 819100 INFO_RESP ID=001 #0x06ef LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element CALLED PARTY NUMBER numeris1: INFO_IND DID digits not used in this state. INFO_IND ID=001 #0x06f0 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 INFO_RESP ID=001 #0x06f0 LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element CHANNEL IDENTIFICATION 89 INFO_IND ID=001 #0x06f1 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0xa1 InfoElement = a1 INFO_RESP ID=001 #0x06f1 LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element Sending Complete CONNECT_RESP ID=001 #0x06f1 LEN=0032 Controller/PLCI/NCCI= 0x101 Reject = 0x1 BProtocol B1protocol = 0x0 B2protocol = 0x0 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default ConnectedNumber = default ConnectedSubaddress = default LLC = default AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default DISCONNECT_IND ID=001 #0x06f2 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x0 DISCONNECT_RESP ID=001 #0x06f2 LEN=0012 Controller/PLCI/NCCI= 0x101 == numeris1: CAPI Hangingup == numeris1: Interface cleanup PLCI=0x101 CONNECT_IND ID=001 #0x06f3 LEN=0042 Controller/PLCI/NCCI= 0x101 CIPValue= 0x4 CalledPartyNumber = 819100 CallingPartyNumber = 83467201717 CalledPartySubaddress = default CallingPartySubaddress = default BC = 90 90 a3 LLC = default HLC = default AdditionalInfo = default -- CONNECT_IND (PLCI=0x101,DID=9100,CID=467201717,CIP=0x4,CONTROLLER=0x1) numeris1: msn='*' DNID='9100' MSN == numeris1: Incoming call '0467201717' - '9100' INFO_IND ID=001 #0x06f4 LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x1e InfoElement = 84 83 INFO_RESP ID=001 #0x06f4 LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element PI 84 83 numeris1: Origination is non ISDN INFO_IND ID=001 #0x06f5 LEN=0020 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = 819100 INFO_RESP ID=001 #0x06f5 LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element CALLED PARTY NUMBER numeris1: INFO_IND DID digits not used in this state. INFO_IND ID=001 #0x06f6 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 INFO_RESP ID=001 #0x06f6 LEN=0012 Controller/PLCI/NCCI= 0x101 -- numeris1: info element CHANNEL IDENTIFICATION 89 INFO_IND ID=001 #0x06f7 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber
[Asterisk-Users] need help-can't not register to asterisk from softphone
Dear all expert, (i posted this question one time, but i couldn't reach the answer -so allow me to post here) 1)I download asterisk realse version 1.2 beta1. After that i compile it successfully and run it with: asterisk -vvvc 2)I follow the instruction in http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html in sip.conf: i add two account: [ivan] type=friend username=ivan secret=pwd_ivan host=dynamic context=tutorial [test] type=friend username=test secret=pwd_test host=dynamic context=tutorial in extension.conf i add like below: [tutorial] extern=1234,1,Dial(SIP/ivan) extern=4321,1,Dial(SIP/test) 3) i use sjphone, make a new profile, in SIP Proxy tag of Profile options windows, i use ip adress of server Fedore (where Asterisk is running) with port 5060 for Proxy domain and User domain. 4) I try to use sjphone to REGISTER to server Asterisk, by using: Ethereal to see the packet. From sjphone:-REGISTERasterisk sjphone ---ICMP Destination unreachable (Host administratively prohibited) asterisk I am sure that from Asterisk server i can ping to client sjphone and vice. And i can navigate internet in Asterisk server. How can i configure Linux server now? I can't understand what it happens? Many thanks for your helping. Julien ___ --Bandwidth and Colocation sponsored by Easynews.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] Teliax users, g729 question
Make sure you have g729 turned on from the Teliax customer panel on their website. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; John Reynolds [EMAIL PROTECTED] Sent: Saturday, October 08, 2005 8:59 AM Subject: Re: [Asterisk-Users] Teliax users, g729 question I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. show translations verifies that the registration took place. When I place a call, having allow=g729 as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a call with g726 as the only allow line, then the call completes as desired. I realize I could just use that (g726), but it seems odd that I cannot connect using g729 when both of us support it. I have checked, double checked, triple checked my account with teliax, and made sure that the g729 box is checked for both sip and iax. I have also contacted support, they have responded, but not with anything that would be considered helpful. My question then to you all is this: Are you connecting to Teliax via g729? if so, how... what are you doing that I might be missing? Your guidance will be most appreciated. I just tried g729 with teliax this morning. It worked fine in both directions using three different did's from them. I did have one test call where audio was one way though. During those test calls I watched the CLI and the calls definitely were g729 without a doubt. I used the exact same teliax server you show above. I'm running cvs-head from yesterday. If you do a iax2 debug, you should be able to spot which system is not compat with g729. That should lead you into diagnosing the problem a little deeper. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote: I find that amusing. I have a lot of experience with disassembly. I have even reverse-engineered machine language code that ran on custom processors which means you have to reverse-engineer the instruction set as part of the task. I think your argument is: Don't require or offer closed source applications since they can be cracked. Similarly we shouldn't lock our doors when we leave home because they can be overridden. Locks, like closed source, are legal barriers that work most of the time for their intended purpose. The discussion of licensing issues on forking Asterisk should assume everyone understands and follows the applicable legal guidelines on software licensing. The earlier point was Asterisk with its commercial license option, and presumably closed source traits, will be required some situations. Having closed source as component of a certified solution is topical ointment enjoyed by purveyors of certificates. If this is true then OpenPBX, lacking a similar license option could be at a competitive disadvantage. But what if OpenPBX attains features that are desireable but uncertifiable because the closed source option does not exist? Then we'll be living in interesting times ( http://www.noblenet.org/reference/inter.htm ). Closed source might delay the cracker but it also delays pre-crack and post-crack countermeasures. What's the alternative? Open source? Cracking is unnecessary with open source. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: need help-can't not register to asterisk from softphone
one more thing is that: in my fedora core 4 where Asterisk is running, i use netstat -ta i can't see: asterisk is listening in port 5060 (which include in sip.conf) On 10/8/05, julien bos [EMAIL PROTECTED] wrote: Dear all expert, (i posted this question one time, but i couldn't reach the answer -so allow me to post here) 1)I download asterisk realse version 1.2 beta1. After that i compile it successfully and run it with: asterisk -vvvc 2)I follow the instruction in http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html in sip.conf: i add two account: [ivan] type=friend username=ivan secret=pwd_ivan host=dynamic context=tutorial [test] type=friend username=test secret=pwd_test host=dynamic context=tutorial in extension.conf i add like below: [tutorial] extern=1234,1,Dial(SIP/ivan) extern=4321,1,Dial(SIP/test) 3) i use sjphone, make a new profile, in SIP Proxy tag of Profile options windows, i use ip adress of server Fedore (where Asterisk is running) with port 5060 for Proxy domain and User domain. 4) I try to use sjphone to REGISTER to server Asterisk, by using: Ethereal to see the packet. From sjphone:-REGISTERasterisk sjphone ---ICMP Destination unreachable (Host administratively prohibited) asterisk I am sure that from Asterisk server i can ping to client sjphone and vice. And i can navigate internet in Asterisk server. How can i configure Linux server now? I can't understand what it happens? Many thanks for your helping. Julien ___ --Bandwidth and Colocation sponsored by Easynews.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 on Solaris SPARC
Hello everybody, Is there anybody successfully have Asterisk running particularly on Sun Fire V100 (64 bits) with Solaris 9? Any hints and suggestions would be much appreciated. Cheers, Anto ___ --Bandwidth and Colocation sponsored by Easynews.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] WiFi Phones
Wait for the next UTStarCom version... Called F3000, Im not sure, but something like that. It will have better battery performance and will have 802.11g support, and many other improvements. It will be available soon. Denis. On 07 de out de 2005, at 00:54, Andy Hamilton wrote: Anyone have good words to say about any of the WiFi handsets currently available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). The unit, I'm guessing, was designed somewhere in Asia, and the language translation shows it a little bit. Sound quality seems pretty good for the few calls I've passed through it. I only have one AP in my house, so I can't comment on roaming. The headset for my cell phone is stereo, and I think the phone would be most happy with a standard 3 conductor plug, but I imagine a headset on a phone is a headset on a phone. The keypad is a touch small, and sometimes I hit the wrong key (and my fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does 802.1x authentication, it sure looks like WEP is the only available security option. Overall: I would recommend purchasing one, for testing at the very least. They are well priced and of good quality. Battery life seems to be pretty good, too. -A ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On Friday 07 October 2005 19:10, Troy Settle wrote: Nice smartass remark... of course anyone can register a domain name. Is forking asterisk legal? Of course it is! Asterisk is under the GPL, which means that anyone can fork it at any time for any reason. Look at this in a positive light... many open source projects have forked, and the branches almost always end up feeding on one another. The difference in this case being of course that OpenPBX can happily continue to feed on any good developments (code wise) that happens in Asterisk but due to Digium's dual license restrictions Asterisk will not be able to feed on code that goes into OpenPBX. This means (and this is already the case if you look at the source tree) that other issues aside, OpenPBX should progress quicker in the long term. (That is if you assume that there is good code in Asterisk that the OpenPBX developers wish to use and visa versa) Cheers -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On Sat, Oct 08, 2005 at 11:59:04AM -0400, Mike M wrote: On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote: Closed source might delay the cracker but it also delays pre-crack and post-crack countermeasures. What's the alternative? Open source? Cracking is unnecessary with open source. Search a bit about security by obscurity. Basically if the security of your system depends on a secret you can't easily change, it will get exposed sooner or later. So you should design it to withstand such leakage. E.g: change a password if it was exposed. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: need help-can't not register to asterisk from softphone
if you do a netstat -an, you will see asterisk on udp port 5060. one more thing is that: in my fedora core 4 where Asterisk is running, i use netstat -ta i can't see: asterisk is listening in port 5060 (which include in sip.conf) On 10/8/05, julien bos [EMAIL PROTECTED] wrote: Dear all expert, (i posted this question one time, but i couldn't reach the answer -so allow me to post here) 1)I download asterisk realse version 1.2 beta1. After that i compile it successfully and run it with: asterisk -vvvc 2)I follow the instruction in http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html in sip.conf: i add two account: [ivan] type=friend username=ivan secret=pwd_ivan host=dynamic context=tutorial [test] type=friend username=test secret=pwd_test host=dynamic context=tutorial in extension.conf i add like below: [tutorial] extern=1234,1,Dial(SIP/ivan) extern=4321,1,Dial(SIP/test) 3) i use sjphone, make a new profile, in SIP Proxy tag of Profile options windows, i use ip adress of server Fedore (where Asterisk is running) with port 5060 for Proxy domain and User domain. 4) I try to use sjphone to REGISTER to server Asterisk, by using: Ethereal to see the packet. From sjphone:-REGISTERasterisk sjphone ---ICMP Destination unreachable (Host administratively prohibited) asterisk I am sure that from Asterisk server i can ping to client sjphone and vice. And i can navigate internet in Asterisk server. How can i configure Linux server now? I can't understand what it happens? Many thanks for your helping. Julien ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
Nice smartass remark... of course anyone can register a domain name. Is forking asterisk legal? Of course it is! Asterisk is under the GPL, which means that anyone can fork it at any time for any reason. Look at this in a positive light... many open source projects have forked, and the branches almost always end up feeding on one another. The difference in this case being of course that OpenPBX can happily continue to feed on any good developments (code wise) that happens in Asterisk but due to Digium's dual license restrictions Asterisk will not be able to feed on code that goes into OpenPBX. This means (and this is already the case if you look at the source tree) that other issues aside, OpenPBX should progress quicker in the long term. (That is if you assume that there is good code in Asterisk that the OpenPBX developers wish to use and visa versa) I'm certainly not an expert on this topic, but if OpenPBX stays with GPL, it would appear that asterisk could use any piece developed under OpenPBX (unless someone there puts restrictions on individual pieces). ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
Nice smartass remark... of course anyone can register a domain name. Is forking asterisk legal? Of course it is! Asterisk is under the GPL, which means that anyone can fork it at any time for any reason. Look at this in a positive light... many open source projects have forked, and the branches almost always end up feeding on one another. The difference in this case being of course that OpenPBX can happily continue to feed on any good developments (code wise) that happens in Asterisk but due to Digium's dual license restrictions Asterisk will not be able to feed on code that goes into OpenPBX. This means (and this is already the case if you look at the source tree) that other issues aside, OpenPBX should progress quicker in the long term. (That is if you assume that there is good code in Asterisk that the OpenPBX developers wish to use and visa versa) I'm certainly not an expert on this topic, but if OpenPBX stays with GPL, it would appear that asterisk could use any piece developed under OpenPBX (unless someone there puts restrictions on individual pieces). Yes they could (I am no expert either) but they can only use it in the GPL version, not in the ABE version. ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
Rich Adamson wrote: I'm certainly not an expert on this topic, but if OpenPBX stays with GPL, it would appear that asterisk could use any piece developed under OpenPBX (unless someone there puts restrictions on individual pieces). No, since Asterisk requires that copyright be assigned to Digium for all patches. Submitters to OpenPBX may be unwilling to do this, especially since that's one of the main reasons for its existance... Tony ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
Rich Adamson wrote: I'm certainly not an expert on this topic, but if OpenPBX stays with GPL, it would appear that asterisk could use any piece developed under OpenPBX (unless someone there puts restrictions on individual pieces). Only if the copyright holder(s) of that code choose to disclaim its use by Digium. ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
Tony Hoyle wrote: No, since Asterisk requires that copyright be assigned to Digium for all patches. Submitters to OpenPBX may be unwilling to do this, especially since that's one of the main reasons for its existance... Please stop spreading misinformation. We have addressed this at least four times in the last six months on this list. Digium does NOT require copyright assignment for contributions to the Asterisk tree. Digium does require either that the code be public domain (unrestricted use), or that Digium be granted a license to reuse the code at our discretion (the disclaimer). ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On Sat, Oct 08, 2005 at 08:43:07PM +0300, Tzafrir Cohen wrote: On Sat, Oct 08, 2005 at 11:59:04AM -0400, Mike M wrote: On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote: Closed source might delay the cracker but it also delays pre-crack and post-crack countermeasures. What's the alternative? Open source? Cracking is unnecessary with open source. Search a bit about security by obscurity. Basically if the security of your system depends on a secret you can't easily change, it will get exposed sooner or later. So you should design it to withstand such leakage. E.g: change a password if it was exposed. As this was related to Mastercard/Visa, they can allow open source, however the software has to be certified to meet their security specs, which may be harder to accomplish for open source. Steve -- NetTek Ltd Fax +44-(0)20 7483 2455 Skype / In stevekennedyuk / UK +442088167166 / US +13106518226 Vonage UK +442079932612 / US +13108577715 / UK mob 07775 755503 Personal Blog http://stevekennedy.blogspot.com Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On 10/8/05, Paul [EMAIL PROTECTED] wrote: Mike M wrote:On Fri, Oct 07, 2005 at 09:45:53PM -0400, Paul wrote:Also consider that there are situations where 100% open source is neverallowed. Check out visa/mastercard processor certification for a good example. Digium dual licensing availability means I could actually standa chance of using asterisk as the basis for systems used by military andlaw enforcement in applications that require extremely high security. There is a popular vendor of closed source products whose security has beencompromised often. The security of OpenSSH is well established.Reading this list iwe learn that the open source version of Asterisk is currently being used by military personnel.Asterisk offers ways for users to implement eavesdropping applications whichundermines the goal of attaining extremely high security.Open source is for sharing if that's feasible and closed source is not. Dual licensing is for both.My point was not to argue that closed source enhances security. I wasjust pointing out that there are situations where the customer will notaccept open source. Credit card processing would be a good example. You could design *-basedsystems for both the client(merchant) and server(processor) functionsbut last I knew visa/mc would not certify open source solutions. Off topic but wanted to correct this.. Its not the software that has to be certified, it's the merchant (or payment processor). Ya you can pay a security auditor to look at your software and say that it's compliant, but it doesn't really mean anything. If you are a qualifying merchant or payment processor you would still have to go through the complete audit even if you used 'certified' software. Also, as a merchant you either have to go through the full audit yourself, or use a certified payment gateway. You cannot for example use 'certified' software as a merchant and connect directly to the bank networks without going through the full audit yourself at an average cost of around $20,000. Chris ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On 10/8/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: Tony Hoyle wrote: No, since Asterisk requires that copyright be assigned to Digium for all patches. Submitters to OpenPBX may be unwilling to do this, especially since that's one of the main reasons for its existance... Please stop spreading misinformation. We have addressed this at leastfour times in the last six months on this list.Digium does NOT require copyright assignment for contributions to theAsterisk tree. Digium does require either that the code be public domain (unrestricteduse), or that Digium be granted a license to reuse the code at ourdiscretion (the disclaimer). Being that Digium wants to be able to sell a commercial version, I don't see how they could have been more accomodating then this. Digium can only put in their commercial version what they themselves have written, or what others have freely given them to use under the public domain. The only people that would have a problem with this are the one that believe so strongly in the GPL that it's the only license they will permit to be used for their contributions.. They won't be happy unless everyone else does things their way. They wouldn't be happy if asterisk was BSD or MIT licensed either. Chris ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
snacktime wrote: Being that Digium wants to be able to sell a commercial version, I don't see how they could have been more accomodating then this. Digium can They could just use the GPL as is, since they chose the license in the first place.. they clearly have no issues with it. They already have the rights to use the code granted by the GPL - that's not what the disclaimer is for. The disclaimer gives them the same rights as the owner so they can relicense the contributed code under a non-GPL license for commercial reasons. Not everyone is happy with that, clearly. TBH I'd rather digium had chosen something like BSD to start with and avoided all the GPL politics but the situation we have is the one we have. Tony ___ --Bandwidth and Colocation sponsored by Easynews.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] Digium G.729 codec modules updated
On 10/08/05 13:32 Kevin P. Fleming said the following: Once I return from Astricon, we will use this new build system to produce FreeBSD modules for the same processor architectures, and also this is wonderful ! how long has it been since licensed g.729 codecs were available from digium for freebsd ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On 10/07/05 23:28 Jon Pounder said the following: There are people out there who wish to contribute, and not have their work lost on an individual project website since they do not choose to accept digium's terms to contribute to asterisk. This gives them an opportunity to do so, and have their work aggregated with everyone else in the same category, so it is one stop shopping for users. that makes a lot of sense, considering that many have voiced similar opinions here in the past. however i would urge the openpbx.org folk to not diverge too much from the main asterisk code base, so as to ensure a proper stream for it. too much divergence and we have two pieces of software competing for each other. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: need help-can't not register to asterisk from softphone
There are a lots of strange things in my server fedora core 4 for asterisk. The first thing is that: - after (make, make install, make sample), i try to run asterisk by asterisk -vc from terminal. Then, i can enter CLI screen. -now i try to use netstat -a or netstat --listening, i can't see the port 5060. What does it means? it means that when i send REGISTER from client to server, cos server doesn't listen in port 5060, so il reponse by ICMP destination unreachable. Am i right? How can i install asterisk? How did you install asterisk? Thank you for your instruction. On 10/8/05, julien bos [EMAIL PROTECTED] wrote: one more thing is that: in my fedora core 4 where Asterisk is running, i use netstat -ta i can't see: asterisk is listening in port 5060 (which include in sip.conf) On 10/8/05, julien bos [EMAIL PROTECTED] wrote: Dear all expert, (i posted this question one time, but i couldn't reach the answer -so allow me to post here) 1)I download asterisk realse version 1.2 beta1. After that i compile it successfully and run it with: asterisk -vvvc 2)I follow the instruction in http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html in sip.conf: i add two account: [ivan] type=friend username=ivan secret=pwd_ivan host=dynamic context=tutorial [test] type=friend username=test secret=pwd_test host=dynamic context=tutorial in extension.conf i add like below: [tutorial] extern=1234,1,Dial(SIP/ivan) extern=4321,1,Dial(SIP/test) 3) i use sjphone, make a new profile, in SIP Proxy tag of Profile options windows, i use ip adress of server Fedore (where Asterisk is running) with port 5060 for Proxy domain and User domain. 4) I try to use sjphone to REGISTER to server Asterisk, by using: Ethereal to see the packet. From sjphone:-REGISTERasterisk sjphone ---ICMP Destination unreachable (Host administratively prohibited) asterisk I am sure that from Asterisk server i can ping to client sjphone and vice. And i can navigate internet in Asterisk server. How can i configure Linux server now? I can't understand what it happens? Many thanks for your helping. Julien ___ --Bandwidth and Colocation sponsored by Easynews.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] ip phones
I am in the process of converging PSTN and internet. I am now looking to migrate from pots to VoIP handsets that are IEEE 802.3af (POE) compliant. My question is this. Does the 3Com 3101 basic phone (P/Ns 3C10401A or 3C10401SPKRA) work with Asterisk??? ___ --Bandwidth and Colocation sponsored by Easynews.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] Digium G.729 codec modules updated
Dinesh Nair wrote: this is wonderful ! how long has it been since licensed g.729 codecs were available from digium for freebsd ? They have been on the web/FTP sites for some time, in the 'unsupported' directory. ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On 10/8/05, Tony Hoyle [EMAIL PROTECTED] wrote: snacktime wrote: Being that Digium wants to be able to sell a commercial version, I don't see how they could have been more accomodating then this. Digium canThey could just use the GPL as is, since they chose the license in the first place.. they clearly have no issues with it.They already have the rights to use the code granted by the GPL - that'snot what the disclaimer is for.The disclaimer gives them the same rights as the owner so they can relicense the contributed code under a non-GPL license for commercialreasons.Not everyone is happy with that, clearly. I understand, that's why I said 'Being that Digium wants to be able to sell a commercial version'. TBH I'd rather digium had chosen something like BSD to start with andavoided all the GPL politics but the situation we have is the one we have. Agreed. ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On Sat, Oct 08, 2005 at 08:41:00PM +0100, Tony Hoyle wrote: TBH I'd rather digium had chosen something like BSD to start with and avoided all the GPL politics but the situation we have is the one we have. But then you wouldn't have to pay them if you wanted your own propritary fork. Not to mention that it prevents evil others to create a propritary fork: even a fork must have a GPL license[*] [*] Almost GPL: openssl and openh323's licenses is incompatible with the GPL, and hence the current modified GPL license will still have to be used. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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] Digium G.729 codec modules updated
On 10/09/05 03:58 Kevin P. Fleming said the following: Dinesh Nair wrote: this is wonderful ! how long has it been since licensed g.729 codecs were available from digium for freebsd ? They have been on the web/FTP sites for some time, in the 'unsupported' directory. and they still go for US$10 a pop ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On 10/09/05 02:46 Rich Adamson said the following: I'm certainly not an expert on this topic, but if OpenPBX stays with GPL, it would appear that asterisk could use any piece developed under OpenPBX (unless someone there puts restrictions on individual pieces). asterisk could, but i doubt digium would commit this to the asterisk cvs due to it being non-disclaimed for their use. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
On 10/09/05 02:46 Rich Adamson said the following: I'm certainly not an expert on this topic, but if OpenPBX stays with GPL, it would appear that asterisk could use any piece developed under OpenPBX (unless someone there puts restrictions on individual pieces). if it's a fork of asterisk, it /has/ to be under the GPL. period. however, pieces from openpbx could make their way into asterisk but not into a closed source version of asterisk (ABE for example) without the consent of the authors of those pieces. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.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] Digium G.729 codec modules updated
Dinesh Nair wrote: and they still go for US$10 a pop ? Patent indemnification licenses are completely separate from the codec binary you choose to use. There is no price difference for CPU type, OS platform or anything else. ___ --Bandwidth and Colocation sponsored by Easynews.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] Outgoing call: hangup after answer
Hi,
When we make an
outgoing call on ISDN (zaphfc) with overlap dialing we get immidiate hangup
after answer. But when we place a full number before dialing everything is ok.
Any help appriciated!! Thanks
here is info with
debug:
== Primary
D-Channel on span 1 up -- Executing Dial("SIP/200-164c",
"zap/g1/|100|tc") in new stack-- Making new call for cr
192 -- Requested transfer capability: 0x00 -
SPEECH Protocol Discriminator: Q.931 (8) len=22 Call Ref:
len= 1 (reference 64/0x40) (Originator) Message type: SETUP (5)
[04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std:
0 Info transfer capability: Speech
(0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: A-Law (35) [18 01 81]
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred
Dchan:
0
ChanSel: B1
channel
] [6c 05 21 80 32 30 30] Calling Number (len= 7) [ Ext: 0
TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163)
(1)
Presentation: Presentation permitted, user number not screened (0) '200'
] [70 01 c1] Called Number (len= 3) [ Ext: 1 TON:
Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
'' ] -- Called g1/ Protocol Discriminator: Q.931
(8) len=11 Call Ref: len= 1 (reference 192/0xC0)
(Terminator) Message type: SETUP ACKNOWLEDGE (13) [18 01
89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive Dchan:
0
ChanSel: B1
channel
] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1
Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network
serving the local user
(2)
Ext: 1 Progress Description: Inband information or appropriate pattern now
available. (8) ]-- Processing IE 24 (cs0, Channel Identification)--
Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931
(8) len=8 Call Ref: len= 1 (reference 64/0x40)
(Originator) Message type: INFORMATION (123) [70 02 c1
39] Called Number (len= 4) [ Ext: 1 TON: Subscriber Number
(4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9' ]
Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1
(reference 64/0x40) (Originator) Message type: INFORMATION (123)
[70 02 c1 35] Called Number (len= 4) [ Ext: 1 TON: Subscriber
Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5'
] Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1
(reference 192/0xC0) (Terminator) Message type: CALL PROCEEDING
(2) -- Zap/1-1 is making progress passing it to
SIP/200-164c Protocol Discriminator: Q.931 (8) len=12 Call
Ref: len= 1 (reference 192/0xC0) (Terminator) Message type: ALERTING
(1) [1e 02 84 88] Progress Indicator (len= 4) [ Ext: 1
Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network
serving the remote user
(4)
Ext: 1 Progress Description: Inband information or appropriate pattern now
available. (8) ] [1e 02 84 82] Progress Indicator (len= 4) [
Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public
network serving the remote user
(4)
Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ]--
Processing IE 30 (cs0, Progress Indicator)-- Processing IE 30 (cs0, Progress
Indicator)
-- Zap/1-1 is ringing, hanging
up.
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered,
peerstate Call Received Protocol Discriminator: Q.931 (8)
len=8 Call Ref: len= 1 (reference 64/0x40) (Originator) Message
type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext:
1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private
network serving the local user
(1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1)
] -- Hungup 'Zap/1-1' Protocol Discriminator:
Q.931 (8) len=4 Call Ref: len= 1 (reference 192/0xC0)
(Terminator) Message type: RELEASE (77)NEW_HANGUP DEBUG: Calling
q931_hangup, ourstate Null, peerstate Release Request Protocol
Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference
64/0x40) (Originator) Message type: RELEASE COMPLETE (90) [08 02
81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: Private network serving the local user
(1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1)
]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate
NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate
Null -- Executing Macro("SIP/200-164c", "hangupcall") in
new stack -- Executing ResetCDR("SIP/200-164c", "w") in
new stack -- Executing NoCDR("SIP/200-164c", "") in new
stack -- Executing Wait("SIP/200-164c", "5") in new
stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/200-164c' in macro 'hangupcall' == Spawn extension
(from-internal, h, 1) exited non-zero on
'SIP/200-164c'
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Re: [Asterisk-Users] Digium G.729 codec modules updated
On 10/09/05 04:45 Kevin P. Fleming said the following: Dinesh Nair wrote: and they still go for US$10 a pop ? Patent indemnification licenses are completely separate from the codec binary you choose to use. There is no price difference for CPU type, OS platform or anything else. understood, thanx for the clarification, kevin. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.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] Digium G.729 codec modules updated
Have you founy any real life performance benefit of x86_64 (particularly EM64T on Xeon) as apposed to plan old x86? On 10/9/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: Dinesh Nair wrote: and they still go for US$10 a pop ?Patent indemnification licenses are completely separate from the codecbinary you choose to use. There is no price difference for CPU type, OS platform or anything else.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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] Digium G.729 codec modules updated
Eric Bishop wrote: Have you founy any real life performance benefit of x86_64 (particularly EM64T on Xeon) as apposed to plan old x86? Yes. On my Athlon-64 2200+ system on my desk, the 'opteron' version encodes a 6722 block sample file in 478ms; the 'i686' version does it in 514ms. The 'i386' version is somewhere near 600ms, since it has no fancy instruction scheduling. Interestingly, _all_ of the 32-bit x86 optimized versions run just fine on that machine, meaning that GCC did not opt to use any instructions that are specific to a processor model/family... the performance improvements come only from scheduling the instruction flow. ___ --Bandwidth and Colocation sponsored by Easynews.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] WiFi Phones
The F3000 is not anticipated to be available for distribution until late December/January, FYI. Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Denis Galvão - iSolve wrote: Wait for the next UTStarCom version... Called F3000, Im not sure, but something like that. It will have better battery performance and will have 802.11g support, and many other improvements. It will be available soon. Denis. On 07 de out de 2005, at 00:54, Andy Hamilton wrote: Anyone have good words to say about any of the WiFi handsets currently available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). The unit, I'm guessing, was designed somewhere in Asia, and the language translation shows it a little bit. Sound quality seems pretty good for the few calls I've passed through it. I only have one AP in my house, so I can't comment on roaming. The headset for my cell phone is stereo, and I think the phone would be most happy with a standard 3 conductor plug, but I imagine a headset on a phone is a headset on a phone. The keypad is a touch small, and sometimes I hit the wrong key (and my fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does 802.1x authentication, it sure looks like WEP is the only available security option. Overall: I would recommend purchasing one, for testing at the very least. They are well priced and of good quality. Battery life seems to be pretty good, too. -A ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] How to check what codec translations are in use in a call?
How does one check what codec translations are in use in a call? I am connecting to sip system which says 488 4XX Not Acceptable Here. I don't know what is stopping the call from being accepted and I'd like to know if there are codec issues involved. /Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.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] Digium G.729 codec modules updated
On a dual processor Xeon (EM64T) would you reccomend turning hypertreading on or off? I tend go for it off dual processor machines just in case 2 processes end up on the one physical processor rather than 2 processes on 2 different physical processors. What do you think?On 10/9/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:Eric Bishop wrote: Have you founy any real life performance benefit of x86_64 (particularly EM64T on Xeon) as apposed to plan old x86?Yes. On my Athlon-64 2200+ system on my desk, the 'opteron' versionencodes a 6722 block sample file in 478ms; the 'i686' version does it in514ms. The 'i386' version is somewhere near 600ms, since it has no fancy instruction scheduling.Interestingly, _all_ of the 32-bit x86 optimized versions run just fineon that machine, meaning that GCC did not opt to use any instructionsthat are specific to a processor model/family... the performance improvements come only from scheduling the instruction flow.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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] Digium G.729 codec modules updated
Eric Bishop wrote: On a dual processor Xeon (EM64T) would you reccomend turning hypertreading on or off? I tend go for it off dual processor machines just in case 2 processes end up on the one physical processor rather than 2 processes on 2 different physical processors. What do you think? I have no opinion; it would require real-world testing to know for sure. My instincts tell me that HT would be counter-productive when running lots of transcoding, though, since HT CPUs don't have two FPUs. Dual-core CPUs do, obviously, so that would be a different situation. ___ --Bandwidth and Colocation sponsored by Easynews.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] How to check what codec translations are in use in a call?
On 10/09/05 06:01 Obelix said the following: I am connecting to sip system which says 488 4XX Not Acceptable Here. I don't know what is stopping the call from being accepted and I'd like to know if there are codec issues involved. it's possible. try connecting with 'sip debug' turned on, and check codec capabilities of both ends of the connection. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.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 Log Color Coding
Hi, Is there anyway to eliminate the color coding (for example [1;36;40m) to be stored in asterisk log file? Regards. __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --Bandwidth and Colocation sponsored by Easynews.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 Log Color Coding
On Sat, Oct 08, 2005 at 04:15:59PM -0700, Samy Antoun wrote: Hi, Is there anyway to eliminate the color coding (for example [1;36;40m) to be stored in asterisk log file? Not to use the color options? It seems that exactly the same prints go to the log and to the CLI, right? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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 Log Color Coding
Tzafrir Cohen wrote: It seems that exactly the same prints go to the log and to the CLI, right? Asterisk already strips the color codes before putting the output into the log files. ___ --Bandwidth and Colocation sponsored by Easynews.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 Log Color Coding
--- Kevin P. Fleming [EMAIL PROTECTED] wrote: Asterisk already strips the color codes before putting the output into the log files. Kevin, This is NOT true, bellow is some of my asterisk log file: Oct 8 16:41:49 VERBOSE[4016]: -- Executing [1;36;40mSetCallerID[0;37;40m([1;35;40mIAX2/[EMAIL PROTECTED]/1[0;37;40m, [1;35;40mSamy Antoun 8184281334[0;37;40m) in new stack Oct 8 16:41:49 VERBOSE[4016]: -- Executing [1;36;40mSetGroup[0;37;40m([1;35;40mIAX2/[EMAIL PROTECTED]/1[0;37;40m, [1;35;40mOUT_3[0;37;40m) in new stack Oct 8 16:41:49 VERBOSE[4016]: -- Executing [1;36;40mCheckGroup[0;37;40m([1;35;40mIAX2/[EMAIL PROTECTED]/1[0;37;40m, [1;35;40m2[0;37;40m) in new stack Oct 8 16:41:49 VERBOSE[4016]: -- Executing [1;36;40mSetVar[0;37;40m([1;35;40mIAX2/[EMAIL PROTECTED]/1[0;37;40m, [1;35;40mDIAL_NUMBER=19056661572[0;37;40m) in new stack Oct 8 16:41:49 VERBOSE[4016]: -- Executing [1;36;40mSetVar[0;37;40m([1;35;40mIAX2/[EMAIL PROTECTED]/1[0;37;40m, [1;35;40mDIAL_TRUNK=3[0;37;40m) in new stack Oct 8 16:41:49 VERBOSE[4016]: -- Executing [1;36;40mAGI[0;37;40m([1;35;40mIAX2/[EMAIL PROTECTED]/1[0;37;40m, [1;35;40mfixlocalprefix[0;37;40m) in new stack Oct 8 16:41:49 VERBOSE[4016]: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix Oct 8 16:41:50 VERBOSE[4016]: -- AGI Script fixlocalprefix completed, returning 0 Oct 8 16:41:50 VERBOSE[4016]: -- Executing [1;36;40mSetVar[0;37;40m([1;35;40mIAX2/[EMAIL PROTECTED]/1[0;37;40m, [1;35;40mOUTNUM=19056661572[0;37;40m) in new stack Oct 8 16:41:50 VERBOSE[4016]: -- Executing [1;36;40mCut[0;37;40m([1;35;40mIAX2/[EMAIL PROTECTED]/1[0;37;40m, [1;35;40mcustom=OUT_3|:|1[0;37;40m) in new stack __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --Bandwidth and Colocation sponsored by Easynews.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] Does anyone know what this means
When I try to dial through a pbx I receive this message to 216.127.66.119:0 Oct 8 23:53:48 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81331cc (len 670) to 216.127.66.119 returned -1: Invalid argument Retransmitting #5 (no NAT): The line is silent and nothing happens. /Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.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] Cannot dial SIP via asterisk
I have been trying to connect via sip and things don't seem to work. What do messages like this mean? Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834 (len 361) to 216.127.66.119 returned -1: Invalid argument Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Destroying call '[EMAIL PROTECTED]' O This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.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] Does anyone know what this means
Check your sip.conf settings and make sure you have nat=yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Obelix Sent: Saturday, October 08, 2005 7:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Does anyone know what this means When I try to dial through a pbx I receive this message to 216.127.66.119:0 Oct 8 23:53:48 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81331cc (len 670) to 216.127.66.119 returned -1: Invalid argument Retransmitting #5 (no NAT): The line is silent and nothing happens. /Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
Steve Kennedy wrote: On Sat, Oct 08, 2005 at 08:43:07PM +0300, Tzafrir Cohen wrote: On Sat, Oct 08, 2005 at 11:59:04AM -0400, Mike M wrote: On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote: Closed source might delay the cracker but it also delays pre-crack and post-crack countermeasures. What's the alternative? Open source? Cracking is unnecessary with open source. Search a bit about security by obscurity. Basically if the security of your system depends on a secret you can't easily change, it will get exposed sooner or later. So you should design it to withstand such leakage. E.g: change a password if it was exposed. As this was related to Mastercard/Visa, they can allow open source, however the software has to be certified to meet their security specs, which may be harder to accomplish for open source. It's not harder. It's just different. A number of things have similar requirements. The ISDN4Linux folk have certain versions of their software approved by the telecoms bodies in Europe. They need to tie down exactly what was approved, so any other versions emit a notice that says they are unapproved versions. They do this with a signature on the approved version. It seems to work out OK. Regards, Steve ___ --Bandwidth and Colocation sponsored by Easynews.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] ATA does not register
I am not able to register an external ATA on my asterisk 2.0 Beta This is the debug Any idea? server01*CLI -- SIP read from CLIENTIP:5060: REGISTER sip:SIPSERVERIP SIP/2.0 Via: SIP/2.0/UDP PRIVATEIP;rport;branch=z9hG4bK455E4AEA9A9954FB135D6D788DA2 From: sip:[EMAIL PROTECTED];tag=1564789518 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Max-Forwards: 70 User-Agent: AVM FRITZ!Box Fon ata 11.03.37 Supported: 100rel, replaces Allow-Events: telephone-event Allow-Events: refer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, PRACK, INFO, SUBSCRIBE, NOTIFY, REFER Accept: application/sdp Accept-Encoding: identity Content-Length: 0 ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: www.openpbx.org
Steve Underwood wrote: Steve Kennedy wrote: On Sat, Oct 08, 2005 at 08:43:07PM +0300, Tzafrir Cohen wrote: On Sat, Oct 08, 2005 at 11:59:04AM -0400, Mike M wrote: On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote: Closed source might delay the cracker but it also delays pre-crack and post-crack countermeasures. What's the alternative? Open source? Cracking is unnecessary with open source. Search a bit about security by obscurity. Basically if the security of your system depends on a secret you can't easily change, it will get exposed sooner or later. So you should design it to withstand such leakage. E.g: change a password if it was exposed. As this was related to Mastercard/Visa, they can allow open source, however the software has to be certified to meet their security specs, which may be harder to accomplish for open source. It's not harder. It's just different. A number of things have similar requirements. The ISDN4Linux folk have certain versions of their software approved by the telecoms bodies in Europe. They need to tie down exactly what was approved, so any other versions emit a notice that says they are unapproved versions. They do this with a signature on the approved version. It seems to work out OK. Regards, Steve I think that the important thing to remember is that a good reverse engineer can take the object code from a rom and produce source files that are better commented than the original source ever was. I close my source because it's mine and it's none of your business but I don't get a false sense of security from doing that. There are people who specialize in taking gate array chips apart in a very careful manner in order to get the programmed logic patterns using a microscope. If I can buy/build a good enough logic analyzer I can get what I need without even powering down your product. So consider that if I can clone your electronic key device, disassembling the binaries for your closed source software is a minor obstacle. ___ --Bandwidth and Colocation sponsored by Easynews.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] Regcontext/regexten broken??
Recently I've noticed two bits of odd behavior with respect to regcontext/regexten in CVS HEAD 1.2 Beta1, and I was wondering if anyone could shed some light on this. I've set up a regcontext in sip.conf. I've set up two users with regexten entries, one in sip.conf and one in a mysql realtime table. The first bit of oddness is that regexten seems to worksomewhat as described for users whose entries are in sip.conf, but for the user whose entries are in a realtime database, it doesn't seem to workat all. Specifically for the sip.conf user, the cli reports adding the extension upon registration, and 'show dialplan' indeed shows the added entry. For a user configured through a realtime database, the cli reports adding the extension upon registration, but 'show dialplan' shows no added extension (and indeed attempts to dial the allegedly registered extension fail). The second bit of oddness is that in the sip.conf.sample it states Patterns may be used in regexten however, while registering a sip user with regexten=_45X does yield an entry (according to 'show dialplan' for the regcontext) of '_45X' = 1. Noop(test)', attempts to dial anything that should match that pattern (451, 452, etc) in that context result in reports ofno such extension...it appears almost as if pattern matching is not being performed on extensions added by SIP. So...question is, what's broken here? Is is Asterisk? My understanding? Or my installation of Asterisk? All three...?? ;-) If anyone can shed some light, I'd greatly appreciate it. Stewart ___ --Bandwidth and Colocation sponsored by Easynews.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] Configuring TDM400 in Australia
Hi, all I have installed TDM400 with 1 FXS and 1 FXP ports. Now I am goig through documentation on how to configure it. It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one do I use? Can someone send me sample zaptel.conf file for Australia? This will save me some time and will be used as a working example. Thanks, Rudolf ___ --Bandwidth and Colocation sponsored by Easynews.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
