RE: [asterisk-users] Digium h/w serial numbers
Hi, You most probably kept the invoice So contact digium. My experience was that they are human Regards, t. jacobson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor Peirce Sent: dimanche 22 avril 2007 19:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Digium h/w serial numbers Hello, I'm at a loss for a way to find the serial number of a Digium analog card without physically removing it from the server. The only time I have physical access to this particular installation is during business hours and that's obviously a bad time to be taking a server down. It seems that I need the serial number to get a free copy of HPEC... but unless someone can convince me otherwise, I have a feeling it would just be easier to shell out the $10 per channel to avoid the downtime and drive out there. Thanks, Trevor ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re:[asterisk-users] T1 cable for Digium T1/E1 Cards
cat5 cables are ok if you use straight cables. crossed cables are different as ethernet signals use other pin layout than e1. and beside the 'official' e1 crossed, there seems to be other layouts. this has been discussed here, so browse the archives. (my pc gives me headaches now, otherwise i would have provided you some urls) regards t. jacobson -- Initial header --- >From : [EMAIL PROTECTED] To : asterisk-users@lists.digium.com CC : Date : Sun, 18 Mar 2007 18:42:31 -0400 Subject : [asterisk-users] T1 cable for Digium T1/E1 Cards > Is there any technical difference between a T1 cable and a cat5e patch > cable as far as using them with Digium T1/E1 cards? C --- Scarlet ADSL Unlimited - Only 24,95 euro per month. Max download Speed up to 6 Mbps, download volume of 30 GB. Order now... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re:[asterisk-users] Looking for starting point?
hi, hi, i did wrote (assembler) programs for cp/m! if your experience is more on telephony', i think you will find trixbox easier. in one cd you will have a ready system. if your hardware is fully recognized, great ! do not use a too old machine nor a too new one. mind that the install will erase your hd. so buy a new cd if you would want to go back to windows. a 'normal' distribution will allow for dual-boot (not trixbox). mandrake, suse, fedora, red-hat & centeos are good candidate. (k)ubuntu, debian, ... are also nice one but then you would have to download, compile, setup all. this could be harder to learn. but you learn a deeper way maybe go to a linux group in your neighborhood most people there are happy to welcome a newbie. buy some books about linux and stick with one distribution (you can change later) my 2c from a young chap of nearly 48 t. jacobson -- Initial header --- >From : [EMAIL PROTECTED] To : asterisk-users@lists.digium.com CC : Date : Sun, 18 Feb 2007 14:05:15 -0500 Subject : [asterisk-users] Looking for starting point? > Hi, > > I am a retired telephone tech/manager who recently had a bad experience with a local company offering digital phone service (VoIP). I have spent the last thirty years in the PSTN network, switching, PBX and key system field and am interested in learning more about VoIP. My background also includes programming, mostly specialized applications to interact with the PSTN network. Most of my experience in this field have been with Borland products, specifically Delphi. I also have been involved with database programming, same platform as the communications. > > My computer experience started with the operating system CPM (I'm not really that old, only 56). The best platform now seems to be Linux so now since I am retired now, it seems a good time to learn something new. I also have been looking at Asterisk which most companies seem to be using for a PBX platform. I found out by accident that the local company I had the problem with uses this PBX software. > > Could someone steer me in the right direction as to where to start? I spent most of my career in the telephone industry in a 'bush' area of Alaska so pretty much had to teach myself what I needed to know about computers but I can learn almost anything from a book and by asking questions when I get stuck. Most of my experience was before the Internet so I plan on using this avenue to advance my knowledge. > > I understand what a broad scope I am asking about so would appreciate any tips to help me get started. Since there are many 'brands' of Linux what is the best one to start with? Which Linux will be better when I get to the point of working with Asterisk? Any tips or ideas on books, online tutors, discussions or anything of this nature would be much appreciated. > > I hope to add to this group if I can be any assistance from the 'other side', the PSTN network. > > Thank You, > > Gary H. Thompson > > --- Scarlet One Unlimited Free national calls, surf up to 6 Mbit/s, 50 GB download volume For only EUR 49,95 per month. No Belgacom subscription needed. All in! http://www.scarlet.be ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
hi,
i think the problem is here :
exten => _321[0123],n,Dial(SIP/${EXTEN}),30,to)
|
replace with
exten => _321[0123],n,Dial(SIP/${EXTEN},30,to)
note, i removed the parenthesis ')' after the {EXTEN}
this should do
regards,
jacobson
---
Scarlet ONE - Combine ADSL with unlimited fixed phone and save 400 euros
http://www.scarlet.be
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re:[asterisk-users] CDR - uniqueid
hi, i think, by default, 'uniqueID' is created by the asterisk. if this is correct, you would (eventually) have non-uniqueID's i saw somewhere in the wiki that someone suggested a change (in the code ?) so that 'uniqueID' would be generated by the database. unique-id being the primary key and autogenerated, it IS unique (cdr+mysql or similar query) it was a rather small change shaoul jacobson -- Initial header --- >From : [EMAIL PROTECTED] To : "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com,asterisk-dev@lists.digium.com CC : Date : Thu, 1 Feb 2007 12:15:21 +0100 Subject : [asterisk-users] CDR - uniqueid > Is uniqueid globally unique? I have three Asterisk installations and I need to store data from all of them in same database, in same table. Will this uniqueid field be unique? > > > -- > Tomislav Parcina > Lama Computers Split --- Scarlet ONE - Combine ADSL with unlimited fixed phone and save 400 euros http://www.scarlet.be ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] FXO module burn out !?
Hi, Try to swap both fxo modules. This way you will notice if the module is out or another problem is present. The tdm400 could be damaged or there is a configuration problem. Regards, -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- Hello list, Howto be shure if one FXO module on TDM400P is not working because is burn out or something like physically demaged. It worked for an almost a year then just stopped. The next FXO module on the same card is working like charm. Thanks. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.5/405 - Release Date: 1/08/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Load balenced (ADSL) network connections, is it possible?
Unless my brain did melt down under the high heat here, there are a few things I would share : - 1 channel u-law uses at least 80k because of encapsulation So, think about compression - You can never use 100% of a bandwidth. (80% is a good max) - The values given are maximum values, far from the guaranteed values - Instead of adsl, I would go for sdsl (symmetrical) Where adsl is possible, sdsl is normally also possible Sdsl is more expensive but you can demand sla, higher guaranteed bandwidth (up to 50 or 100%) Never forget that the guaranteed bandwidth is between your router and the first equipment of your provider. You get NO warranty you have that bandwidth further away even on your isp backbone. Isp sales men always forget to talk about that If they say you get 100% to use all the time, sign immediately IF they put that WRITTEN alongside financial compensation if they fail. Dear mister sales man, you do not have to worry about compensation since you guaranteed it would not happen ;-) Other problems you will face with several IP’s : Even if you can manage several IP’s for the outgoing traffic, you cannot for incoming traffic as those are unaware of your situation Or they need to put multiple entries for you asterisk and try those sequentially Or you could go up for incoming calls and down for outgoing calls That is a simple solution, that will statistically work but NO warranty NOT really a telco way Routing problems might occur … Regards, T. Jacobson -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.2/393 - Release Date: 19/07/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.2/393 - Release Date: 19/07/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] best hardphone for Asterisk?
If you do not have a budget, grandstream is not bad You do not get what you do not pay But you do not allways get what you paid for t. Jacobson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Victor Sent: vendredi 23 juin 2006 14:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] best hardphone for Asterisk? Crazy Boy schrieb: > We have implemented "Asterisk" in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost? -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.2/373 - Release Date: 22/06/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk server
Hi, With 30 users and NO transcoding, that is certainly enough. Even if you use ‘real-time’ configuration (that requires a SQL server) Now, if you system will be accessible both from inside (LAN) and outside (Internet), I would advice a second network card (10/100) Regards, T. Jacobson From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Nowrot Sent: mercredi 14 juin 2006 11:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk server Hi, I have to build Asterisk server for about 30 user (30 concurrent calls). I decided to buy this box: -- motherboard Intel E7210 + Hence Rapids -- processor P4 3.0 GHz -- RAM 2x512 MB DDR ECC -- network interface Intel 82541 GI Is this configuration enough to handle 30 users at the same time. I am not planning to use any transcoding (everything will be alaw). Cheers Andrew -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13/06/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13/06/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OOH323 towards ciscogateway(2691)callsetupfailsat q931: Mandatory informationelement ismissing (96)
Dan, > I'd recommend reporting the problem along with a debug log and > traces from the router to: ooh323c-devel@lists.sourceforge.net > The developers have been quite responsive to problems I've had. Thank you for tips. I thing I must recompile the ooh323 module with 'make debug' then 'make install' in order to have a debug log. Correct me if I am wrong. With those & some traces, I will contact the developers. Regards -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.13.12/193 - Release Date: 6/12/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OOH323 towards cisco gateway(2691)callsetupfails at q931: Mandatory information element ismissing (96)
Dan, The version from objective systems is indeed 0.8 This is a general h323 stack (or driver) The latest asterisk-specific version (named asterisk-ooh323c) I found is 0.3 >From http://ftp.digium.com/pub/asterisk/h323/ there is 0.2 Since I did 'stabilize' my cvs, I just wanted to add a h323 channel. I could just take the h323 part from the 1.2.1-addon. But I fear that will be a mix of versions. Now if I must upgrade all ... Regards, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin Sent: jeudi 8 décembre 2005 22:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OOH323 towards cisco gateway(2691)callsetupfails at q931: Mandatory information element ismissing (96) I thought the version was up to.7 or .8. In any case I am using the version in Asterisk-Addons 1.2.0. Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jacobso1 Sent: Thursday, December 08, 2005 1:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OOH323 towards cisco gateway (2691)callsetupfails at q931: Mandatory information element is missing (96) Hi, I upgraded my chan-ooh323 Same problem I was running 0.2, now 0.3 (that was the latest I did found) Do I need to upgrade asterisk too ? Up to 1.2.1 ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.13.12/193 - Release Date: 6/12/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.13.12/193 - Release Date: 6/12/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OOH323 towards cisco gateway (2691) callsetupfails at q931: Mandatory information element is missing (96)
Hi, I upgraded my chan-ooh323 Same problem I was running 0.2, now 0.3 (that was the latest I did found) Do I need to upgrade asterisk too ? Up to 1.2.1 ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin Sent: jeudi 8 décembre 2005 18:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OOH323 towards cisco gateway (2691) callsetupfails at q931: Mandatory information element is missing (96) Upgrade if you can. I remember submitting a report to the ooH323c developers about this some months ago and the fixed it right away. Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jacobso1 Sent: Thursday, December 08, 2005 8:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OOH323 towards cisco gateway (2691) call setupfails at q931: Mandatory information element is missing (96) Hi, I am using ooh323. I cannot setup a call towards a cisco gateway. The cisco rejects the call right away with : Cause value: Mandatory information element is missing (96) This is in the q931 part. Cisco ‘explanation’ Indicates that the equipment that is sending this code has received a message that is missing an information element that must be present in the message before that message can be processed. Show version gives : Cvs-head-06/21/05-23:51:26 Someone any clue ? H323.conf : ; Objective System's H323 Configuration example for Asterisk ; 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 asterisk 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 asterisk, 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 asterisk box, from where it will be routed as per dial plan. [general] ;Define the asetrisk server h323 endpoint ;The port asterisk should listen for incoming H323 connections. ;Default - 1720 port=1720 ;The dotted IP address asterisk should listen on for incoming H323 ;connections ;Default - tries to find out local ip address on it's own bindaddr=0.0.0.0 ;UPDATE this to proper ip address of your asterisk box ;Whether asterisk should use fast-start and tunneling for H323 connections. ;Default - yes faststart=yes h245tunneling=yes ;H323-ID to be used for asterisk server ;Default - Asterisk PBX h323id=TK_BRU_AST1 e164=100 ;CallerID to use for calls ;Default - Same as h323id callerid=TK_BRU_AST1 ;Whether this asterisk server will use gatekeeper. ;Default - DISABLE ;gatekeeper = DISCOVER ;gatekeeper = a.b.c.d gatekeeper = DISABLE ;Location for H323 log file ;Default - /var/log/asterisk/h323_log logfile=/var/log/asterisk/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 - default context=from-sip2 ;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds ;rtptimeout=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 = default ;The account code used by default for all clients. ;accountcode=h3230101 ;The codecs to be used for all clients. ;Default - ulaw ; ONLY ulaw, alaw, gsm, g729 and g723 (g723.1) are supported as of now disallow=all ;Note order of disallow/allow is important. allow=g729 allow=alaw allow=ulaw ; dtmf mode to be used by default for all clients. Only rfc2833 supported as ; of now. ;Default - rfc 2833 dtmfmode=rfc2833 ; User/peer/friend definitions: [TK_BRU_GW1] type=friend context=from-sip2 ip=195.xxx.yyy.zzz port=1720 disallow=all allow=g729 incominglimit=3 outgoinglimit=3 rtptimeout=60 dtmfmode=rfc2833
[Asterisk-Users] OOH323 towards cisco gateway (2691) call setup fails at q931: Mandatory information element is missing (96)
Hi, I am using ooh323. I cannot setup a call towards a cisco gateway. The cisco rejects the call right away with : Cause value: Mandatory information element is missing (96) This is in the q931 part. Cisco ‘explanation’ Indicates that the equipment that is sending this code has received a message that is missing an information element that must be present in the message before that message can be processed. Show version gives : Cvs-head-06/21/05-23:51:26 Someone any clue ? H323.conf : ; Objective System's H323 Configuration example for Asterisk ; 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 asterisk 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 asterisk, 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 asterisk box, from where it will be routed as per dial plan. [general] ;Define the asetrisk server h323 endpoint ;The port asterisk should listen for incoming H323 connections. ;Default - 1720 port=1720 ;The dotted IP address asterisk should listen on for incoming H323 ;connections ;Default - tries to find out local ip address on it's own bindaddr=0.0.0.0 ;UPDATE this to proper ip address of your asterisk box ;Whether asterisk should use fast-start and tunneling for H323 connections. ;Default - yes faststart=yes h245tunneling=yes ;H323-ID to be used for asterisk server ;Default - Asterisk PBX h323id=TK_BRU_AST1 e164=100 ;CallerID to use for calls ;Default - Same as h323id callerid=TK_BRU_AST1 ;Whether this asterisk server will use gatekeeper. ;Default - DISABLE ;gatekeeper = DISCOVER ;gatekeeper = a.b.c.d gatekeeper = DISABLE ;Location for H323 log file ;Default - /var/log/asterisk/h323_log logfile=/var/log/asterisk/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 - default context=from-sip2 ;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds ;rtptimeout=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 = default ;The account code used by default for all clients. ;accountcode=h3230101 ;The codecs to be used for all clients. ;Default - ulaw ; ONLY ulaw, alaw, gsm, g729 and g723 (g723.1) are supported as of now disallow=all ;Note order of disallow/allow is important. allow=g729 allow=alaw allow=ulaw ; dtmf mode to be used by default for all clients. Only rfc2833 supported as ; of now. ;Default - rfc 2833 dtmfmode=rfc2833 ; User/peer/friend definitions: [TK_BRU_GW1] type=friend context=from-sip2 ip=195.xxx.yyy.zzz port=1720 disallow=all allow=g729 incominglimit=3 outgoinglimit=3 rtptimeout=60 dtmfmode=rfc2833 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.13.12/193 - Release Date: 6/12/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
