[Asterisk-Users] hylafax
Good day all Can hylafax work with asterisk..and how I'm trying to find a way to send a fax over my E1 connection Please Help Altus ___ 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] * or X100P dropping analog calls
I also have x100p cards but ... Shouldn't signalling in the zapata.conf file be signalling=fks_ks instead of signalling=fxo_ks? I keep forgetting if you have to set fxo on zapata or zaptel... One thing that I see when I do a reload on my asterisc box is this always: Feb 23 02:36:47 WARNING[10343]: chan_zap.c:10015 setup_zap: Ignoring signalling -- Reconfigured channel 1, FXS Kewlstart signalling -- Reconfigured channel 2, FXS Kewlstart signalling -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Martes, 22 de Febrero de 2005 10:49 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * or X100P dropping analog calls Andrew Duey wrote: I have a * box running * version 1.0.3 with two X100P line cards in it and Cisco 7960 IP phones. Everything seems to work pretty well with the exeption that the system hangs up on phone calls for no apparent reason. It does this on both incoming and outgoing calls through the POTS line (currently only have one). The only thing in the asterisk console with maximum verbousity is -- Hungup 'Zap/1-1' and then the phones act like the other side hung up. The only thing that might be related is that I here beeping or chirping shortly before this happens (it happens randomly thorught calls every couple of minutes but seems to be more prevelent shortly before a disconnect). Things we've tried to correct the problem: -Remove all analog phones from analog line except the X100P -Got rid of off-brand cards and replaced with authentic X100p's -Setup 80mm case fan to blow over cards (the behavior seemed to be overheating related). Right now the signaling is set to: signalling=fxo_ks in the zapatel.conf file. Please let me know if anyone else has seen similar behavior (or if you need more information let me know what). Yes. It's frequently caused by busydetect=yes or callprogress=yes in /etc/asterisk/zapata.conf --Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Trunking capacity enforcement
Hello, I am trying to come up with a good way to enforce a limit on the number of simultaneous calls that can occupy an IAX trunk at any given time. I have searched around and so far can't locate a config option that would directly label a IAX trunk with a specific number to obey (is there one?). Based on examples for the SetGroup and CheckGroup commands, I am thinking of using SetGroup and CheckGroup in the dialplan using the same group namefor both incoming and outgoing calls (on one end of the trunk), hopefully resulting in a group count equal to the total of all active incoming and outgoing calls. Does this seem reasonable? If any of you are accomplishing this in other ways, please share your examples... Marty Mastera M3 Resources [EMAIL PROTECTED] Phone: 303.680.1283 x200 FAX: 206.666.1786 ___ 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] install BRIstuff on *@home?
Each release of BRIstuff is made for a specific * version. BRIstuff installer automatically downloads the correct version, patches and installs it. You should just run the install.sh and it will replace your current * installation. Your existing configuration (extensions.conf etc.) will not be changed. Bear in mind that you have to copy the BRI modules (qozap.ko and zaphfc.ko) manually to your /lib/modules/`uname -r`/misc directory. Erwin de Raad wrote: I'm still trying to install a HFC-s BRI card onto [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] .6 I'm new to this so I probably am overlooking the obvious. Can I just install BRIstuff onto a fresh [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] install? The BRIstuff installer downloads another * from Digium. Will this interfere with the @home install and must I comment out the Asterisk install in the BRIstuff install.sh file? Any pointers are much appreciated. Regards, Erwin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Repost: How do I install Skinny support for non sip cisco phones
I have a server setup that runs sip no problem. I want to try a cisco phone. how do I a) Tell if I have skinny support loaded b) Load it onto a debian system How about 'show modules' from the cli? Might look at contents of /etc/asterisk/modules.conf as well. ___ 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] I need an interactive automated telephone system - Is AGI my solution?
Hey Guys, I need to create an automated system that will allow people to call in, enter their code and be able to access all the various features (such as pre-recorded messages with news, sports...etc). Also need to be able to perform tasks based on buttons pressed on the phone. For example, I need a server monitoring system where it will say something like Press 1 for server load average, Press 2 for Memory usage, Press 3 for HDD usage, Press 4 to reboot the server. The responses will all be pre-recorded, but I just need to be able to interact with asterisk from PHP on the fly. I know AGI is a method for languages like PHP to interact with asterisk, but can it do this kind of stuff? Gabe ___ 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] IVR stats
Guys.. Im doing a simple IVR system with some menus but I was wondering, maybe it already does but does asterisk keep track of themenu hoices that each call did? for example, is a caller calls in and then hits 1,3,2,6 does that stay on some log file? __ Anton Krall ___ 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] I need an interactive automated telephone system -Is AGI my solution?
Would it be nice if it could also say stuff in realtime... Like using festival or something??? Anybody using festival in text to speech apps using php or something... Or even better, anybodu using festival with spanish??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Afana Sent: Miércoles, 23 de Febrero de 2005 03:07 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I need an interactive automated telephone system -Is AGI my solution? Hey Guys, I need to create an automated system that will allow people to call in, enter their code and be able to access all the various features (such as pre-recorded messages with news, sports...etc). Also need to be able to perform tasks based on buttons pressed on the phone. For example, I need a server monitoring system where it will say something like Press 1 for server load average, Press 2 for Memory usage, Press 3 for HDD usage, Press 4 to reboot the server. The responses will all be pre-recorded, but I just need to be able to interact with asterisk from PHP on the fly. I know AGI is a method for languages like PHP to interact with asterisk, but can it do this kind of stuff? Gabe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I need an interactive automated telephone system - Is AGI my solution?
On Wed, 2005-02-23 at 01:07 -0800, Gabriel Afana wrote: Hey Guys, I need to create an automated system that will allow people to call in, enter their code and be able to access all the various features (such as pre-recorded messages with news, sports...etc). Also need to be able to perform tasks based on buttons pressed on the phone. For example, I need a server monitoring system where it will say something like Press 1 for server load average, Press 2 for Memory usage, Press 3 for HDD usage, Press 4 to reboot the server. The responses will all be pre-recorded, but I just need to be able to interact with asterisk from PHP on the fly. I know AGI is a method for languages like PHP to interact with asterisk, but can it do this kind of stuff? First: In email, when you press reply to a message there is a header line that tells which message you where replying to. This helps threaded mail readers keep messages together that should be together and make reading mail more efficient for those of us with a clue, or at least a par or better mail reader. You responded to a mail with subject * or X100P dropping analog calls. The normal excuse of being lazy isn't valid for hitting reply and emptying the body and subject line as most all email clients will create a message when you click on the address in the same message you hit reply to. Second: Sounds either like a homework question, ie. for school, or work. You either need to spend the slight effort to do the work and leg work to build up your knowledge or you need to hire this done. IF you aren't willing to read the plentiful documentation that is scattered about, I suggest you visit the Wiki and consult a consultant. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Zaptel Red Alarm
Guys.. I just saw this for the first time... I did some google and wiki without any luck.. what does a red or yellow alarm mean in zaptel? Feb 23 02:54:16 WARNING[16890]: chan_zap.c:5865 handle_init_event: Detected alarm on channel 2: Red Alarm Feb 23 02:54:24 NOTICE[16890]: chan_zap.c:5860 handle_init_event: Alarm cleared on channel 2 This just happened by itself.. __ Anton Krall ___ 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] I need an interactive automated telephone system -Is AGI my solution?
I know AGI is a method for languages like PHP to interact with asterisk, but can it do this kind of stuff? Should be quite straight forward to implement your requirement. An AGI script using your favourite programming language is a way to go. We have implement all bits you mentioned so far except interacting with system statuses. Ta Senad ___ 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] Teleconferencing using Zapta cards.
Hi, I would like to use the asterisk box with zapta card to enable some conferencing. I would like to use only TDM connections without VoIP. I'd like also use the Meetme app. I have some questions: 1. Does any one use it for a few conference rooms at ones ? 2. Is it possible to restrict the number of users connected to one conference room ? Regards. Pawel. ___ 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] ISDN/SIP videophone gatewaying?
sure, but what about using asterisk? On Feb 22, 2005, at 12:39, googleplex wrote: google for inalp isdn sip gateway On Tue, 22 Feb 2005 12:23:39 +0100, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: Hi Is it, or could it be possible to gateway from ISDN videophones to IP videophoning with asterisk using libpri/zaptel etc? roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] * or X100P dropping analog calls
I have a * box running * version 1.0.3 with two X100P line cards in it and Cisco 7960 IP phones. Everything seems to work pretty well with the exeption that the system hangs up on phone calls for no apparent reason. It does this on both incoming and outgoing calls through the POTS line (currently only have one). The only thing in the asterisk console with maximum verbousity is -- Hungup 'Zap/1-1' and then the phones act like the other side hung up. The only thing that might be related is that I here beeping or chirping shortly before this happens (it happens randomly thorught calls every couple of minutes but seems to be more prevelent shortly before a disconnect). Things we've tried to correct the problem: -Remove all analog phones from analog line except the X100P -Got rid of off-brand cards and replaced with authentic X100p's -Setup 80mm case fan to blow over cards (the behavior seemed to be overheating related). Right now the signaling is set to: signalling=fxo_ks in the zapatel.conf file. Please let me know if anyone else has seen similar behavior (or if you need more information let me know what). Yes. It's frequently caused by busydetect=yes or callprogress=yes in /etc/asterisk/zapata.conf Or, at the top of zapata.conf, use: busydetect=yes busycount=6 Its known to have corrected the above problem. ___ 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] Zaptel Red Alarm
It means for some reason you lost your CO line for 10 Seconds. Either someone pulled the plug out by mistake or the Exchange line went away for 10 seconds. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anton Krall Sent: 23 February 2005 09:35 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Zaptel Red Alarm Guys.. I just saw this for the first time... I did some google and wiki without any luck.. what does a red or yellow alarm mean in zaptel? Feb 23 02:54:16 WARNING[16890]: chan_zap.c:5865 handle_init_event: Detected alarm on channel 2: Red Alarm Feb 23 02:54:24 NOTICE[16890]: chan_zap.c:5860 handle_init_event: Alarm cleared on channel 2 This just happened by itself.. __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone had a Cisco 7970 working with Asterisk?
Hi Guys, Subject says it all I guess. If so, can you post working config for me Thanks ___ 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] cdr_odbc logging insane integer values
Title: cdr_odbc logging insane integer values
I'm having a problem with * (tried both HEAD and STABLE). When logging with cdr_odbc through unixODBC to MySQL, I get insane integer values in the duration, billsec, disposition and amaflags fields. I have enabled MySQL logging, and that's the query I get:
INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield) VALUES ('2005-02-23 11:47:04','1234','1234','161','auth-out','SIP/xxx','','Hangup','',8589934594,17179869186,12884901892,3630868242827837443,'xxx','1109155624.0','161')
I've tried MySQL 4.1.7 or 4.1.10, with unixODBC 2.2.9 or 2.2.10, with MyODBC 3.51.11 or 2.50.39. All of these give the same error.
Logging to csv file works correctly. I have also tried inserting a debug line in cdr_odbc.c, to print out the values of cdr-duration etc, and they are all correct. The problem seems to happen somewhere between * and MySQL.
The only thing I have on this system which is different than all my other systems is the platform: x86_64 (AMD Opteron) instead of 32bit i686.
Anyone?
Thank you
-Manuel
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[Asterisk-Users] mixing sound files?
As soon as a call hits the asterisk a menu is played "Press 1 for ... and 2 for..." I have got the speech in different mp3 file and the music in different mp3 file. is there anyway to mix these two files? Kindest Muhammad Muzzamil Luqman ___ 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] mixing sound files?
is there anyway to mix these two files? Use the soxmix utility. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Muhammad Muzzamil Luqman Sent: February 23, 2005 11:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] mixing sound files? As soon as a call hits the asterisk a menu is played Press 1 for ... and 2 for... I have got the speech in different mp3 file and the music in different mp3 file. is there anyway to mix these two files? Kindest Muhammad Muzzamil Luqman ___ 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] Chanspy and current version of cvs
Hi everybody ! I had patched asterisk to install chanspy weeks ago and everything was ok. With the current version of cvs i am having failures when i try to apply the same patch and the url where i originally downloaded it seems no longer active. Is the patch any longer maintained or has it been replaced with another function. Thanks in advance Lamine ___ 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] ISDN/SIP videophone gatewaying?
Hi Roy, -Original Message- sure, but what about using asterisk? On Feb 22, 2005, at 12:39, googleplex wrote: google for inalp isdn sip gateway Asterisk currently doesn't understand the ISDN video side. If you use one of those gateways you probably could get it to interop with SIP videophones via Asterisk. Florian ___ 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] Help With Adit 600 Configuration
Sorry to have had to post this, But I need urgent help with configuring one adit 600 I picked up from e-bay. Issues. I cannot access the console port, I am using HyperTerminal with settings VT100, 9600, 8-N-1 I also do not have any user-manual so I am kind of stuck. Any help in getting me started would be really appreciated. Any default settings like Ethernet port address, that can help me connect to this Box. Anything at all Jon, Any Idea from you as well ? thanks bst rgds Etoenyo Ntumi ___ 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] Digium BRI or quad BRI
Hi there, quick question...do digium make any BRI cards (ISDN2) or even better a quad port BRI, maybe im going blind, but I cant see any on their website Cheers Gary ___ 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] Chanspy and current version of cvs
I don't know of any download source, sorry. However, if you get it working, please let me know - we need this :) Julian. Mamadou Lamine KA wrote: Hi everybody ! I had patched asterisk to install chanspy weeks ago and everything was ok. With the current version of cvs i am having failures when i try to apply the same patch and the url where i originally downloaded it seems no longer active. Is the patch any longer maintained or has it been replaced with another function. Thanks in advance Lamine ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Difference between E1 and PRI
Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ 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] Error connecting to remote mysql database.
I have this error when i try to conect to my remote mysql server: Host xxx.xxx.xxx.xxx is not allow to connect to this MySQL server. can some bady tell me what i have to do??? thanks in advance wert __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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] Difference between E1 and PRI
PRI comes in 2versions E1 European and T1 US E1 30 channels T1 23 channels On Wed, 2005-02-23 at 14:15, Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help With Adit 600 Configuration
On February 23, 2005 06:57 am, Etoenyo Ntumi wrote: Sorry to have had to post this, But I need urgent help with configuring one adit 600 I picked up from e-bay. Sorry we don't provide urgent help here. That we charge for. Don't leave things to the absolute last minute and then ask for immediate assistance due to your lack of planning. You had at least a few days from the time you paid for it until the courier shipped it, what were you doing during that time? Issues. I cannot access the console port, I am using HyperTerminal with settings VT100, 9600, 8-N-1 Are you using a regular or null modem cable? You need a regular one. Does your serial port work? i.e. if you short out pins 2 and 3 on the cable do you see what you're typing? They're not difficult ot set up at all, but you do need working equipment. :-) Also, carrier access has an incredible support site that they do not charge for. You do need to register with them but that's free. -A. ___ 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
Subject: [Asterisk-Users] Welltech with Asterisk Registration
Hi, All This is the technical support from Welltech. About the registration with the Asterisk, this is because the old sip stack version. In the new sip stack of our firmware, it could resolve this kind of problems with Asterisk. If you need the new testing firmware version for our CPE, please contact with your sales from Welltech. Thanks Message: 9 Date: Tue, 22 Feb 2005 16:28:27 -0800 From: Vice President - Lamsre [EMAIL PROTECTED] Subject: [Asterisk-Users] Welltech with Asterisk Registration To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Please help me, i can only able to register 1 port of my 6 port fxo (sip) with asterisk, it alway last one register. not all port. how to fix this proble. -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050222/5832b741/attachment-0001.htm -- ___ 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: Linux Bridge + QoS Shaper HOWTO available
On February 23, 2005 02:20 am, Jean-Michel Hiver wrote: NB: I use this modified script I found on the voip-info wiki. It makes a very noticeable difference... That's my rc.tc script. The latest and greatest version is at http://www.mixdown.ca/~andrew/dump/rc.tc. I really should put that at a permanent location on that site. :-) I'm glad it works well for you. It works well for me, too. -A. ___ 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] IVR stats
Guys.. Im doing a simple IVR system with some menus but I was wondering, maybe it already does but does asterisk keep track of themenu hoices that each call did? for example, is a caller calls in and then hits 1,3,2,6 does that stay on some log file? I doubt it, but you could always run off an AGI script to log it roy ___ 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] Difference between E1 and PRI
Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? Think of E-1 as Ethernet (transport) and PRI as IP (protocol). You could also think of E-1 as IP and PRI as TCP. You can also think of E-1 as TCP/IP and PRI as FTP. ___ 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] Voicemail as email attachment not working individually i.e. extensions specific
asterisk user wrote: Hello I am using asterisk 1.0.0, here i am facing one problem that the email-aatchment setting for each extesion is not working individually. When globally attach=yes is set the voicemail will be sent as attachment no matter for any extension if attach=no is set for it. Same in the case with if attach=no is set globally then attach=yes will not work if set for some particular extension. Upgrade to 1.0.5 ___ 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] Help With Adit 600 Configuration
Title: RE: [Asterisk-Users] Help With Adit 600 Configuration ;) thanks Andrew, and I do take the reprimand in good faith. I am using the Usual Cisco type roll-over cable. When you say regular do you mean straight through ?, and can you kindly confirm for me if the HyperTerminal settings I am using are OK?. I am definitely not using a null-modem cable, and yes the port works cos I use it every day on various Cisco gear I have called Carrier Access, I am just waiting for them to get in to the office, so I can register, any help in the mean time is still very much appreciated though thanks bst rgds Etoenyo Ntumi -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]] Sent: Wednesday, February 23, 2005 7:23 AM To: 'asterisk-users@lists.digium.com' Subject: Re: [Asterisk-Users] Help With Adit 600 Configuration On February 23, 2005 06:57 am, Etoenyo Ntumi wrote: Sorry to have had to post this, But I need urgent help with configuring one adit 600 I picked up from e-bay. Sorry we don't provide urgent help here. That we charge for. Don't leave things to the absolute last minute and then ask for immediate assistance due to your lack of planning. You had at least a few days from the time you paid for it until the courier shipped it, what were you doing during that time? Issues. I cannot access the console port, I am using HyperTerminal with settings VT100, 9600, 8-N-1 Are you using a regular or null modem cable? You need a regular one. Does your serial port work? i.e. if you short out pins 2 and 3 on the cable do you see what you're typing? They're not difficult ot set up at all, but you do need working equipment. :-) Also, carrier access has an incredible support site that they do not charge for. You do need to register with them but that's free. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 7, Issue 284
hello how to register with irc. i want to connect to #asterisk through x-chat thanks kamran __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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] Difference between E1 and PRI
E1 is a European T1. T1/E1 is the transport. PRI is the protocol. PRI on an T1 id 23B+D, PRI on an E1 is 30B+D. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, February 23, 2005 7:50 AM To: Eric Bishop; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between E1 and PRI Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? Think of E-1 as Ethernet (transport) and PRI as IP (protocol). You could also think of E-1 as IP and PRI as TCP. You can also think of E-1 as TCP/IP and PRI as FTP. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between E1 and PRI
On Wed, 23 Feb 2005, Eric Bishop wrote: I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? E1 is a serial line capable of 2048 Mbit. After channelization you have 31 usable 64 kbit channels. One channel (number 16) is used for signalling (even when using CAS) and the remaining 30 channels are available for voice. When you run ISDN PRI over the E1 the ISDN signalling is placed in the signalling channel of the E1. Peter ___ 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 BRI or quad BRI
Hi, -Original Message- Hi there, quick question...do digium make any BRI cards (ISDN2) or even better a quad port BRI, maybe im going blind, but I cant see any on their website They don't. If you are in need of a european ISDN2 type, see if http://www.junghanns.net/asterisk/page17.html helps you out. Florian ___ 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] Difference between E1 and PRI
Eric, E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe, which can be used for data, or can be split into 32 64Kbps telephone channels, or a mixture. If used for telephone channels, 30 of these channels can carry one telephone conversation each, and 2 carry signalling and timing information. T1 is similar to E1. It is used in North America. It is 1.544Mbps, and can carry 24 telephone channels, each of which can carry a telephone conversation (but see below). There are a number of protocols which can run on top of E1. Some of these are called CAS, Channel Associated Signalling. Examples are FXS loop start and EM wink start. They provide information such as the number that was called, and what state the call is in. They're limited in what information they can carry, and are slow to set up. A more modern protocol which overcomes these problems is ISDN. On E1, EuroISDN is the standard. On T1, there are different standards from different providers. DMS100, DMS250, NI1, and NI2 are common examples. ISDN uses one channel (called the D channel) for signalling call information. On E1, this is one of the 2 signalling channels, leaving 30 channels for voice (called B channels). On T1, there aren't any spare signalling channels, so one of the voice channels is used, leaving 23 B channels for voice. A PRI (Primary Rate ISDN) is simply an E1 or T1 with ISDN on top of it. ISDN gives fast, reliable call setup and hangup detection, and detailed information about the call. In the UK, PRI is also called ISDN30. An important extension to ISDN is Q.SIG, which provides extra signalling information that is used when connecting PBX systems. An alternative to PRI is BRI (Basic Rate ISDN), which is a cheaper system for small offices. It has 2 64Kbps B channels for voice, and 1 16Kbps D channel for signalling. It is sold as an alternative to analogue telephone lines. IN the UK, it is also called ISDN2e. I hope this answers your question! My company offers commercial support and installation services for PRI and Asterisk if you need help for specific scenarios. This email may form the basis of a future Integrics Tip. See: http://integrics.com/tips/ Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail as email attachment not working individually i.e. extensions specific
Hello I am using asterisk 1.0.0, here i am facing one problem that the email-aatchment setting for each extesion is not working individually. When globally attach=yes is set the voicemail will be sent as attachment no matter for any extension if attach=no is set for it. Same in the case with if attach=no is set globally then attach=yes will not work if set for some particular extension. Upgrade to 1.0.5 And if you don't/can't upgrade for whatever reason, here's another solution Set globally attach=yes and for each account that you don't want to send it, just don't put an email address. hth ___ 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] IVR stats
I´m just using the CDR user field to store all the IVR information Guillermo From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] IVR stats Date: Wed, 23 Feb 2005 13:47:12 +0100 MIME-Version: 1.0 (Apple Message framework v619.2) Received: from lists.digium.com ([69.16.138.164]) by mc1-f35.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Wed, 23 Feb 2005 04:49:06 -0800 Received: from [69.16.138.164] (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTPid B86C92FDE1B; Wed, 23 Feb 2005 06:47:12 -0600 (CST) Received: from psmtp.com (exprod5mx118.postini.com [64.18.0.32])by lists.digium.com (Postfix) with SMTP id 5DCD72FCDA4for asterisk-users@lists.digium.com;Wed, 23 Feb 2005 06:47:09 -0600 (CST) Received: from source ([62.73.241.54]) by exprod5mx118.postini.com([64.18.4.10]) with SMTP; Wed, 23 Feb 2005 07:47:15 EST Received: from [80.239.107.80] (unknown [80.239.107.80])by pebbles.fastcom.no (Postfix) with ESMTP id 8378A16C0for asterisk-users@lists.digium.com;Wed, 23 Feb 2005 13:33:08 +0100 (NFT) X-Message-Info: N4u0pqWW+O22+yGpcQFDItgtUK3Il2KvHnyK4r6jjDs= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com References: [EMAIL PROTECTED] X-Mailer: Apple Mail (2.619.2) X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [65/3] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 23 Feb 2005 12:49:06.0761 (UTC) FILETIME=[0FA42B90:01C519A6] Guys.. Im doing a simple IVR system with some menus but I was wondering, maybe it already does but does asterisk keep track of themenu hoices that each call did? for example, is a caller calls in and then hits 1,3,2,6 does that stay on some log file? I doubt it, but you could always run off an AGI script to log it roy ___ 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 _ Las mejores tiendas, los precios mas bajos, entregas en todo el mundo, YupiMSN Compras: http://latam.msn.com/compras/ ___ 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] Help With Adit 600 Configuration
Title: RE: [Asterisk-Users] Help With Adit 600 Configuration Thanks Guys, Straight through cable works just fine. bst rgds Etoenyo Ntumi -Original Message- From: Etoenyo Ntumi Sent: Wednesday, February 23, 2005 7:51 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help With Adit 600 Configuration ;) thanks Andrew, and I do take the reprimand in good faith. I am using the Usual Cisco type roll-over cable. When you say regular do you mean straight through ?, and can you kindly confirm for me if the HyperTerminal settings I am using are OK?. I am definitely not using a null-modem cable, and yes the port works cos I use it every day on various Cisco gear I have called Carrier Access, I am just waiting for them to get in to the office, so I can register, any help in the mean time is still very much appreciated though thanks bst rgds Etoenyo Ntumi -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]] Sent: Wednesday, February 23, 2005 7:23 AM To: 'asterisk-users@lists.digium.com' Subject: Re: [Asterisk-Users] Help With Adit 600 Configuration On February 23, 2005 06:57 am, Etoenyo Ntumi wrote: Sorry to have had to post this, But I need urgent help with configuring one adit 600 I picked up from e-bay. Sorry we don't provide urgent help here. That we charge for. Don't leave things to the absolute last minute and then ask for immediate assistance due to your lack of planning. You had at least a few days from the time you paid for it until the courier shipped it, what were you doing during that time? Issues. I cannot access the console port, I am using HyperTerminal with settings VT100, 9600, 8-N-1 Are you using a regular or null modem cable? You need a regular one. Does your serial port work? i.e. if you short out pins 2 and 3 on the cable do you see what you're typing? They're not difficult ot set up at all, but you do need working equipment. :-) Also, carrier access has an incredible support site that they do not charge for. You do need to register with them but that's free. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phone problem
We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10) and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are configured in TE mode and connected to the PSTN; the other 8 are in NT mode and connected to isdn phones. the other outbound calls to PSTN are fine, however, when we call cellular phones, often audio is one-way (i.e.: the cell phone user can not hear, while the speaker at the internal side hears perfectly. CPU usage is quite low, and asterisk -rvvv does not show anything particular Any suggestion ___ 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] List tips for new subscribers
To all new subscribers: This is not a friendly list, and the members are mostly sick of giving away knowledge they reckon someone should pay for. Although you will find friendly replies here and there, you'll have to wade through many obnoxious replies from the * elite. You will have to help yourself until you know enough to not need this list anymore. Then and only then, can join and show off your technical prowess to new subscribers. Nobody here really wants to help you in learning asterisk, since everyone here has their own blackbox solution sitting on a shopshelf somewhere, and they would therefore be helping the aspiring competition. What really ticks us off is that we had to figure out so much ourselves, and you won't go through the pain we did. So we try to force you. This list is similar to many of the *BSD user lists. These are actually not for users, but for hackers/experts interested in the more trivial challenges of their chosen technology. I suggest you look at the archives, I know there is no search option on the pages, but its not our problem. Cant you use google? I know google doesn't give you much but you must figure out *how* to ask it. Its been mentioned before on this list, so just download every digest of every month, and you can search manually for the answer. Once you have done this you'll know how to search these lists using google. Further more look at voip-info.org. We know you've been there. We know you didn't find your answer there. We don't care. This is Open Source: so fix it yourself. You get what you pay for: You want a pbx? Buy one - we suggest froogle. Hope This helps Teddy Versfeld ___ 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] Error connecting to remote mysql database.
On Wed, Feb 23, 2005 at 04:21:04AM -0800, R A said: I have this error when i try to conect to my remote mysql server: Host xxx.xxx.xxx.xxx is not allow to connect to this MySQL server. can some bady tell me what i have to do??? This has nothing to do with Asterisk. The error message tells you the problem. The manual tells you the solution. See: http://dev.mysql.com/doc/mysql/en/access-denied.html and: http://dev.mysql.com/doc/mysql/en/adding-users.html If you need further assistance, please use the mysql users mailing list. ___ 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] Help With Adit 600 Configuration
When you say regular do you mean straight through ?, and can you kindly confirm for me if the HyperTerminal settings I am using are OK?. I am definitely not using a null-modem cable, and yes the port works cos I use it every day on various Cisco gear I have called Carrier Access, I am just waiting for them to get in to the office, so I can register, any help in the mean time is still very much appreciated though Press Enter several times after connecting to the adit 600, because it doesn't send any login prompt. I copied these setup instructions from the adit manual: Setting up a CLI Connection If connecting via RS-232, the port settings should be set to: Bits per second: 9600 Data bits: 8 Parity: None Stop bits: 1 Flow control: None Set your Terminal Emulation to: VT100 NOTE: When using Tera Term TCP/IP, CLI commands will not be recognized until the following setup is completed. In Tera Term go to Setup/Terminal. Set the New-line/Transmit valueCR+LF. ___ 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 BRI or quad BRI
Thanks Florian, that's great, is this card (junghanns QuadBRI) really stable with * ? Do you or anybody else have any experiences with this card and also is it ok to run multiple cards in one machine cheers -Original Message- From: Florian Overkamp [mailto:[EMAIL PROTECTED] Sent: 23 February 2005 12:58 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Digium BRI or quad BRI Hi, -Original Message- Hi there, quick question...do digium make any BRI cards (ISDN2) or even better a quad port BRI, maybe im going blind, but I cant see any on their website They don't. If you are in need of a european ISDN2 type, see if http://www.junghanns.net/asterisk/page17.html helps you out. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] List tips for new subscribers
RIGHT ON! Too bad you also didn't post in HTML as well Perhaps this list needs to be split? One for the folks who simply want to get it working, and another for the self appointed list police who want to be rude and nasty and are only interested in feeding their own egos, all the while demonstrating their ignorance in basic Telephony? Unrepentant top poster [EMAIL PROTECTED] wrote: To all new subscribers: This is not a friendly list, and the members are mostly sick of giving away knowledge they reckon someone should pay for. Although you will find friendly replies here and there, you'll have to wade through many obnoxious replies from the * elite. You will have to help yourself until you know enough to not need this list anymore. Then and only then, can join and show off your technical prowess to new subscribers. Nobody here really wants to help you in learning asterisk, since everyone here has their own "blackbox" solution sitting on a shopshelf somewhere, and they would therefore be helping the aspiring competition. What really ticks us off is that we had to figure out so much ourselves, and you won't go through the pain we did. So we try to force you. This list is similar to many of the *BSD user lists. These are actually not for users, but for hackers/experts interested in the more trivial challenges of their chosen technology. I suggest you look at the archives, I know there is no search option on the pages, but its not our problem. Cant you use google? I know google doesn't give you much but you must figure out *how* to ask it. Its been mentioned before on this list, so just download every digest of every month, and you can search manually for the answer. Once you have done this you'll know how to search these lists using google. Further more look at voip-info.org. We know you've been there. We know you didn't find your answer there. We don't care. This is Open Source: so fix it yourself. You get what you pay for: You want a pbx? Buy one - we suggest froogle. Hope This helps Teddy Versfeld ___ 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] storing cdr in two databases
Hi, Is it possible to send CDR to a database (cdr_mysql.so for example) and to files (cdr_csv.so) ? As soon as I activated CDR writes to mysql, Master.csv stopped to grow, and since CDRs seems to be registered in a linked list in cdr.c I thought it was possible... TIA, -- Ludovic DROLEZ Linbox / FreeALter Soft 152 rue de Grigy - Technopole Metz 2000 57070 METZ tel : 03 87 50 87 90fax : 03 87 75 19 26 ___ 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] Help With Adit 600 Configuration
On February 23, 2005 07:51 am, Etoenyo Ntumi wrote: ;) thanks Andrew, and I do take the reprimand in good faith. I am using the Usual Cisco type roll-over cable. Nope this isn't Cisco gear. You want a regular 9-pin serial cable (straight through). When you say regular do you mean straight through ?, and can you kindly confirm for me if the HyperTerminal settings I am using are OK?. I am definitely not using a null-modem cable, and yes the port works cos I use it every day on various Cisco gear Those settings (9600n81) are fine. No flow control. I have called Carrier Access, I am just waiting for them to get in to the office, so I can register, any help in the mean time is still very much appreciated though Good good. Basically what you do on an Adit600 is connect T1 channels to card channels. There are two T1s, and up to 6 cards. Each card has 8 channels, so the typical connection is something like this: connect a:1:1-8 1:1-8 that connects T1#1 (whcih is in slot 'A') channels 1-8 to card #1's channels 1-8. There is pretty good context-sensitive help built in. Just type ? where you want help. Also don't forget to set the clocking and framing or you'll likely get frame slips and odd little behaviours. -A. ___ 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] List tips for new subscribers
ok, not that i'm such an expert myself, but 1. there's a big difference between newbies asking specific question and the i want asterisk to run my life, make me coffee, and solve my problems, does asterisk do that? questions that are appearing lately. I'm not a member of the list police and they annoy the hell out of me. 2. many of the list police are active in the development process well, so your remarkably clever comments about the lack of help are uncalled for and untrue. People will help you, but they won't hold your hand. If you want your hand held, then hire a consultant. 3. get a gmail account and your search issues on the mailing list are over. In addition, the remarkable new gmail system doesn't mangle your email with HTML tags, rendering them readable to all. how revolutionary. the downside is, no smileys. (oh the horror.). 4.almost everyone here has been quite helpful. once or twice i didn't follow netiquette (posted once without a subject by mistake) and quite rightly got called for it. If your ego is so fragile a dressing-down on a email list from people you don't know bothers you, you have issues. And specifically in this case, recyclying a subject line that has nothing to do with your email is just lazy and screws up threads. seriously, get over it. -yair ___ 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] List tips for new subscribers
On February 23, 2005 08:25 am, [EMAIL PROTECTED] wrote: This is not a friendly list, and the members are mostly sick of giving away knowledge they reckon someone should pay for. Although you will find friendly replies here and there, you'll have to wade through many obnoxious replies from the * elite. You will have to help yourself until you know enough to not need this list anymore. Then and only then, can join and show off your technical prowess to new subscribers. Please remove your head from your arse and reread what I wrote. THEN hit the reply button and post something relevant. You get what you pay for: You want a pbx? Buy one - we suggest froogle. Feel free to unsubscribe at any time, Teddy. I can't imagine how much this list is costing you in time and frustration. -A. ___ 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] Zaptel Red Alarm
could be that someone plugged out ur telephone line and plugged it back in. ___ 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] SpanDSP - Still can't send
BTW what versions of libtiff spandsp u using, cuz i can't recieve faxes at all. ___ 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] Zaptel Red Alarm
Is there a way for asterisk to notify you of this? Send an email? Send a page? Call you? -Matthew - Original Message - From: Giovanni Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 23, 2005 8:26 AM Subject: Re: [Asterisk-Users] Zaptel Red Alarm could be that someone plugged out ur telephone line and plugged it back in. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between E1 and PRI
Alistair- Good writeup! Question regarding Q.SIG: Can it be used to solve the problem of signaling a remote switch to take a call back and extend it to another channel instead? This, as you know, is always a challenge when using IVR in a call centre environment, when one wants to extend an IVR call to a live operator without holding up channels in the IVR. Regards, Scott Stingel Emerging Voice Technology, Inc. www.evtmedia.com Alistair Cunningham wrote: Eric, E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe, which can be used for data, or can be split into 32 64Kbps telephone channels, or a mixture. If used for telephone channels, 30 of these channels can carry one telephone conversation each, and 2 carry signalling and timing information. T1 is similar to E1. It is used in North America. It is 1.544Mbps, and can carry 24 telephone channels, each of which can carry a telephone conversation (but see below). There are a number of protocols which can run on top of E1. Some of these are called CAS, Channel Associated Signalling. Examples are FXS loop start and EM wink start. They provide information such as the number that was called, and what state the call is in. They're limited in what information they can carry, and are slow to set up. A more modern protocol which overcomes these problems is ISDN. On E1, EuroISDN is the standard. On T1, there are different standards from different providers. DMS100, DMS250, NI1, and NI2 are common examples. ISDN uses one channel (called the D channel) for signalling call information. On E1, this is one of the 2 signalling channels, leaving 30 channels for voice (called B channels). On T1, there aren't any spare signalling channels, so one of the voice channels is used, leaving 23 B channels for voice. A PRI (Primary Rate ISDN) is simply an E1 or T1 with ISDN on top of it. ISDN gives fast, reliable call setup and hangup detection, and detailed information about the call. In the UK, PRI is also called ISDN30. An important extension to ISDN is Q.SIG, which provides extra signalling information that is used when connecting PBX systems. An alternative to PRI is BRI (Basic Rate ISDN), which is a cheaper system for small offices. It has 2 64Kbps B channels for voice, and 1 16Kbps D channel for signalling. It is sold as an alternative to analogue telephone lines. IN the UK, it is also called ISDN2e. I hope this answers your question! My company offers commercial support and installation services for PRI and Asterisk if you need help for specific scenarios. This email may form the basis of a future Integrics Tip. See: http://integrics.com/tips/ Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users 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] Using as FAX 100% IP
Hello, Before putting any effort, I would like to know if somebody has successfully run asterisk receiving FAXs in IP and sending them out in IP as well? If yes using which components please? Any help is greatly appreciated ! ___ 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 BRI or quad BRI
Hi Gary, -Original Message- Thanks Florian, that's great, is this card (junghanns QuadBRI) really stable with * ? Do you or anybody else have any experiences with this card and also is it ok to run multiple cards in one machine We have built and use systems with one QuadBRI and one TE410P card, works fine. I know of others who use multiple QuadBRI cards in a single box. I guess your answer is plain and simple: yes, and yes. Florian ___ 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] A bit of a survey: What do do if you need morethan 4 C.O. lines
I'm was having a couple issues also, mainly callerid when turned on was crashing asterisk, but its was my fault still. But does any of the digium cards beside x100p offer redundancy. ___ 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] List tips for new subscribers
Yair wrote: 1. there's a big difference between newbies asking specific question and the i want asterisk to run my life, make me coffee, and solve my problems, does asterisk do that? questions that are appearing lately. Who else is going to tell them that they need a wife? Do we have to be rude about it? 2. many of the list police are active in the development process well, so your remarkably clever comments about the lack of help are uncalled for and untrue. People will help you, but they won't hold your hand. If you want your hand held, then hire a consultant. Look, I appreciate your display of respect to the asterisk co-developers, but the lack of respect shown to new members here is appalling. I'm active on many lists, and this place is not the friendliest. 3. get a gmail account and your search issues on the mailing list are over. In addition, the remarkable new gmail system doesn't mangle your email with HTML tags, rendering them readable to all. how revolutionary. the downside is, no smileys. (oh the horror.). So will you send me an invite? After I accepted it, please forward me the mails I've missed up to today. Maybe we can add a gmail invite donation mechanism on the asterisk support page. 4.almost everyone here has been quite helpful. once or twice i didn't follow netiquette (posted once without a subject by mistake) and quite rightly got called for it. If your ego is so fragile a dressing-down on a email list from people you don't know bothers you, you have issues. When was that, back in 2001? Yes, I do have issues. I think this list should split into something like Asterisk-oldschool-elite, and Asterisk-users-friendly And specifically in this case, recyclying a subject line that has nothing to do with your email is just lazy and screws up threads. Oh beg I your pardon, but I didn't have access to previous messages to spot that this subject was already taken. Is it so impossible to believe that I typed that subject myself? Regards Teddy ___ 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] List tips for new subscribers
Andrew wrote: Please remove your head from your arse and reread what I wrote. THEN hit the reply button and post something relevant. What the hell makes you think this post had anything to do with you? Thanks for being polite! Teddy ___ 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] Using as FAX 100% IP
I'd say that would depend on the configuration you are considering. We have a number of fax machines running off of sipura spa-2000's that connect to a remote asterisk server and terminate to the pstn via voip as well. I'd say it's about 90% reliable at this point. However, we've noticed quite a bit of variability around the quality of the connection and underlying provider you are using for termination - so your results will vary. It also seems a couple of other tricks help: (1)put an ADSL filter at the fax machine end, this seems to help filter out noise from the signal and slow down the analog modem (not sure if this is an old maid's tale or not, but seems to help). (2)if you are using the sipura as your ATA, disable all of the fax detection. Just make sure you are only allowing ULAW as your codec. Other than that it's pretty straightforward. If you are looking to do something to send receive to files, etc. we haven't had any luck getting something along those lines to work w/pure voip. -Steve Hakem Taourchi wrote: Hello, Before putting any effort, I would like to know if somebody has successfully run asterisk receiving FAXs in IP and sending them out in IP as well? If yes using which components please? Any help is greatly appreciated ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Difference between E1 and PRI
Scott, Do a search on Tromboning I have no idea if asterisk is capable of doing this but I remember this was a feature introduce into Fujitsu Qsig stack in or about 94-95 which solved a heap of customer problems at the time so I remember it was a big deal. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: Wednesday, February 23, 2005 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between E1 and PRI Alistair- Good writeup! Question regarding Q.SIG: Can it be used to solve the problem of signaling a remote switch to take a call back and extend it to another channel instead? This, as you know, is always a challenge when using IVR in a call centre environment, when one wants to extend an IVR call to a live operator without holding up channels in the IVR. Regards, Scott Stingel Emerging Voice Technology, Inc. www.evtmedia.com Alistair Cunningham wrote: Eric, E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe, which can be used for data, or can be split into 32 64Kbps telephone channels, or a mixture. If used for telephone channels, 30 of these channels can carry one telephone conversation each, and 2 carry signalling and timing information. T1 is similar to E1. It is used in North America. It is 1.544Mbps, and can carry 24 telephone channels, each of which can carry a telephone conversation (but see below). There are a number of protocols which can run on top of E1. Some of these are called CAS, Channel Associated Signalling. Examples are FXS loop start and EM wink start. They provide information such as the number that was called, and what state the call is in. They're limited in what information they can carry, and are slow to set up. A more modern protocol which overcomes these problems is ISDN. On E1, EuroISDN is the standard. On T1, there are different standards from different providers. DMS100, DMS250, NI1, and NI2 are common examples. ISDN uses one channel (called the D channel) for signalling call information. On E1, this is one of the 2 signalling channels, leaving 30 channels for voice (called B channels). On T1, there aren't any spare signalling channels, so one of the voice channels is used, leaving 23 B channels for voice. A PRI (Primary Rate ISDN) is simply an E1 or T1 with ISDN on top of it. ISDN gives fast, reliable call setup and hangup detection, and detailed information about the call. In the UK, PRI is also called ISDN30. An important extension to ISDN is Q.SIG, which provides extra signalling information that is used when connecting PBX systems. An alternative to PRI is BRI (Basic Rate ISDN), which is a cheaper system for small offices. It has 2 64Kbps B channels for voice, and 1 16Kbps D channel for signalling. It is sold as an alternative to analogue telephone lines. IN the UK, it is also called ISDN2e. I hope this answers your question! My company offers commercial support and installation services for PRI and Asterisk if you need help for specific scenarios. This email may form the basis of a future Integrics Tip. See: http://integrics.com/tips/ Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between E1 and PRI
Scott, Yes, and this is one of the principal reasons people choose Q.SIG. I've worked on quite a few large voicemail servers, and these tend to do a lot of transfers for follow-me and operator features. Q.SIG support can significantly reduce the number of telephony channels needed, as not only are there zero channels in use rather than two during the transfer, but transferred calls last significantly longer on average than calls to leave or retrieve messages. You do need to check that the remote end supports this; some older PBXs only support parts of the Q.SIG standard. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Scott Stingel wrote: Alistair- Good writeup! Question regarding Q.SIG: Can it be used to solve the problem of signaling a remote switch to take a call back and extend it to another channel instead? This, as you know, is always a challenge when using IVR in a call centre environment, when one wants to extend an IVR call to a live operator without holding up channels in the IVR. Regards, Scott Stingel Emerging Voice Technology, Inc. www.evtmedia.com Alistair Cunningham wrote: Eric, E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe, which can be used for data, or can be split into 32 64Kbps telephone channels, or a mixture. If used for telephone channels, 30 of these channels can carry one telephone conversation each, and 2 carry signalling and timing information. T1 is similar to E1. It is used in North America. It is 1.544Mbps, and can carry 24 telephone channels, each of which can carry a telephone conversation (but see below). There are a number of protocols which can run on top of E1. Some of these are called CAS, Channel Associated Signalling. Examples are FXS loop start and EM wink start. They provide information such as the number that was called, and what state the call is in. They're limited in what information they can carry, and are slow to set up. A more modern protocol which overcomes these problems is ISDN. On E1, EuroISDN is the standard. On T1, there are different standards from different providers. DMS100, DMS250, NI1, and NI2 are common examples. ISDN uses one channel (called the D channel) for signalling call information. On E1, this is one of the 2 signalling channels, leaving 30 channels for voice (called B channels). On T1, there aren't any spare signalling channels, so one of the voice channels is used, leaving 23 B channels for voice. A PRI (Primary Rate ISDN) is simply an E1 or T1 with ISDN on top of it. ISDN gives fast, reliable call setup and hangup detection, and detailed information about the call. In the UK, PRI is also called ISDN30. An important extension to ISDN is Q.SIG, which provides extra signalling information that is used when connecting PBX systems. An alternative to PRI is BRI (Basic Rate ISDN), which is a cheaper system for small offices. It has 2 64Kbps B channels for voice, and 1 16Kbps D channel for signalling. It is sold as an alternative to analogue telephone lines. IN the UK, it is also called ISDN2e. I hope this answers your question! My company offers commercial support and installation services for PRI and Asterisk if you need help for specific scenarios. This email may form the basis of a future Integrics Tip. See: http://integrics.com/tips/ Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP - Still can't send
Neither can I. Check out my bug for my list of versions: http://www.opencall.org/mantis/bug_view_page.php?bug_id=029 -Matthew - Original Message - From: Giovanni Powell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 23, 2005 8:28 AM Subject: Re: [Asterisk-Users] SpanDSP - Still can't send BTW what versions of libtiff spandsp u using, cuz i can't recieve faxes at all. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] List tips for new subscribers
On February 23, 2005 09:43 am, [EMAIL PROTECTED] wrote: And specifically in this case, recyclying a subject line that has nothing to do with your email is just lazy and screws up threads. Oh beg I your pardon, but I didn't have access to previous messages to spot that this subject was already taken. Is it so impossible to believe that I typed that subject myself? You did type it yourself, but you replied to a message in a thread and erased everything, thus screwing up the threading. I think that's what he was referring to. Since you are so vocal, I do have to ask, is hitting the reply button and then erasing everything faster than simply clicking on the Asterisk-users@lists.digium.com email address to start a new, blank message to the list? I've never gotten a clear answer. It seems like a hell of a lot of work to reply to a message, erase everything and start anew. -A. ___ 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] List tips for new subscribers
On Wed, 23 Feb 2005 [EMAIL PROTECTED] wrote: 2. many of the list police are active in the development process well, so your remarkably clever comments about the lack of help are uncalled for and untrue. People will help you, but they won't hold your hand. If you want your hand held, then hire a consultant. Look, I appreciate your display of respect to the asterisk co-developers, but the lack of respect shown to new members here is appalling. I'm active on many lists, and this place is not the friendliest. New Asterisk users should try to help themselves first. Perhaps we should create a list asterisk-newbies for thos who do not want to go through the effort of reading up enough to ask well formulated questions. If you ask a question that even google answers on the first try then you ought to be chewed out in public. If you ask a question that is answered in the sparse documentation there is then you ought to be chewed out in public. This list is for discussions among users of Asterisk, not a getting started hotline for beginners. Beginners learn by reading documentation and examining the sample files included. Peter ___ 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] Using as FAX 100% IP
On February 23, 2005 09:31 am, Hakem Taourchi wrote: Hello, Before putting any effort, I would like to know if somebody has successfully run asterisk receiving FAXs in IP and sending them out in IP as well? No. Not without t.38. Googling for asterisk fax IP site:lists.digium.com should give you lots of the why. :-) -A. ___ 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] List tips for new subscribers
On February 23, 2005 10:03 am, Peter Svensson wrote: New Asterisk users should try to help themselves first. Perhaps we should create a list asterisk-newbies for thos who do not want to go through the effort of reading up enough to ask well formulated questions. That already exists. asterisk-biz. :-) -A. ___ 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] Mandrake CAPI EPIA!
Looked at the mISDN bits and pieces, which looked promising, so decided
to take the plung and go for Mandrake 10.1 (Kernel 2.6.8.1) and well -
What a disaster!
I should probably have mentioned I am using an EPIA 5000/classic
motherboard, which between Mandrake 10.1 and the motherboard come up
with an i686 processor which clearly it isn't. Anyway after numerous
rebuilds etc, I have made various changes to the makefiles for libpri
and * and they believe it is now an i586 and asterisk loads fine.
Trying the get back to position of a running * PBX, I tried to install
the zaptel drivers, using the following process -
CD zaptel.
Make linux26
Make install
When I modprobe zaptel I get the following errors -
[EMAIL PROTECTED] zaptel-1.0.4]# modprobe zaptel
FATAL: Error inserting zaptel
(/lib/modules/2.6.8.1-12mdk-i586-up-1GB/misc/zaptel.ko): Invalid module
format
When I modprobe wcfxo I get the following errors -
[EMAIL PROTECTED] zaptel-1.0.4]# modprobe wcfxo
WARNING: Error inserting zaptel
(/lib/modules/2.6.8.1-12mdk-i586-up-1GB/misc/zaptel.ko): Invalid module
format
WARNING: Error inserting zaptel
(/lib/modules/2.6.8.1-12mdk-i586-up-1GB/misc/zaptel.ko): Invalid module
format
FATAL: Error inserting wcfxo
(/lib/modules/2.6.8.1-12mdk-i586-up-1GB/misc/wcfxo.ko): Invalid module
format
FATAL: Error running install command for wcfxo
I picked up the following article
http://lists.digium.com/pipermail/asterisk-users/2004-November/072043.ht
ml and made the changes described, although i'm still getting the same
errors.
I think there may be a problem with the kernal description/location
mismatch e.g. -
/lib/modules/2.6.8.1-12mdk-i586-up-1GB as opposed to the source being
/usr/src/linux-2.6.8.1-12mdk
Can anyone help me?
and at least i'm learning!
Ray
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: 22 February 2005 03:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Mandrake CAPI
I've been using FC2 with Kernel 2.6.9, the hardest thing for me was
getting my capi startup script right, you should not have any capi
related stuff in modprobe.conf. I have included my startup script. If
you are using a DID or Point to Point line for the Fritz! then change
protocol=2 to protocol=34 for the avmfritz driver. The mISDN fritz!
driver will support up to four cards and I am successfully using both a
Eicon Diva Server 4-BRI card (With melware drivers) and Fritz! card in
the same system. The Fritz! must be loaded first with the capi script,
followed by divas_cfg for the Diva card. The divas_cfg script must have
its own modprobe capi line removed or commented out.
Craig
#!/bin/bash
#
# System startup script for the isdn-capi subsystem
case $1 in
start)
echo -n Starting mISDN and CAPI
modprobe capi
modprobe mISDN_core
modprobe mISDN_l1
modprobe mISDN_l2
modprobe l3udss1
modprobe mISDN_capi
modprobe mISDN_isac
modprobe avmfritz protocol=2
;;
stop)
echo -n Stopping mISDN and CAPI
rmmod avmfritz
rmmod mISDN_isac
rmmod mISDN_capi
rmmod l3udss1
rmmod mISDN_l2
rmmod mISDN_l1
rmmod mISDN_dtmf
rmmod mISDN_core
rmmod capi
rmmod kernelcapi
;;
restart)
$0 stop
$0 start
;;
*)
echo Usage:$0{start|stop|restart}
;;
esac
exit 0
- Original Message -
From: Razza [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, February 22, 2005 3:02 AM
Subject: RE: [Asterisk-Users] Mandrake CAPI
I was looking at the exercise as a bit of Linux lerning for myself, so
I guess Mandrake 10.1 and mISDN? Does anyone have working examples?
Ray
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig
Guy
Sent: 20 February 2005 23:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Mandrake CAPI
Or you could go to a 2.6 kernel and use the mISDN drivers.
Craig
- Original Message -
From: Razza [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, February 20, 2005 8:00 PM
Subject: [Asterisk-Users] Mandrake CAPI
All,
I have been trying to get CAPI4Linux working on my machine and being
frank am failing miserably! I am looking for any help available, I
am real newbie (so please be gentle) and choose to run Mandrake 9.2
as it
appears quite friendly (or so I thought!).
I have been following the guidance found at
http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI for
the AVM card (actually I have a BT Speedway - apparently the same
thing).
I guess the best approach is to detail what I have done in tandem
with
the guidance? So here we go -
Type -
# modprobe capi
Great! I get no response (which is expected!), so move to step 2
RE : [Asterisk-Users] Using as FAX 100% IP
Thank you ver much for this help Steven What I am planning is this: 1-) Receive fax on a DID that is being routed in IP to the asterisk server; 2-) Based on the rule on that incoming fax, Asterisk needs to capture it, store it as pdf file and e-mail it to a predefined destinoatin (based on DID); 3-) If 2 is not possible, then send fax to PSTN destination using voip; Did you use this http://tafm.sourceforge.net/ to make ATA sepura work ? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Steven Kokinos Envoyé : mercredi 23 février 2005 15:46 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Using as FAX 100% IP I'd say that would depend on the configuration you are considering. We have a number of fax machines running off of sipura spa-2000's that connect to a remote asterisk server and terminate to the pstn via voip as well. I'd say it's about 90% reliable at this point. However, we've noticed quite a bit of variability around the quality of the connection and underlying provider you are using for termination - so your results will vary. It also seems a couple of other tricks help: (1)put an ADSL filter at the fax machine end, this seems to help filter out noise from the signal and slow down the analog modem (not sure if this is an old maid's tale or not, but seems to help). (2)if you are using the sipura as your ATA, disable all of the fax detection. Just make sure you are only allowing ULAW as your codec. Other than that it's pretty straightforward. If you are looking to do something to send receive to files, etc. we haven't had any luck getting something along those lines to work w/pure voip. -Steve Hakem Taourchi wrote: Hello, Before putting any effort, I would like to know if somebody has successfully run asterisk receiving FAXs in IP and sending them out in IP as well? If yes using which components please? Any help is greatly appreciated ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] List tips for new subscribers
Peter Svensson wrote: New Asterisk users should try to help themselves first. Perhaps we should create a list asterisk-newbies for thos who do not want to go through the effort of reading up enough to ask well formulated questions. Strikes me as a very good suggestion. Perhaps all subscribers to it should be provided with a welcome note including key URLs - voip-info, a dedicated noob faq, the list archives and Google among 'em. -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ 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] List tips for new subscribers
*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way. ___ 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] List tips for new subscribers
to the list? I've never gotten a clear answer. It seems like a hell of a lot of work to reply to a message, erase everything and start anew. Oh I'm sorry. This is the first list I've joined where this is such a big issue! Forgive me for not having your superior understanding of mail clients, and/or list servers! Maybe I did it because I do it in all the lists I've posted and youre the first to complain. Maybe I did it because I use squirrelmail as my mail client, and clicking on an email address opens up evolution which I don't want to use. But thanks for the reply, thanks for being polite, thanks for kicking my head in for not doing it exactly like you would want it done . Get a punching bag, watch some WWF - Whatever. Just do SOMETHING to relax. PLEASE! ___ 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] List tips for new subscribers
Please remove your head from your arse and reread what I wrote. THEN hit the reply button and post something relevant. What the hell makes you think this post had anything to do with you? Children! PLEASE! You'll ALL be going to bed with no tea at this rate. I don't care WHO started it, you can ALL stop it NOW. Thanks for being polite! snort of laughter -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ 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 manager
What is the best Asterisk manager to use, i do not mind web based or GUI. Thank You Kanishka ___ 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] Creating extension groups
Hi I want to create 2 groups of extensions, for example group 1 cant make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of theextensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please explain how to do this in detailed. I want to know how to route all outgoing calls through our SIP server and how to stop some of the extensions from taking outgoing calls Thank you, Kanishka ___ 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] logger reload/restart hanging
Hi, We're running a very old version of Asterisk (CVS-HEAD-08/03/04) and we're having some problems with logging. Our logger.conf has the following: full = notice,warning,error,debug,verbose After having started Asterisk, asterisk will hang in /usr/sbin/asterisk -rx 'logger reload' unless some output has been sent to the file. I can't find anything on bugs.digium.com related to this problem. Am I the only one? Also no useful output will be sent to the log file, unless I run asterisk -rdn and exit from the console. Is this normal? How do I prevent neeeding this step? I know we shoul move to at least 1.0, but we're running this in production and we haven't felt the need to upgrade. If necessary I can backport... -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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: RE : [Asterisk-Users] Using as FAX 100% IP
I don't believe this will work, but haven't tried myself. TAFM requires spandsp. I'd do some investigation there as to whether spandsp can function with g711 (the last i checked it didn't). Good luck - if you make any progress please post to the list. -Steve Hakem Taourchi wrote: Thank you ver much for this help Steven What I am planning is this: 1-) Receive fax on a DID that is being routed in IP to the asterisk server; 2-) Based on the rule on that incoming fax, Asterisk needs to capture it, store it as pdf file and e-mail it to a predefined destinoatin (based on DID); 3-) If 2 is not possible, then send fax to PSTN destination using voip; Did you use this http://tafm.sourceforge.net/ to make ATA sepura work ? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Steven Kokinos Envoyé : mercredi 23 février 2005 15:46 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Using as FAX 100% IP I'd say that would depend on the configuration you are considering. We have a number of fax machines running off of sipura spa-2000's that connect to a remote asterisk server and terminate to the pstn via voip as well. I'd say it's about 90% reliable at this point. However, we've noticed quite a bit of variability around the quality of the connection and underlying provider you are using for termination - so your results will vary. It also seems a couple of other tricks help: (1)put an ADSL filter at the fax machine end, this seems to help filter out noise from the signal and slow down the analog modem (not sure if this is an old maid's tale or not, but seems to help). (2)if you are using the sipura as your ATA, disable all of the fax detection. Just make sure you are only allowing ULAW as your codec. Other than that it's pretty straightforward. If you are looking to do something to send receive to files, etc. we haven't had any luck getting something along those lines to work w/pure voip. -Steve Hakem Taourchi wrote: Hello, Before putting any effort, I would like to know if somebody has successfully run asterisk receiving FAXs in IP and sending them out in IP as well? If yes using which components please? Any help is greatly appreciated ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] List tips for new subscribers
-Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: 23 February 2005 15:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] List tips for new subscribers This list is for discussions among users of Asterisk, not a getting started hotline for beginners. Beginners learn by reading documentation and examining the sample files included. Peter I have been holding back from perpetuating this thread but it beats doing real work so here goes my two cents. Digium need to take some of the flak on this imho. I say this because I feel their marketing should make it abundantly clear that Asterisk is not a simple product to configure admin. (side note: If you havent bought their hardware and are using Asterisk for free them again you should expect even less assistance imo) That is to say I don't see how people without either programming experience or some good scripting knowledge would really be able to get the most out of Asterisk (certainly in a business environment). If its for a business then you should pay someone to help. If your playing with it at home then that's great but just don't expect quick replies for beginner advice on this list. My take is by all means ask just don't expect pleasant replies if your installation / config questions are answered on the WIKI or if you are requesting dumbing down of the info given. That said some people do reply to questions in an overly aggressive and unconstructive manner and this helps nobody. Best course of action is ignore the question and move on with whatever keeps you off the streets during the day ;-) Good luck Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in a writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ 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] List tips for new subscribers
On Wed, 2005-02-23 at 15:25 +0200, [EMAIL PROTECTED] wrote: To all new subscribers: This is not a friendly list, and the members are mostly sick of giving away knowledge they reckon someone should pay for. Oh fun, we hit slashdot once again. Here comes a new wave of annoying, lazy, and 3-10 months from being useful newbs. While I don't mind the idea of projects I like getting some press, slashdot newbies are almost always looking for a quick payoff for any thing they get involved in. As soon as you get posted to slashdot the useful messages on a list drop significantly until either the users lose interest and drop off or they spend enough time to learn how to do their own simple research. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Creating extension groups
in extensions.conf, create a context for your internal extensions. In the context for outgoing calls, add include = internalextensions. Then in zapata.conf, for each extension put context=internalextensions for people with no outgoing access, and put the others in the context of the outgoing calls. On Wed, 2005-02-23 at 09:27, Kanishka Somaratne wrote: Hi I want to create 2 groups of extensions, for example group 1 cant make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of theextensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please explain how to do this in detailed. I want to know how to route all outgoing calls through our SIP server and how to stop some of the extensions from taking outgoing calls Thank you, Kanishka ___ 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 Dennis Webb Delta Express #3 Afco Road West Memphis, AR 72301 (901)508-9711 ___ 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] Digium TE405P and Cirpack Switch
Hi, I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch (www.cirpack.com). IP Network--*--Cirpack--Public PSTN Network ISDN interco is EuroIsdn, ports are configured ccs, hdb3, crc4. Cirpack is Network, * is Terminal/User. As I encountered some pb with Sip to Zap transcoding (* to Cirpack way poor quality, the other way fine), I tryed to put a loop on * in order to make a 100 % TDM call and qualify only the ISDN side. in zapata.conf, echocancel is on, echocancelwhenbridge is off. A=calling one B=called one both are PSTN terminals 1) Incoming call on my Cirpack from PSTN. 2) Routing to * 3) On *, routing back to Cirpack. 4) On Cirpack, routing to PSTN. = Quality B to A is perfect. Quality A to B is very poor, crackling sound, a strong crackling echo. I am observing the same effect when performing a Sip-to-Zap call. That's a one way problem. If you have any idea, or tweak settings which could help, you're welcome. thieumS * 1.0.5 zaptel 1.0.4 ___ 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] List tips for new subscribers
On Wed, 2005-02-23 at 15:19 +, Adrian Chapman wrote: Peter Svensson wrote: New Asterisk users should try to help themselves first. Perhaps we should create a list asterisk-newbies for thos who do not want to go through the effort of reading up enough to ask well formulated questions. Strikes me as a very good suggestion. Perhaps all subscribers to it should be provided with a welcome note including key URLs - voip-info, a dedicated noob faq, the list archives and Google among 'em. And when they don't get any answers or worse they get incorrect and nonreviewed answers, we will get the cross posting and or just repeats of the question here or on the -dev list. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] 7960 Not Picking up new firmware.
G'Day All. So I got the TFTP server all set up -thanks to much help from this list- the 7960 found it and updated to SIP the first firmware P0S30200. What I am now trying to do is upgrate through all the versions, as recommended, to the latest version, P003-07-3-00. I thought this would be accomplished by simply changing the sole line in the OS79XX.TXT file to P0S30203 and reboot the phone. But no success. Any pointers? Thanks Ferg ___ 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] Mandrake CAPI EPIA!
Razza wrote: Trying the get back to position of a running * PBX, I tried to install the zaptel drivers, using the following process - CD zaptel. Make linux26 Make install When I modprobe zaptel I get the following errors - [EMAIL PROTECTED] zaptel-1.0.4]# modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.8.1-12mdk-i586-up-1GB/misc/zaptel.ko): Invalid module format Razza, Two things, First. Run the remove software option and search for kernel-source. Make sure that it's for the version of the kernel you have installed. If not, remove it and download the correct source. Two. Make sure you've read the README.udev file in zaptel directory. Doug ___ 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] Creating extension groups
On Wed, 2005-02-23 at 15:27 +, Kanishka Somaratne wrote: Hi I want to create 2 groups of extensions, for example group 1 cant make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please explain how to do this in detailed. I want to know how to route all outgoing calls through our SIP server and how to stop some of the extensions from taking outgoing calls Please do not use HTML when sending email as it adds nothing to the display and more than doubles the data sent for the same message on top of including the non HTML message. What you need is to separate the users by using contexts in the extensions conf. This is a regularly asked question. [sip-outbounds] exten = _9..,1,Dial(Your sip provider details) [extensions] Your internal extensions [inbound] The matching for inbound calls. include = extensions [restricted-users] include = extensions [not-restricted-users] include = extensions include = sip-outbounds Hopefully this makes sense. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Music on Hold
Hi, I did recognice an rather strage behaviour on Music on Hold: Situation Caller C does call Person A Person A puts C on hold to ask B MOH is (correctly) activated for C After talking to B A does hangup to transfer C to B In this moment MOH is activated for C for a moment before C is transferred to B The MOD can be seen on the asterisk console and it can be heared as a short buzz on Phone B. Any Idea how to avoid this? Elmar ___ 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] Question about DTMF
I am using an IAX connection via Free World Dial Up. On a incoming call from any of the Michigan based numbers the DTMF for the menus work just fine. But when calling in from any number from http://www.notaduck.com/My_Homepage_Files/Page1.html DTMF does not work. Is this a problem on my end ? ___ 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] List tips for new subscribers --sorry for 2 nd post, missed this.
This list is for discussions among users of Asterisk, not a getting started hotline for beginners. Beginners learn by reading documentation and examining the sample files included. Mmm, I (respectfully) disagree. One of the unstated objectives of mechanisms like this list is to evangelize the platform. Obviously, we all want it to do well. You can only do that by creating mindshare. You create mindshare by winning people over. You win people over in a technical context by helping them wrap their heads around the concept and implement it. Sometimes, this involves hand-holding, as I do with my boss, my boss' boss etc every day. Look, there are two kinds of people (on the list): One that can deal with the technical implementation of Asterisk and have no problem with it, and the other kind, that get fired up about the *concept* but are short of the chops to make it happen. We ignore the second kind at our own peril (I actually should have said you guys instead of we because I am in-between the two types). I can see a scenario where if the platform becomes inaccessible to PHB / noob types because of things like attitude, Asterisk will be relegated to also-ran status with such illustrious company as the Amiga, which still has an incredibly vocal minority that insists that Amiga still r00lz, but nobody listens to them and considers them crackpots who should Just Get Over It. We *know* Asterisk is a category-redefining platform. We *know* it is Insanely Great. But ticking people off with brusque answers and flames will *not* win the hearts and minds of potential adopters. *1-2 *1 As far as the previous black box comment goes (where an implementor doesn't want someone to know about the inner workings, so they can charge $$$), there is a certain truth to that, and, while GPL allows for that, it is completely contrary to the spirit of the platform and makes the Asterisk community no better than (insert your favorite telecom player whipping boy here) *2 Are you so fussy about how your inbox or whatever is displayed that you are willing to alienate a potential adopter because he top posts or uses HTML? Come on you guys, who cares? What if you pissed off the CTO of a Fortune 500 and he ruled out an Asterisk rollout because he took your flames personally? (sounds like a stretch, but I find it plausible) You didn't do too much to help the platform that day, did you? ___ 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] install BRIstuff on *@home?
From: Niksa Baldun [EMAIL PROTECTED] Each release of BRIstuff is made for a specific * version. BRIstuff installer automatically downloads the correct version, patches and installs it. You should just run the install.sh and it will replace your current * installation. Your existing configuration (extensions.conf etc.) will not be changed. Bear in mind that you have to copy the BRI modules (qozap.ko and zaphfc.ko) manually to your /lib/modules/`uname -r`/misc directory. Erwin de Raad wrote: I'm still trying to install a HFC-s BRI card onto [EMAIL PROTECTED] Hi Niksa, Thank you for your response. I will go ahead then with the install. I'm deducting what cannot be wrong when it doesn't work... ;-) Cheers! Erwin. ___ 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] List tips for new subscribers
[EMAIL PROTECTED] wrote: (side note: If you havent bought their hardware and are using Asterisk for free them again you should expect even less assistance imo) Right, so I have to buy hardware I don't need? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] List tips for new subscribers
On February 23, 2005 10:21 am, [EMAIL PROTECTED] wrote: Oh I'm sorry. This is the first list I've joined where this is such a big issue! Forgive me for not having your superior understanding of mail clients, and/or list servers! You have a *servere* inferiority complex. I asked a simple question. The only people who don't see why it's a problem use inferior mail user agents which don't support threading, or perhaps they don't realize that they can do threading. Neither one means the user is stupid or in any way a lessor human. A little uninformed perhaps but that's nothing that isn't curable, unless they develop full-out ignorance or, in your case, an inferiority complex. Aside: I can't fathom *anyone* reading a busy mailing list without threading, no wonder people such as yourself seem so uptight and fidgety when this is brought up. I'm almost positive that Evolution can handle threads so you know what relief and welcome organization a threaded MUA can bring so reading this list in a nonthreaded mailer is your own doing and ultimately a cause of so much of your frustration, I think. But thanks for the reply, thanks for being polite, thanks for kicking my head in for not doing it exactly like you would want it done . My exact text: Since you are so vocal, I do have to ask, is hitting the reply button and then erasing everything faster than simply clicking on the Asterisk-users@lists.digium.com email address to start a new, blank message to the list? I've never gotten a clear answer. It seems like a hell of a lot of work to reply to a message, erase everything and start anew. Grow a thicker skin. That wasn't in any way shape or form abusive or even the slightest bit condescending. I asked a question and offered my opinion on what I figured it was you were doing. If you can't handle it, you don't belong here. Plain and simple. Get a punching bag, watch some WWF - Whatever. Just do SOMETHING to relax. PLEASE! I'm not the anxious one. Read over what I wrote, and then read over what you wrote. Who's the one who's agitated? -A. ___ 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] Newbie Help - Auto Fallthrough
I am a serious Asterisk newbie: just installed asterisk last week and it is
now running with our Voicetronix OpenLine4 hardware.
All is working as expected with one exception, in the following sequence
(extracted from my extensions.conf file):
[GetConfirmation]
exten = s,n,SetVar(TimeOut=0) ; if timeout and TimeOut=1 then user
already timed out once, so hangup
exten = s,n,SetVar(State=GetConfirmation) ; set up for
time-out return
exten = s,n,ResponseTimeout(10); Set
Response Timeout to 10 seconds
exten = s,n,Background(mymenu}) ; play menu
msg (press 1..., press 2... press 3...), wait for response
include = TimeOut ; include
timeout handler
exten = 1,1,Goto(DoTransaction,s,1); submit
transaction
exten = 2,1,Goto(GetFare,s,1) ; send user
back to re-enter fare, tip and cab #
exten = 3,1,Goto(s,1) ; replay
confirmation msg
[TimeOut]
; user timed out, so see if TimeOut flag set. If set, then this is the
second time in a row that user has timed out
; so hang up. If not set, then set TimeOut and let user try again
; Usage: include = TimeOut (placed as last entry in any context
(routine) where timeout handling required)
exten = t,1,NoOp(In TimeOut: ${TimeOut})
exten = t,2,Gotoif($[${TimeOut}]?:5:3)
exten = t,3,SetVar(TimeOut=1) ; users first time-out so
set flag
exten = t,4,Goto(${State},s,1)); start current sub-routine
all over again
exten = t,5,Hangup ; users second time-out, so
hang up
In the above sequence (context) the menu message plays as expected, and as
long as the user enters a DTMF digit BEFORE the message playback completes,
all words as it should. However, if the message playback completes, there
is no 10 second wait for the user's entry, instead the call hangs up
immediately and the following debug info is displayed on the console:
-- Executing SetVar(vpb/1-1, State=GetConfirmation) in new stack
-- Executing ResponseTimeout(vpb/1-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing BackGround(vpb/1-1, 1-1) in new stack
-- Playing '1-1' (language 'taxi')
== Auto fallthrough, channel 'vpb/1-1' status is 'UNKNOWN'
== vpb/1-1: Hangup requested
== vpb/1-1: Ending record mode (1/yes)
vpb/1-1: stopped record thread on vpb/1-1
== vpb/1-1: Ending play mode on vpb/1-1
vpb/1-1: Setting state down
== vpb/1-1: Hangup complete
Restarting monitor
Trying to reawake monitor
Monitor restarted
Monitor got null event
vpb/1-4: Event [12=[03] Loop Drop]
vpb/1-4: handle_notowned: mode=3, event[12][[03] Loop Drop
]=[0]
vpb/1-4: handle_notowned: mode=3, [12=0]
Can anyone tell me:
(1) why there is no 10 second wait time?
(2) why the TimeOut code did not execute?
Any and all ideas, comments, suggestions appreciated!
Jonathan
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RE: [Asterisk-Users] List tips for new subscribers --sorry for 2nd post, missed this.
Colin wrote: A lot of good sensible stuff. Well done Colin. Bill Seddon Lyquidity Solutions Limited -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: February 23, 2005 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] List tips for new subscribers --sorry for 2nd post, missed this. This list is for discussions among users of Asterisk, not a getting started hotline for beginners. Beginners learn by reading documentation and examining the sample files included. Mmm, I (respectfully) disagree. One of the unstated objectives of mechanisms like this list is to evangelize the platform. Obviously, we all want it to do well. You can only do that by creating mindshare. You create mindshare by winning people over. You win people over in a technical context by helping them wrap their heads around the concept and implement it. Sometimes, this involves hand-holding, as I do with my boss, my boss' boss etc every day. Look, there are two kinds of people (on the list): One that can deal with the technical implementation of Asterisk and have no problem with it, and the other kind, that get fired up about the *concept* but are short of the chops to make it happen. We ignore the second kind at our own peril (I actually should have said you guys instead of we because I am in-between the two types). I can see a scenario where if the platform becomes inaccessible to PHB / noob types because of things like attitude, Asterisk will be relegated to also-ran status with such illustrious company as the Amiga, which still has an incredibly vocal minority that insists that Amiga still r00lz, but nobody listens to them and considers them crackpots who should Just Get Over It. We *know* Asterisk is a category-redefining platform. We *know* it is Insanely Great. But ticking people off with brusque answers and flames will *not* win the hearts and minds of potential adopters. *1-2 *1 As far as the previous black box comment goes (where an implementor doesn't want someone to know about the inner workings, so they can charge $$$), there is a certain truth to that, and, while GPL allows for that, it is completely contrary to the spirit of the platform and makes the Asterisk community no better than (insert your favorite telecom player whipping boy here) *2 Are you so fussy about how your inbox or whatever is displayed that you are willing to alienate a potential adopter because he top posts or uses HTML? Come on you guys, who cares? What if you pissed off the CTO of a Fortune 500 and he ruled out an Asterisk rollout because he took your flames personally? (sounds like a stretch, but I find it plausible) You didn't do too much to help the platform that day, did you? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Send outgoing calls to a SIP gateway
How do I route all the outgoing calls througha SIP gateway, this should send more than one outgoing call to the sip gateway at once. please show me the sample configurations on how to do this. my SIP gatway can accecpt direct IP traffic or SIP proxy traffc. Thank You Kanishka ___ 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] Problem connecting a TE410P to an E1/IP equipment
Hi, I'm trying to connect a PC with a TE410P to an E1/IP equipment. Unfortunately I keep getting a yellow alarm from zaptel (in zttool) and a Loss of Framing alarm on the remote equipment. The E1/IP is connected on the other side to a PRI interface on a GSM MSC. I have configured the span as: span=1,1,0,ccs,hdb3 (also tried span=1,0,0,ccs,hdb3) and the channels as: bchan=1-15,17-31 dchan=16 The LED on the port is green. I tried to load both the se and us tonezones (I'm in Sweden). Is there a way I can know why the yellow alarm was triggered? And do you have ideas why the other equipment reports a LOF alarm? Here's the content of /proc/zaptel/1: cat /proc/zaptel/1 Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 HDB3/CCS YELLOW ClockSource IRQ misses: 4 1 TE4/0/1/1 Clear (In use) 2 TE4/0/1/2 Clear (In use) 3 TE4/0/1/3 Clear (In use) 4 TE4/0/1/4 Clear (In use) (...) 10 TE4/0/1/10 Clear (In use) 11 TE4/0/1/11 Clear (In use) 12 TE4/0/1/12 Clear (In use) 13 TE4/0/1/13 Clear (In use) 14 TE4/0/1/14 Clear (In use) 15 TE4/0/1/15 Clear (In use) 16 TE4/0/1/16 HDLCFCS (In use) 17 TE4/0/1/17 Clear (In use) 18 TE4/0/1/18 Clear (In use) (...) 27 TE4/0/1/27 Clear (In use) 28 TE4/0/1/28 Clear (In use) 29 TE4/0/1/29 Thanks, Johan ___ 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] 7960 Not Picking up new firmware.
You have to change the image name in the OS79XX.txt and SIPDefault.cnf files to match the name of BIN file you are trying to load ... With versions of the firmware prior to 7.x, the name you put in the OS79XX.txt file and the SIPDefault.cnf files are the same; simply the BIN file name less the BIN extension ... As you get to version 7.x and up, the file name you put in OS79XX.txt is actually the name of a Universal Loader ... The name of the SIP binary image is entered in SIPDefault.cnf ... I got a help on this one from a pretty decent article on the WIKI at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx ... Look at the section header Software Upgrade Requirements ... This gave me the clues I needed to get the 7.3 Sip image to load properly ... G.Hendershot -Original Message- From: Ferguson, Michael [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 23, 2005 10:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 7960 Not Picking up new firmware. G'Day All. So I got the TFTP server all set up -thanks to much help from this list- the 7960 found it and updated to SIP the first firmware P0S30200. What I am now trying to do is upgrate through all the versions, as recommended, to the latest version, P003-07-3-00. I thought this would be accomplished by simply changing the sole line in the OS79XX.TXT file to P0S30203 and reboot the phone. But no success. Any pointers? Thanks Ferg ___ 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] Problem connecting a TE410P to an E1/IP equipment
On Wed, 23 Feb 2005, Johan Bilien wrote: I'm trying to connect a PC with a TE410P to an E1/IP equipment. Unfortunately I keep getting a yellow alarm from zaptel (in zttool) and a Loss of Framing alarm on the remote equipment. The E1/IP is connected on the other side to a PRI interface on a GSM MSC. I have configured the span as: span=1,1,0,ccs,hdb3 (also tried span=1,0,0,ccs,hdb3) and the channels as: bchan=1-15,17-31 dchan=16 The LED on the port is green. I tried to load both the se and us tonezones (I'm in Sweden). Is there a way I can know why the yellow alarm was triggered? And do you have ideas why the other equipment reports a LOF alarm? Yellow alarm is the same as remote alarm - i.e. the other side is saying that it cannot hear you. Given the Loss Of Framing on the other end this seems resonable. Is the remote end set to use CRC4? Try adding ,crc4 to the end of the span line. Peter ___ 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] avaya 4602
After downloading the firmware from a tftp server for avaya 4602 ip phone, my phone console keeps getting message saying contacting http server. according to this link: http://voip-info.org/wiki-Avaya+4602+configuration My phone hangs at Contacting HTTP Server while performing a firmware update Either your phone cannot download the sip_4602D01A.txt file (in which case it will request an appsip.ebin application file) or it cannot download the correct boot and application firmware files. My question is how do I reset the phone to the factory default, I have the user's guide about resetting, tried, also tried use mute button and 73736#. still not working. there is no way i could have the phone to look for the tftp server again either dhcp or static. __ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail ___ 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
