Re: [asterisk-users] German date format in voicemail emails
27 jan 2009 kl. 01.14 skrev Tilghman Lesher: On Monday 26 January 2009 08:21:10 am Danny Nicholas wrote: Did you read the source for app_voicemail? Line 239 says you have to set locale in the config and have the sound file einE. Of course an easier way would be to locate the 19 day and month files and just replace them with German equivalents (assuming that 26 and 2009 sound the same in a German pronunciation). You might want to read his message before you start recommending things. The OP was interested in emails, not message playback. The language setting only affects prompt playback, not email messages. Minivoicemail actually has - multiple e-mail formats - locale support so you get the date in local language and format. This has not been backported to the old voicemail application. If you only need voicemail to e-mail, then use minivoicemail instead. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't start Asterisk after installing Digium G729 licence
2009/1/27 Olivier oza-4...@myamail.com Hi, I carefully followed instructions in README file lasting with : /root/register ... blabla asterisk -r CLI restart now Then asterisk -r fails with : # asterisk -r Asterisk 1.6.1-beta4, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Running as group 'asterisk' == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf == Found Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) # tail /var/log/asterisk/messages [Jan 27 08:17:40] WARNING[23451] chan_dahdi.c: Ignoring hasiax at line 39. [Jan 27 08:17:40] WARNING[23451] chan_dahdi.c: Ignoring hasmanager at line 47. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: G.729 transcoding module version 34, Copyright (C) 1999-2007 Digium, Inc. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: This module is supplied under a commercial license granted by Digium, Inc. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: Please see the full license text supplied by the accompanying [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: register utility, or ask for a copy from Digium. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: This product includes software developed by the OpenSSL Project [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: Copyright (C) 1998-2006 The OpenSSL Project Before opening a ticket to Digium, is there something obvious I missed ? Google didn't show much hint ... After reading this see register utility line, I ran /root/register once again but it didn't change anything ... Cheers Including noload = codec_g279a.so in /etc/asterisk/modules.conf at least allow me to restart asterisk successfully. Changing ownership of /var/lib/asterisk/licences files to asterisk:asterisk didn't change the fact it didn't start. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 54, Issue 83
Hi, How can view the list of RFCs and other drafts that asterisk supports? Basically I would like to know does asterisk supports RFC 3891? Thanks, Niraj Roy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] server sizing for ~ 200 simultaneous call
Hi to all i'm planning the migration of a company on Asterisk, i have planned this scenario: 2 server with * 4 GB RAM * 2 CPU 64 bit dual core * RAID 1 * 2 network interfaces 1000 Mbit/s Each server will have a virtual interface that will be switched from one to the other in case of hardware problem. The question is: can one server with those settings manage up to 200 simultaneous call? The server will receive SIP calls and forward them through a CISCO router. Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] server sizing for ~ 200 simultaneous call
Am Dienstag, den 27.01.2009, 11:30 +0100 schrieb nik600: Hi to all i'm planning the migration of a company on Asterisk, i have planned this scenario: 2 server with * 4 GB RAM * 2 CPU 64 bit dual core * RAID 1 * 2 network interfaces 1000 Mbit/s Each server will have a virtual interface that will be switched from one to the other in case of hardware problem. The question is: can one server with those settings manage up to 200 simultaneous call? The server will receive SIP calls and forward them through a CISCO router. Thanks to all It depends what the dialplan does with these 200 calls. If the asterisk transcodes, has conference rooms and ivr features as well as call recording, than it might not be sufficient. For simpler cases it should. Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue time to answer/abandon
Hi all, Is there a way to get the time that a specific queued call took to be answered or abandoned? Thanks, Gabriel Ortiz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] server sizing for ~ 200 simultaneous call
If you use reinvites (the peers are all internal, right?), definitely. Without reinvites - maybe. On Jan 27, 2009, at 5:30 AM, nik600 nik...@gmail.com wrote: Hi to all i'm planning the migration of a company on Asterisk, i have planned this scenario: 2 server with * 4 GB RAM * 2 CPU 64 bit dual core * RAID 1 * 2 network interfaces 1000 Mbit/s Each server will have a virtual interface that will be switched from one to the other in case of hardware problem. The question is: can one server with those settings manage up to 200 simultaneous call? The server will receive SIP calls and forward them through a CISCO router. Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk freezes with Fixup failed on channel SIP/...MASQ
Grygoriy, [...] A practice that was once described in the code comments as being nasty. thanks for your input. My knowledge of 'hard core' programming is limited, so I cannot judge on what is written on freeswitch.org. Though it sounds logical to me. But as I said, this is on a production system and we have no way of changing it. So the question is: What can we do to avoid the situation. One will often see 3 or 4 channels up for a single call during a call transfer because of this. Am I right that this seems to be the same when calling a group of (SIP) phones? I'm calling Dial(Local/1Local/2Local/3Local/4) now. Each Local channel does some (MySQL) database to decide which phone to call actually. Well, it seemed to work 99% of the time, but these two calls crashed. Is the Dial(Local/) command something one should avoid? Udo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk freezes with Fixup failed on channel SIP/...MASQ
2009/1/27 Udo Schacht-Wiegand aster...@wiegand.name: Grygoriy, [...] A practice that was once described in the code comments as being nasty. thanks for your input. My knowledge of 'hard core' programming is limited, so I cannot judge on what is written on freeswitch.org. Though it sounds logical to me. But as I said, this is on a production system and we have no way of changing it. So the question is: What can we do to avoid the situation. [snip] Just a thought - I have not put any time into it myself. Have you checked the changelog between 1.4.17 and 1.4.23.1 to see if there is an obvious fixed SIP masq channel hangs or similar? 1.4.17 is fairly old by now. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G726 Codec
Dear Sir, I would like to ask please about how I can force asterisk to send all G726 codecs without translation... g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- -- -- gsm- -222 21 3- -- 2- ulaw- 3-12 21 3- -- 2- alaw- 31-2 21 3- -- 2- g726aal2- 322- 21 3- -- 1- adpcm- 3222 -1 3- -- 2- slin- 2111 1- 2- -- 1- lpc10- 3222 21 -- -- 2- g729- ---- -- -- -- -- speex- ---- -- -- -- -- ilbc- ---- -- -- -- -- g726- 3221 21 3- -- -- g722- ---- -- -- -- -- Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need help
Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' if you have made no changes to your asterisk configuration since it last worked Contact your service provider as they have an issue more likely though is you have a bad dial command. /telepathy -- Kind Regards Max Brooks - Developer Legatio Technologies Limited Phone: 01793 520 506 www.legatio.com, www.ftax.co.uk Legatio is part of the Callcredit Information Group: www.skipton.com, www.callcredit.co.uk, www.eurodirect.co.uk Legatio Technologies Limited, One Park Lane, Leeds, West Yorkshire, LS3 1EP Registered in England and Wales No. 4519902 - Powered by Legatio.com This message is confidential. It may not be disclosed to, or used by, anyone other than the addressee(s). If you receive this message in error, please advise us immediately using the email address i...@legatio.com. Internet e-mail is not necessarily secure. Legatio will not accept responsibility for alterations or additions to any e-mail message or attached documents that occur after transmission. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue time to answer/abandon
SURE! you can use the manager event handler and a function dump_event! you can dump all kind off events to a text file or a database! I did it successfully! GOOD LUCK:) 2009/1/27 Gabriel Ortiz Lour ortiz.ad...@gmail.com Hi all, Is there a way to get the time that a specific queued call took to be answered or abandoned? Thanks, Gabriel Ortiz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reading/Writing the Astdb
Thanks for the feedback, but unfortunately, there is still no joy. If I do this: echo -n database put FOO BAR 1 | socat STDIO UNIX-CONNECT:/var/run/asterisk/asterisk.ctl I get the following output on the command line: pbx-75/2395/1.4.20.1 pbx-75 is the hostname, 2395 is the PID of the asterisk process, and 1.4.20.1 is the version of asterisk I am running. I also see this in the asterisk CLI: -- Remote UNIX connection -- Remote UNIX connection disconnected But the database value did not update. Thanks FSD Date: Mon, 26 Jan 2009 18:31:30 -0800 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Reading/Writing the Astdb On Mon, 26 Jan 2009, cbbs...@hotmail.com wrote: That's a really good idea, however, I am having problems getting it to work. I tried the following: echo -n asterisk -rx \database put FOO BAR 1\ | socat - /var/run/asterisk/asterisk.ctl and echo -n asterisk -rx \database put FOO BAR 1\ | socat - UNIX-CONNECT: /var/run/asterisk/asterisk.ctl echo -n show channels\ | sudo socat STDIO UNIX-CONNECT:/var/run/asterisk.ctl and echo -n show channels\ | sudo socat STDIO /var/run/asterisk.ctl work for me. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live™: E-mail. Chat. Share. Get more ways to connect. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t2_allup_explore_012009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Webcall app needed
Hello all, I need to configure an application which let me to call from a web page. Someone has experience using apps to make webcalls? Which software do you use? Thanks. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Webcall app needed
This conversation has been done to deathare archive search not available.? But the answers are www.mexuar.com www.phonefromhere.com and there are also a free open source versions but they take work on your part to setup. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of voip crazy Sent: Tuesday, 27 January 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Webcall app needed Hello all, I need to configure an application which let me to call from a web page. Someone has experience using apps to make webcalls? Which software do you use? Thanks. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need help
The problem is I have 20 agents calling all but 3 of them get this error when calling On Tue, Jan 27, 2009 at 6:49 AM, Max Brooks m...@legatio.com wrote: Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' if you have made no changes to your asterisk configuration since it last worked Contact your service provider as they have an issue more likely though is you have a bad dial command. /telepathy -- Kind Regards Max Brooks - Developer Legatio Technologies Limited Phone: 01793 520 506 www.legatio.com, www.ftax.co.uk Legatio is part of the Callcredit Information Group: www.skipton.com, www.callcredit.co.uk, www.eurodirect.co.uk Legatio Technologies Limited, One Park Lane, Leeds, West Yorkshire, LS3 1EP Registered in England and Wales No. 4519902 - Powered by Legatio.com This message is confidential. It may not be disclosed to, or used by, anyone other than the addressee(s). If you receive this message in error, please advise us immediately using the email address i...@legatio.com. Internet e-mail is not necessarily secure. Legatio will not accept responsibility for alterations or additions to any e-mail message or attached documents that occur after transmission. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Nortel integration via SIP protocol
Hi, I need to integrate my Asterisk with a Nortel Meridian 11, but I can´t use PRI, Analog lines, etc. It has to be via SIP protocol, and there is few information about this type of integration. Could someone please help me?? Thanks, Pablo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need help
What is your call-limit set to in sip.conf? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bayardo Sanchez Sent: Tuesday, January 27, 2009 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] I need help The problem is I have 20 agents calling all but 3 of them get this error when calling On Tue, Jan 27, 2009 at 6:49 AM, Max Brooks m...@legatio.com wrote: Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' if you have made no changes to your asterisk configuration since it last worked Contact your service provider as they have an issue more likely though is you have a bad dial command. /telepathy -- Kind Regards Max Brooks - Developer Legatio Technologies Limited Phone: 01793 520 506 www.legatio.com, www.ftax.co.uk Legatio is part of the Callcredit Information Group: www.skipton.com, www.callcredit.co.uk, www.eurodirect.co.uk Legatio Technologies Limited, One Park Lane, Leeds, West Yorkshire, LS3 1EP Registered in England and Wales No. 4519902 - Powered by Legatio.com This message is confidential. It may not be disclosed to, or used by, anyone other than the addressee(s). If you receive this message in error, please advise us immediately using the email address i...@legatio.com. Internet e-mail is not necessarily secure. Legatio will not accept responsibility for alterations or additions to any e-mail message or attached documents that occur after transmission. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA-3102 in India - Problem dialing out PTSN
Good morning,
I've been having some problems getting the SPA-3102 working properly in
India. Specific problem is that calls from the Asterisk server out the FXS
port is failing. When trying to make calls, I'm getting this message:
[Jan 26 23:00:31] NOTICE[2136]: chan_sip.c:13774 handle_request_invite: Call
from '' to extension '66200' rejected because extension not found.
I believe everythign is set up correctly because when I was doing tests here
in the US everything worked perfectly before sending the unit to India. Now
that its there, I'm having some troubles. If there's more information
that's required I'd be more than happy to provide. Thank you!
Extensions.conf
--
ignorepat = 66
exten = _66XX.,1,Dial(${PHONE7}/${EXTEN:2},30,r)
exten = _66XX.,2,Congestion
sip.conf
--
[215] ; Linksys India Internal
type=peer
username=215
secret= [SOMETHING]
host=dynamic
context=superuser
port=5060
dtmfmode=rfc2833
callerid=India Internal 215
disallow=all
allow=ulaw
canreinvite=no
qualify=yes
nat=yes
[220] ; Linksys India PSTN Line
type=peer
username=220
secret=[SOMETHING]
host=dynamic
context=superuser
port=5060
dtmfmode=rfc2833
disallow=all
allow=ulaw
canreinvite=no
qualify=yes
nat=yes
Logs
---
[Jan 26 23:00:05] NOTICE[2136]: chan_sip.c:13774 handle_request_invite: Call
from '' to extension '668002525' rejected because extension not found.
[Jan 26 23:00:31] NOTICE[2136]: chan_sip.c:13774 handle_request_invite: Call
from '' to extension '66200' rejected because extension not found.
localhost*CLI
___
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[asterisk-users] T.38
Dear All, I'm trying to send Fax using T.38 protocol but the FAX is not going through..I'm getting the following error om /var/log/messages [Jan 27 16:46:31] WARNING[25435] channel.c: No path to translate from SIP/80.169.210.181-0896bfd0(4) to SIP/028949469-b7703d40(256) [Jan 27 16:46:40] WARNING[24349] channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw) [Jan 27 16:46:40] WARNING[24349] channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw) Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RFC -- Improving the quality of the mailing lists
The -user and -dev mailing lists are a valuable resource -- when they are not cluttered by posts unrelated to the charter of the lists. In my limited memory, this last weekend represents a new low in the relevant subject to noise ratio. Replying to requests with meaningless, misleading, or misspelled subject lines (I need help, asterisk help, Ntework Card) encourage careless posting and obfuscate useful replies from search engines. Also, while replying to such requests may seem helpful, some of the requests indicate such a lack of basic understanding that giving the answer is like giving a small child a very sharp knife when they ask for a slice of bread. For example: How do I delete these files that end in that squiggly thing in my current directory and all directories below? Since most of these users are probably running as root, a simple extra space here and a missed character there (rm --force --recursive /* ~ vs rm --force --recursive ./*~ can have catastrophic consequences. In an attempt to improve the quality of the lists, I propose the following: For a user's first 10 posts, they will receive a reply with a link to a web page and have to answer the following questions: 0) I acknowledge that I am asking for free help and I acknowledge that following the conventions below increase my chances of engaging another list member with relevant expertise and resolving my request. 1) I am posting a new request. a) My request cannot be answered on a more general list such as Beginning Unix, or on a distribution specific list. b) My request cannot be answered on a more specific list such as an AsteriskNow or Trixbox list. c) I have attempted to search for an answer using a search engine such as Google. d) I know what thread hijacking is and I created this request from scratch. e) I have created a meaningful subject line that indicates with as much specificity as reasonable which part of Asterisk I need help with and why. f) I am not posting a self-serving message directing someone to my product that would be better posted to the -biz list. g) I am not posting in HTML. h) I am posting in English. i) I am fluent in English or I have attempted to have someone who is review my request. j) I have run my request through my spell checking resources. or 2) I am posting a reply to a post. a) I know what top posting is and I am not ignoring the convention of the list. b) I am not posting a self-serving message directing someone to my product that would be better posted to the -biz list or only to the requester. c) I am not posting in HTML. d) I am posting in English. e) I am fluent in English or I have attempted to have someone who is review my post. f) I have trimmed the previous post down to just the point(s) I am replying to. g) I have run my request through my spell checking resources. For -dev, the following questions would be added: ) My post directly relates to changes in the Asterisk C source code. ) I am not reporting a bug or a posting a patch that should be directed to bugs.digium.com. Included in the web page would be the original message with the ability to change the list the message is to be posted to, the subject line, and the body of the message. Comments? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need help
24 chanels On Tue, Jan 27, 2009 at 9:23 AM, Danny Nicholas da...@debsinc.com wrote: What is your call-limit set to in sip.conf? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Tuesday, January 27, 2009 9:12 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] I need help The problem is I have 20 agents calling all but 3 of them get this error when calling On Tue, Jan 27, 2009 at 6:49 AM, Max Brooks m...@legatio.com wrote: Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' if you have made no changes to your asterisk configuration since it last worked Contact your service provider as they have an issue more likely though is you have a bad dial command. /telepathy -- Kind Regards Max Brooks - Developer Legatio Technologies Limited Phone: 01793 520 506 www.legatio.com, www.ftax.co.uk Legatio is part of the Callcredit Information Group: www.skipton.com, www.callcredit.co.uk, www.eurodirect.co.uk Legatio Technologies Limited, One Park Lane, Leeds, West Yorkshire, LS3 1EP Registered in England and Wales No. 4519902 - Powered by Legatio.com This message is confidential. It may not be disclosed to, or used by, anyone other than the addressee(s). If you receive this message in error, please advise us immediately using the email address i...@legatio.com. Internet e-mail is not necessarily secure. Legatio will not accept responsibility for alterations or additions to any e-mail message or attached documents that occur after transmission. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need help
Is it the same 3 or the first 3? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bayardo Sanchez Sent: Tuesday, January 27, 2009 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] I need help 24 chanels On Tue, Jan 27, 2009 at 9:23 AM, Danny Nicholas da...@debsinc.com wrote: What is your call-limit set to in sip.conf? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bayardo Sanchez Sent: Tuesday, January 27, 2009 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] I need help The problem is I have 20 agents calling all but 3 of them get this error when calling On Tue, Jan 27, 2009 at 6:49 AM, Max Brooks m...@legatio.com wrote: Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' if you have made no changes to your asterisk configuration since it last worked Contact your service provider as they have an issue more likely though is you have a bad dial command. /telepathy -- Kind Regards Max Brooks - Developer Legatio Technologies Limited Phone: 01793 520 506 www.legatio.com, www.ftax.co.uk Legatio is part of the Callcredit Information Group: www.skipton.com, www.callcredit.co.uk, www.eurodirect.co.uk Legatio Technologies Limited, One Park Lane, Leeds, West Yorkshire, LS3 1EP Registered in England and Wales No. 4519902 - Powered by Legatio.com This message is confidential. It may not be disclosed to, or used by, anyone other than the addressee(s). If you receive this message in error, please advise us immediately using the email address i...@legatio.com. Internet e-mail is not necessarily secure. Legatio will not accept responsibility for alterations or additions to any e-mail message or attached documents that occur after transmission. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue time to answer/abandon + OrderlyStats Server Edition.
Hi Gabriel, Yes this information is shown in real-time and also in historical reports with the OrderlyStats system. OrderlyStats is now available as a Server Edition you can download and install yourself, as well as the FREE managed service. You can get it at http://www.orderlyq.com/statistics.html Hope this helps, Matt. Gabriel Ortiz wrote: Hi all, Is there a way to get the time that a specific queued call took to be answered or abandoned? Thanks, Gabriel Ortiz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] German date format in voicemail emails
On Tuesday 27 January 2009 02:03:37 Johansson Olle E wrote: 27 jan 2009 kl. 01.14 skrev Tilghman Lesher: On Monday 26 January 2009 08:21:10 am Danny Nicholas wrote: Did you read the source for app_voicemail? Line 239 says you have to set locale in the config and have the sound file einE. Of course an easier way would be to locate the 19 day and month files and just replace them with German equivalents (assuming that 26 and 2009 sound the same in a German pronunciation). You might want to read his message before you start recommending things. The OP was interested in emails, not message playback. The language setting only affects prompt playback, not email messages. Minivoicemail actually has - multiple e-mail formats - locale support so you get the date in local language and format. Unfortunately, it's using setlocale(3), which is not thread-safe. Note that there is a new thread-safe locale interface in POSIX 2008. See the library calls newlocale(3), uselocale(3), and freelocale(3). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 dahdi only?
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non existent. You go to the main Asterisk page (digium.org) and really just old install instructions for 1.2 are in the examples. Download links only give you asterisk itself and not dahdi or libpri which also are needed to run asterisk? It's very confusing to anyone who is new. Someone take notice! we need a link to instructions right of the main asterisk page. My 1st question is am I missing a good step-by step for 1.6 and how to compile/install it along with it's side components (dahdi/libpri)? when/if those side components are actually needed? When would you run asterisk without them entirely? 2nd question is for an IP/SIP only system do I only need DAHDI or do I need DAHDI and LIBPRI? Is libpri only needed if interfacing to a pri? Is 1.6 so cutting edge that I should not expect to find complete documentation (yet)like I seem to be expecting very easily? Thanks much! Steve -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38
check the codecs in sip.conf 2009/1/27 michel freiha mich...@gmail.com Dear All, I'm trying to send Fax using T.38 protocol but the FAX is not going through..I'm getting the following error om /var/log/messages [Jan 27 16:46:31] WARNING[25435] channel.c: No path to translate from SIP/80.169.210.181-0896bfd0(4) to SIP/028949469-b7703d40(256) [Jan 27 16:46:40] WARNING[24349] channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw) [Jan 27 16:46:40] WARNING[24349] channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw) Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need help
only 3 On Tue, Jan 27, 2009 at 10:03 AM, Danny Nicholas da...@debsinc.com wrote: Is it the same 3 or the first 3? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Tuesday, January 27, 2009 9:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] I need help 24 chanels On Tue, Jan 27, 2009 at 9:23 AM, Danny Nicholas da...@debsinc.com wrote: What is your call-limit set to in sip.conf? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Tuesday, January 27, 2009 9:12 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] I need help The problem is I have 20 agents calling all but 3 of them get this error when calling On Tue, Jan 27, 2009 at 6:49 AM, Max Brooks m...@legatio.com wrote: Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' if you have made no changes to your asterisk configuration since it last worked Contact your service provider as they have an issue more likely though is you have a bad dial command. /telepathy -- Kind Regards Max Brooks - Developer Legatio Technologies Limited Phone: 01793 520 506 www.legatio.com, www.ftax.co.uk Legatio is part of the Callcredit Information Group: www.skipton.com, www.callcredit.co.uk, www.eurodirect.co.uk Legatio Technologies Limited, One Park Lane, Leeds, West Yorkshire, LS3 1EP Registered in England and Wales No. 4519902 - Powered by Legatio.com This message is confidential. It may not be disclosed to, or used by, anyone other than the addressee(s). If you receive this message in error, please advise us immediately using the email address i...@legatio.com. Internet e-mail is not necessarily secure. Legatio will not accept responsibility for alterations or additions to any e-mail message or attached documents that occur after transmission. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer -
Re: [asterisk-users] Webcall app needed
search for the correlata thread there is a java applet work over iax so you will not have any problems whit the routers/firewalls nats and that stuff. at the end of the thread there is an example made by wolfgang. David 2009/1/27 Dean Collins d...@cognation.net This conversation has been done to deathare archive search not available.? But the answers are www.mexuar.com www.phonefromhere.com and there are also a free open source versions but they take work on your part to setup. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of voip crazy Sent: Tuesday, 27 January 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Webcall app needed Hello all, I need to configure an application which let me to call from a web page. Someone has experience using apps to make webcalls? Which software do you use? Thanks. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
The higher you raise the barrier for entry to the mailing list, the more you decrease the amount good the mailing list is actually capable of doing. (barrier height is inversely related to how much help we can provide to the people that need help the most) I agree with you regarding the subject spelling/misspelling as it pertains to indexing on the search engines, etc. But if you require those posting to jump through your *10* hoops for the first *10* times they post something (yes, that's 100 hoops. I'm tired of jumping already), you are artificially limiting the number of users that this list can actually help. I don't like getting broken English replies and questions that don't make any sense any more than the next person, but I also get a good chuckle out of reading them. And reading replies that tell people to 'rm -rf /*' gives me a good laugh, too. The only way to REALLY learn is to make mistakes, even if you're making those mistakes because you took the 'advice' that someone gave you for free on the mailing list... Give me a break :) Mailing lists are supposed to be fun and get off topic sometimes. That's what makes them interesting. --Dave PS: Can anyone help me with my broken *.? the ntework card is blinking red and the sips are dropping with echoes. Tai? LOL. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Tuesday, January 27, 2009 10:58 AM To: Asterisk Users Mailing List Subject: [asterisk-users] RFC -- Improving the quality of the mailing lists The -user and -dev mailing lists are a valuable resource -- when they are not cluttered by posts unrelated to the charter of the lists. In my limited memory, this last weekend represents a new low in the relevant subject to noise ratio. Replying to requests with meaningless, misleading, or misspelled subject lines (I need help, asterisk help, Ntework Card) encourage careless posting and obfuscate useful replies from search engines. Also, while replying to such requests may seem helpful, some of the requests indicate such a lack of basic understanding that giving the answer is like giving a small child a very sharp knife when they ask for a slice of bread. For example: How do I delete these files that end in that squiggly thing in my current directory and all directories below? Since most of these users are probably running as root, a simple extra space here and a missed character there (rm --force --recursive /* ~ vs rm --force --recursive ./*~ can have catastrophic consequences. In an attempt to improve the quality of the lists, I propose the following: For a user's first 10 posts, they will receive a reply with a link to a web page and have to answer the following questions: 0) I acknowledge that I am asking for free help and I acknowledge that following the conventions below increase my chances of engaging another list member with relevant expertise and resolving my request. 1) I am posting a new request. a) My request cannot be answered on a more general list such as Beginning Unix, or on a distribution specific list. b) My request cannot be answered on a more specific list such as an AsteriskNow or Trixbox list. c) I have attempted to search for an answer using a search engine such as Google. d) I know what thread hijacking is and I created this request from scratch. e) I have created a meaningful subject line that indicates with as much specificity as reasonable which part of Asterisk I need help with and why. f) I am not posting a self-serving message directing someone to my product that would be better posted to the -biz list. g) I am not posting in HTML. h) I am posting in English. i) I am fluent in English or I have attempted to have someone who is review my request. j) I have run my request through my spell checking resources. or 2) I am posting a reply to a post. a) I know what top posting is and I am not ignoring the convention of the list. b) I am not posting a self-serving message directing someone to my product that would be better posted to the -biz list or only to the requester. c) I am not posting in HTML. d) I am posting in English. e) I am fluent in English or I have attempted to have someone who is review my post. f) I have trimmed the previous post down to just the point(s) I am replying to. g) I have run my request through my spell checking resources. For -dev, the following questions would be added: ) My post directly relates to changes in the Asterisk C source code. ) I am not reporting a bug or a posting a patch that should be directed to bugs.digium.com. Included in the web page would be the original message with the ability to change the list the message is to be posted to, the subject line, and the body of the message. Comments? Thanks in advance, Steve Edwards sedwa...@sedwards.com
Re: [asterisk-users] I need help
Yes, but is it agents a,b,c or a,b,etc? If huey, dewey and louie always get in, but Donald never does, something may be wrong with how Donald is set up. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bayardo Sanchez Sent: Tuesday, January 27, 2009 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] I need help only 3 On Tue, Jan 27, 2009 at 10:03 AM, Danny Nicholas da...@debsinc.com wrote: Is it the same 3 or the first 3? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bayardo Sanchez Sent: Tuesday, January 27, 2009 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] I need help 24 chanels On Tue, Jan 27, 2009 at 9:23 AM, Danny Nicholas da...@debsinc.com wrote: What is your call-limit set to in sip.conf? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bayardo Sanchez Sent: Tuesday, January 27, 2009 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] I need help The problem is I have 20 agents calling all but 3 of them get this error when calling On Tue, Jan 27, 2009 at 6:49 AM, Max Brooks m...@legatio.com wrote: Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' if you have made no changes to your asterisk configuration since it last worked Contact your service provider as they have an issue more likely though is you have a bad dial command. /telepathy -- Kind Regards Max Brooks - Developer Legatio Technologies Limited Phone: 01793 520 506 www.legatio.com, www.ftax.co.uk Legatio is part of the Callcredit Information Group: www.skipton.com, www.callcredit.co.uk, www.eurodirect.co.uk Legatio Technologies Limited, One Park Lane, Leeds, West Yorkshire, LS3 1EP Registered in England and Wales No. 4519902 - Powered by Legatio.com This message is confidential. It may not be disclosed to, or used by, anyone other than the addressee(s). If you receive this message in error, please advise us immediately using the email address i...@legatio.com. Internet e-mail is not necessarily secure. Legatio will not accept responsibility for alterations or additions to any e-mail message or attached documents that occur after transmission. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient.
Re: [asterisk-users] T.38
Hi Michel, it seems there is a codec translation in between, have you tried to avoid it setting the codec from g729 to ulaw? I personally make Asterisk use alaw/ulaw codecs when sending faxes without any kind of codec translation and it seems to work. Giorgio michel freiha wrote: Dear All, I'm trying to send Fax using T.38 protocol but the FAX is not going through..I'm getting the following error om /var/log/messages [Jan 27 16:46:31] WARNING[25435] channel.c: No path to translate from SIP/80.169.210.181-0896bfd0(4) to SIP/028949469-b7703d40(256) [Jan 27 16:46:40] WARNING[24349] channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw) [Jan 27 16:46:40] WARNING[24349] channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw) Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
On Tuesday 27 January 2009 09:57:54 Steve Edwards wrote: The -user and -dev mailing lists are a valuable resource -- when they are not cluttered by posts unrelated to the charter of the lists. In my limited memory, this last weekend represents a new low in the relevant subject to noise ratio. Replying to requests with meaningless, misleading, or misspelled subject lines (I need help, asterisk help, Ntework Card) encourage careless posting and obfuscate useful replies from search engines. Also, while replying to such requests may seem helpful, some of the requests indicate such a lack of basic understanding that giving the answer is like giving a small child a very sharp knife when they ask for a slice of bread. For example: How do I delete these files that end in that squiggly thing in my current directory and all directories below? Since most of these users are probably running as root, a simple extra space here and a missed character there (rm --force --recursive /* ~ vs rm --force --recursive ./*~ can have catastrophic consequences. In an attempt to improve the quality of the lists, I propose the following: For a user's first 10 posts, they will receive a reply with a link to a web page and have to answer the following questions: While I agree with your overall sentiment, I believe a few of these items are a bit over the top, and perhaps I'm reading this with more seriousness than it merits. i) I am fluent in English or I have attempted to have someone who is review my request. In many cases, this just isn't possible. While it would be nice to have all posts in the King's English, a great many users are in locales which don't have an English-speaking population. These are likely the only lists to which they have ready access which understand both enough English, as well as enough telephony knowledge to process their questions intelligently. j) I have run my request through my spell checking resources. Even I don't do this, and I know that I occasionally misspell some words. Included in the web page would be the original message with the ability to change the list the message is to be posted to, the subject line, and the body of the message. Comments? I think we'd be better off posting a regular FAQ, perhaps weekly, with some of these suggestions, as well as providing a link to that FAQ from the mailing list signup page, along with a STRONG suggestion to peruse the FAQ first. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
I meant digium.com. Yay for messups! It's been one of those weeks. Really. New to Aserisk 1.6 and find the 'installation tutorials' seem low to non existent. You go to the main Asterisk page (digium.org) and really just old install instructions for 1.2 are in the examples. Download links only give you asterisk itself and not dahdi or libpri which also are needed to run asterisk? It's very confusing to anyone who is new. Someone take notice! we need a link to instructions right of the main asterisk page. My 1st question is am I missing a good step-by step for 1.6 and how to compile/install it along with it's side components (dahdi/libpri)? when/if those side components are actually needed? When would you run asterisk without them entirely? 2nd question is for an IP/SIP only system do I only need DAHDI or do I need DAHDI and LIBPRI? Is libpri only needed if interfacing to a pri? Is 1.6 so cutting edge that I should not expect to find complete documentation (yet)like I seem to be expecting very easily? Thanks much! Steve -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
you can use any 1.4 how to but just use dahdi (both modules and tools) David 2009/1/27 Steve Gladden aster...@michiganbroadband.com I meant digium.com. Yay for messups! It's been one of those weeks. Really. New to Aserisk 1.6 and find the 'installation tutorials' seem low to non existent. You go to the main Asterisk page (digium.org) and really just old install instructions for 1.2 are in the examples. Download links only give you asterisk itself and not dahdi or libpri which also are needed to run asterisk? It's very confusing to anyone who is new. Someone take notice! we need a link to instructions right of the main asterisk page. My 1st question is am I missing a good step-by step for 1.6 and how to compile/install it along with it's side components (dahdi/libpri)? when/if those side components are actually needed? When would you run asterisk without them entirely? 2nd question is for an IP/SIP only system do I only need DAHDI or do I need DAHDI and LIBPRI? Is libpri only needed if interfacing to a pri? Is 1.6 so cutting edge that I should not expect to find complete documentation (yet)like I seem to be expecting very easily? Thanks much! Steve -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailinglists
-- In many cases, this just isn't possible. While it would be nice to -- have all -- posts in the King's English, a great many users are in locales which -- don't King's English??? Anyway - to quote Ralph Wigham Me fail English? That's unpossible!. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] German date format in voicemail emails
Tilghman Lesher schrieb: On Tuesday 27 January 2009 02:03:37 Johansson Olle E wrote: Minivoicemail actually has - multiple e-mail formats - locale support so you get the date in local language and format. Unfortunately, it's using setlocale(3), which is not thread-safe. Note that there is a new thread-safe locale interface in POSIX 2008. See the library calls newlocale(3), uselocale(3), and freelocale(3). Bug number? ;-) Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't start Asterisk after installing Digium G729 licence [SOLVED]
2009/1/27 Olivier oza-4...@myamail.com 2009/1/27 Olivier oza-4...@myamail.com Hi, I carefully followed instructions in README file lasting with : /root/register ... blabla asterisk -r CLI restart now Then asterisk -r fails with : # asterisk -r Asterisk 1.6.1-beta4, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Running as group 'asterisk' == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf == Found Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) # tail /var/log/asterisk/messages [Jan 27 08:17:40] WARNING[23451] chan_dahdi.c: Ignoring hasiax at line 39. [Jan 27 08:17:40] WARNING[23451] chan_dahdi.c: Ignoring hasmanager at line 47. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: G.729 transcoding module version 34, Copyright (C) 1999-2007 Digium, Inc. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: This module is supplied under a commercial license granted by Digium, Inc. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: Please see the full license text supplied by the accompanying [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: register utility, or ask for a copy from Digium. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: This product includes software developed by the OpenSSL Project [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: Copyright (C) 1998-2006 The OpenSSL Project Before opening a ticket to Digium, is there something obvious I missed ? Google didn't show much hint ... After reading this see register utility line, I ran /root/register once again but it didn't change anything ... Cheers Including noload = codec_g279a.so in /etc/asterisk/modules.conf at least allow me to restart asterisk successfully. Changing ownership of /var/lib/asterisk/licences files to asterisk:asterisk didn't change the fact it didn't start. It was a bug in 1.6.1 beta4. Upgrading to current (1.6.0.5xxx) fixed it. Thanks to Digium support ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
I think we'd be better off posting a regular FAQ, perhaps weekly, with some of these suggestions, as well as providing a link to that FAQ from the mailing list signup page, along with a STRONG suggestion to peruse the FAQ first. I agree with this 100% I'm still pretty new to the mailing lists myself. I don't consider myself a novice Asterisk user, but one of my biggest 'complaints' is the lack of a well documented FAQ or Manual for Asterisk. (Unless one is willing to buy or read O'Reilly's Book - http://www.asteriskdocs.org - which quickly will be outdated again.) I have made it a personal aim to document all my findings in a blog, so that it's at least searchable by others through Google, in hopes that others might find it useful. But if we had a REGULARLY updated FAQ/Manual ... I think that would greatly cut down on the clutter posts. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
It seems to me that everything one may want to know would be contained on voip-info.org People don't ask stupid questions because of a lack of a FAQ to read, they ask stupid questions because they're too lazy do to the footwork. Robert Broyles wrote: I think we'd be better off posting a regular FAQ, perhaps weekly, with some of these suggestions, as well as providing a link to that FAQ from the mailing list signup page, along with a STRONG suggestion to peruse the FAQ first. I agree with this 100% I'm still pretty new to the mailing lists myself. I don't consider myself a novice Asterisk user, but one of my biggest 'complaints' is the lack of a well documented FAQ or Manual for Asterisk. (Unless one is willing to buy or read O'Reilly's Book - http://www.asteriskdocs.org - which quickly will be outdated again.) I have made it a personal aim to document all my findings in a blog, so that it's at least searchable by others through Google, in hopes that others might find it useful. But if we had a REGULARLY updated FAQ/Manual ... I think that would greatly cut down on the clutter posts. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Nortel integration via SIP protocol
You will need to have a Nortel NRS server in your network. Sent from my iPhone Eric Moniz On Jan 27, 2009, at 10:17 AM, Pablo Bernasconi pbernasc...@isbel.com.uy wrote: Hi, I need to integrate my Asterisk with a Nortel Meridian 11, but I can ´t use PRI, Analog lines, etc. It has to be via SIP protocol, and th ere is few information about this type of integration. Could someone please help me?? Thanks, Pablo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G726 Codec
michel freiha schrieb: I would like to ask please about how I can force asterisk to send all G726 codecs without translation... Huh? g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- -- -- gsm- -222 21 3- -- 2- ulaw- 3-12 21 3- -- 2- alaw- 31-2 21 3- -- 2- g726aal2- 322- 21 3- -- 1- adpcm- 3222 -1 3- -- 2- slin- 2111 1- 2- -- 1- lpc10- 3222 21 -- -- 2- g729- ---- -- -- -- -- speex- ---- -- -- -- -- ilbc- ---- -- -- -- -- g726- 3221 21 3- -- -- g722- ---- -- -- -- -- Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
I wouldn't say that voip-info.org has everything that a person would want to know. This is especially true of any recent changes to dialplan applications (and their available options) Voip-info.org is a great place to start, and often you will find an answer there. But not always. People are always going to ask stupid questions. There's no way to avoid that. But I do believe the documentation is somewhat lacking. Mik Cheez wrote: It seems to me that everything one may want to know would be contained on voip-info.org People don't ask stupid questions because of a lack of a FAQ to read, they ask stupid questions because they're too lazy do to the footwork. Robert Broyles wrote: I think we'd be better off posting a regular FAQ, perhaps weekly, with some of these suggestions, as well as providing a link to that FAQ from the mailing list signup page, along with a STRONG suggestion to peruse the FAQ first. I agree with this 100% I'm still pretty new to the mailing lists myself. I don't consider myself a novice Asterisk user, but one of my biggest 'complaints' is the lack of a well documented FAQ or Manual for Asterisk. (Unless one is willing to buy or read O'Reilly's Book - http://www.asteriskdocs.org - which quickly will be outdated again.) I have made it a personal aim to document all my findings in a blog, so that it's at least searchable by others through Google, in hopes that others might find it useful. But if we had a REGULARLY updated FAQ/Manual ... I think that would greatly cut down on the clutter posts. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] USA BRI -- any hope at all?
Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have found. What I did find left me with the impression that USA 'BRI', uh, '2B1Q' protocol(?) is not supported by *any* hardware vendor, at all, period, nor is it tested and proved in the software... stack(?), in one related branch or another on the OS side. A couple of direct inquiries to card vendors have dead-ended with a flat no, or requests for development funds(!) -- apparently there is code for one card, one vendor, that runs against 'bristuff', or did at one time, but wasn't maintained through several Asterisk releases (if the code was even released to the community... IDK). Is this common, that someone codes to their chip on their card and sells it to one or two consumers, then lets it drop and never gives the code up for continued development? (It seems contrary to GNU/Linux licensing conventions, but, again, I'm not paid as a software developer. I just think they might have sold more cards with a less proprietary approach.) Anyway, can I, with confidence, state (to the $employer) that Asterisk on linux via USA 'BRI' digital lines simply isn't possible? (In that, obviously, I can't pay for development nor do beta testing, each with vague hope that it might work okay someday...) If this is the case, then I must use multiple analog lines to access PSTN, or pay premium for 'PRI' pipes (80% of which we will never need)... is that about correct? Thanks in advance for any pointers, specific RTFM suggestions, any help appreciated. If there is a different list to post this query to, I'm not (yet) aware of it. Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
Hi Steve - New to Aserisk 1.6 and find the 'installation tutorials' seem low to non existent. Welcome to Open Source! Seriously, look at the README files accompanying asterisk, dahdi, and libpri. They will give you compilation/installation instructions. You can also search this list with google: Search term site:lists.digium.com Someone take notice! we need a link to instructions right of the main asterisk page. If you have a need for documentation, you're more than welcome to write it (once you've figured out how to install asterisk). We all contribute however we're able. Well, some of us do. Now to answer your questions: My 1st question is am I missing a good step-by step for 1.6 and how to compile/install it along with it's side components (dahdi/libpri)? when/if those side components are actually needed? When would you run asterisk without them entirely? 2nd question is for an IP/SIP only system do I only need DAHDI or do I need DAHDI and LIBPRI? If you have no dahdi compatible hardware, you don't need dahdi. The one exception to this is meetme, for which you need a dahdi timing source. You can use the dummy timing driver. Is libpri only needed if interfacing to a pri? Yes, mostly. I think you may need it if you have any card that takes a T1/E1. I think you may also need it for BRI cards. Is 1.6 so cutting edge that I should not expect to find complete documentation (yet)like I seem to be expecting very easily? The short answer is yes, given the glacial pace of documentation creation, 1.6 is that cutting edge. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
Michael Higgins wrote: At least here in Canada - DSL just seems to have killed BRI - you practically have to know the secret handshake to even be allowed to provision one any more. It killed it as an internet transport which was its most widespread use, however its many benefits as a digital phone line are being largely ignored. I barked up the same tree you are barking for a while and just gave up - lots of you could buy this and try it, but no proven solution. Kind of expensive to get a line put in and buy hardware for a maybe. Years ago we had tons of BRI circuits around I could have tried this on, but thats long gone. Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have found. What I did find left me with the impression that USA 'BRI', uh, '2B1Q' protocol(?) is not supported by *any* hardware vendor, at all, period, nor is it tested and proved in the software... stack(?), in one related branch or another on the OS side. A couple of direct inquiries to card vendors have dead-ended with a flat no, or requests for development funds(!) -- apparently there is code for one card, one vendor, that runs against 'bristuff', or did at one time, but wasn't maintained through several Asterisk releases (if the code was even released to the community... IDK). Is this common, that someone codes to their chip on their card and sells it to one or two consumers, then lets it drop and never gives the code up for continued development? (It seems contrary to GNU/Linux licensing conventions, but, again, I'm not paid as a software developer. I just think they might have sold more cards with a less proprietary approach.) Anyway, can I, with confidence, state (to the $employer) that Asterisk on linux via USA 'BRI' digital lines simply isn't possible? (In that, obviously, I can't pay for development nor do beta testing, each with vague hope that it might work okay someday...) If this is the case, then I must use multiple analog lines to access PSTN, or pay premium for 'PRI' pipes (80% of which we will never need)... is that about correct? Thanks in advance for any pointers, specific RTFM suggestions, any help appreciated. If there is a different list to post this query to, I'm not (yet) aware of it. Cheers, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Gladden wrote: Is 1.6 so cutting edge that I should not expect to find complete documentation (yet)like I seem to be expecting very easily? Most of what is applicable to 1.4 is applicable to 1.6. I'm running 1.6 without any hiccups -- YMMV. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJf0s4CFu3bIiwtTARApXZAJ9kse5IimuCkzFG7FqlmQRzbxOlGgCfY8wA CeGjEgTSVagAovNT/TaNjDM= =z1O2 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
It seems to me that everything one may want to know would be contained on voip-info.org Hmm. Dangerous statement. There are many things on the WIKI that are quite outdated, and a great many other things that aren't there at all. People don't ask stupid questions because of a lack of a FAQ to read, they ask stupid questions because they're too lazy do to the footwork. True. They may not know how to look up the answers to the stupid questions, though. I think a FAQ would help greatly in these cases. - Noah Robert Broyles wrote: I think we'd be better off posting a regular FAQ, perhaps weekly, with some of these suggestions, as well as providing a link to that FAQ from the mailing list signup page, along with a STRONG suggestion to peruse the FAQ first. I agree with this 100% I'm still pretty new to the mailing lists myself. I don't consider myself a novice Asterisk user, but one of my biggest 'complaints' is the lack of a well documented FAQ or Manual for Asterisk. (Unless one is willing to buy or read O'Reilly's Book - http://www.asteriskdocs.org - which quickly will be outdated again.) I have made it a personal aim to document all my findings in a blog, so that it's at least searchable by others through Google, in hopes that others might find it useful. But if we had a REGULARLY updated FAQ/Manual ... I think that would greatly cut down on the clutter posts. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
Instead you could always get a SIP/IAX provider. On Jan 27, 2009, at 11:56 AM, Jon Pounder wrote: Michael Higgins wrote: At least here in Canada - DSL just seems to have killed BRI - you practically have to know the secret handshake to even be allowed to provision one any more. It killed it as an internet transport which was its most widespread use, however its many benefits as a digital phone line are being largely ignored. I barked up the same tree you are barking for a while and just gave up - lots of you could buy this and try it, but no proven solution. Kind of expensive to get a line put in and buy hardware for a maybe. Years ago we had tons of BRI circuits around I could have tried this on, but thats long gone. Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have found. What I did find left me with the impression that USA 'BRI', uh, '2B1Q' protocol(?) is not supported by *any* hardware vendor, at all, period, nor is it tested and proved in the software... stack(?), in one related branch or another on the OS side. A couple of direct inquiries to card vendors have dead-ended with a flat no, or requests for development funds(!) -- apparently there is code for one card, one vendor, that runs against 'bristuff', or did at one time, but wasn't maintained through several Asterisk releases (if the code was even released to the community... IDK). Is this common, that someone codes to their chip on their card and sells it to one or two consumers, then lets it drop and never gives the code up for continued development? (It seems contrary to GNU/ Linux licensing conventions, but, again, I'm not paid as a software developer. I just think they might have sold more cards with a less proprietary approach.) Anyway, can I, with confidence, state (to the $employer) that Asterisk on linux via USA 'BRI' digital lines simply isn't possible? (In that, obviously, I can't pay for development nor do beta testing, each with vague hope that it might work okay someday...) If this is the case, then I must use multiple analog lines to access PSTN, or pay premium for 'PRI' pipes (80% of which we will never need)... is that about correct? Thanks in advance for any pointers, specific RTFM suggestions, any help appreciated. If there is a different list to post this query to, I'm not (yet) aware of it. Cheers, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Muted sound on a Linksys 962
Hi, One of our customers has an issue with the callee not being able to hear them. It seems to happen very frequently on one number in particular where there are about 3 IVR menus to dial through before getting to a live person. However, this does not happen on every call. Running tcpdump on the RTP packets, I can see that RTP is setting sent, but the values in the packet are all very close to 0xFF or 0x7F (which is 0 or -1 once you translate it using G.711). Could this be some issue with the phone muting audio because it's stuck sending DTMF? DTMFMode is rfc2833 on the Asterisk side and Auto on the phone side. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailinglists
On Tuesday 27 January 2009 10:46:40 Andrew Thomas wrote: -- In many cases, this just isn't possible. While it would be nice to -- have all -- posts in the King's English, a great many users are in locales which -- don't King's English??? I would have said Queen's English, but that evokes Freddy Mercury. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, Jan 27, 2009 at 09:49:41AM -0800, Michael Higgins wrote: What I did find left me with the impression that USA 'BRI', uh, '2B1Q' protocol(?) is not supported by *any* hardware vendor, at all, period, nor is it tested and proved in the software... stack(?), in one related branch or another on the OS side. To the best of my understanding, latest Asterisk should support it through chan_dahdi . No need for extra bristuff or whatever. But this needs some testing. A couple of direct inquiries to card vendors have dead-ended with a flat no, or requests for development funds(!) -- apparently there is code for one card, one vendor, that runs against 'bristuff', or did at one time, but wasn't maintained through several Asterisk releases (if the code was even released to the community... IDK). Actually from the little I can tell, BRI there behaves very much like PRI. No extra ptmp complications as in the rest of the world. And then you have to recall that chan_dahdi's libpri was developed by US people originally, and hence actually supports the crazy mess of ISDN signalling there ;-) It seems, though you may need to do some custom wiring. Is this common, that someone codes to their chip on their card and sells it to one or two consumers, then lets it drop and never gives the code up for continued development? (It seems contrary to GNU/Linux licensing conventions, but, again, I'm not paid as a software developer. I just think they might have sold more cards with a less proprietary approach.) Anyway, can I, with confidence, state (to the $employer) that Asterisk on linux via USA 'BRI' digital lines simply isn't possible? (In that, obviously, I can't pay for development nor do beta testing, each with vague hope that it might work okay someday...) A cheap HFC-S -based BRI card (the type supported by virtually all candidate ISDN/BRI channels for Asterisk. Except chan_capi, I guess) shouldn't cost you much (naturally I personally would prefer you used our hardware, but then again, you may have other considerations) I figure that 20$-30$ or so. I have no idea about other setup costs (getting a line, etc.). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non existent. I first looked at * about four months ago and rapidly came to the same conclusion. Even with the O-Reilly book, which I purchased in paper, although it is freely downloadable, I feel there is a huge dearth of information. As I have become a bit involved, I find there is more than meets the eye, but it is spread across the entire internet! So far I am not aware of anything that fits any of three categories I feel are essential: 1. A good tutorial with enough detail to allow a person with a CS degree, years of telephony experience and limited Linux experience (myself) to install and configure a reasonable * system (something more complex that an FXO or two and a couple of SIP phones. 2. A reference guide that lists all commands and options with explanations of why they are useful and how to use them. Even the book doesn't attempt to touch this one. Such a reference needs to include things like Dahdi and other pieces that aren't strictly part of * but without which few installations could exist. 3. A decent cross-reference that can quickly allow someone to find the scattered information available on the web. Even this mailing list is so hopelessly linear in nature compared to most other newsgroups I am involved in as to be almost useless to me. My conclusion after installing a worthless * demo (that actually does allow two SIPs to talk to each other) is that Asterisk is not of any value to anyone other than a person who makes a full time career out of running Asterisk systems. I've installed and maintained several traditional PABXs and even wrote the control firmware (in 6502 assembly) for one, with sizes from 6 stations to 300 stations, including things like DID. It was kindergarten compared to Asterisk, and primarily because of the huge information vacuum. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
It seems to me that everything one may want to know would be contained on voip-info.org My own experience is that it covers a very broad spectrum (far broader than Asterisk) and in a rather terse manner. I have spent an hour or two at a time pouring over a topic there and come away little more enlightened than when I started. Most people who know enough to create useful entries there, assume too much of the reader. They assume that everyone reading the post works with Linux 40 hours a week at the command line level, and only needs a few VoIP clues to take an idea and run with it. A better assumption would be that they know how to log on. It shouldn't even be assumed that they know the difference between suand su - I realize that this is challenging because different distros do things different ways. That is another topic of its own, but is also one of the banes of Linux that is hurting its usability considerably. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
I'm in the same boat and have been looking at this for several months, but haven't actually jumped in, hands-on, yet. No, I don't think the situation is as dismal as you paint it, although the lack of appropriate marketing for BRI in the US has all but killed it here, making it relatively unattractive to vendors. I have been advised of several cards that support it, but they were two or four port cards that were serious overkill for my application, and expectedly pricy as a result. I am trying to use an HFC card, and have been advised that it is possible, but will take some digging and experimenting. As near as I have been able to figure out, mISDN is the appropriate place to begin, but I haven't tried to sort it out, yet. There are a couple of other paths that might lead to a solution, but they seem to be less supported than mISDN at present (if that is even possible). It sounds like Dahdi is moving towards that, but I don't think its quite there, yet. Qwest is one of the few providers I know of that prices BRI reasonably. I have one up and running on a TA and use it for my voice service. As soon as I can figure out mISDN, I plan to move it to Asterisk. (I'll keep a TA around for backup). Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
If you find something on a WIKI that is outdated, guess what you have an opportunity to do . . . Noah Miller wrote: It seems to me that everything one may want to know would be contained on voip-info.org Hmm. Dangerous statement. There are many things on the WIKI that are quite outdated, and a great many other things that aren't there at all. People don't ask stupid questions because of a lack of a FAQ to read, they ask stupid questions because they're too lazy do to the footwork. True. They may not know how to look up the answers to the stupid questions, though. I think a FAQ would help greatly in these cases. - Noah Robert Broyles wrote: I think we'd be better off posting a regular FAQ, perhaps weekly, with some of these suggestions, as well as providing a link to that FAQ from the mailing list signup page, along with a STRONG suggestion to peruse the FAQ first. I agree with this 100% I'm still pretty new to the mailing lists myself. I don't consider myself a novice Asterisk user, but one of my biggest 'complaints' is the lack of a well documented FAQ or Manual for Asterisk. (Unless one is willing to buy or read O'Reilly's Book - http://www.asteriskdocs.org - which quickly will be outdated again.) I have made it a personal aim to document all my findings in a blog, so that it's at least searchable by others through Google, in hopes that others might find it useful. But if we had a REGULARLY updated FAQ/Manual ... I think that would greatly cut down on the clutter posts. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] German date format in voicemail emails
On Tuesday 27 January 2009 10:54:37 Philipp Kempgen wrote: Tilghman Lesher schrieb: On Tuesday 27 January 2009 02:03:37 Johansson Olle E wrote: Minivoicemail actually has - multiple e-mail formats - locale support so you get the date in local language and format. Unfortunately, it's using setlocale(3), which is not thread-safe. Note that there is a new thread-safe locale interface in POSIX 2008. See the library calls newlocale(3), uselocale(3), and freelocale(3). Bug number? ;-) http://bugs.digium.com/view.php?id=14333 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of themailinglists
We all need the Univeral Language translators from Star Trek. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Tuesday, January 27, 2009 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RFC -- Improving the quality of themailinglists On Tuesday 27 January 2009 10:46:40 Andrew Thomas wrote: -- In many cases, this just isn't possible. While it would be nice to -- have all -- posts in the King's English, a great many users are in locales which -- don't King's English??? I would have said Queen's English, but that evokes Freddy Mercury. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Muted sound on a Linksys 962
This worked for me Exten = s,1,Answer() Exten = s,n,Dial(Zap/g1/w5551212) What happens is that * doesn't go full duplex until it does a Native Bridge. The Answer Command creates a temporary bridge until the real one can take effect. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Tuesday, January 27, 2009 12:38 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Muted sound on a Linksys 962 Hi, One of our customers has an issue with the callee not being able to hear them. It seems to happen very frequently on one number in particular where there are about 3 IVR menus to dial through before getting to a live person. However, this does not happen on every call. Running tcpdump on the RTP packets, I can see that RTP is setting sent, but the values in the packet are all very close to 0xFF or 0x7F (which is 0 or -1 once you translate it using G.711). Could this be some issue with the phone muting audio because it's stuck sending DTMF? DTMFMode is rfc2833 on the Asterisk side and Auto on the phone side. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
On Tue, Jan 27, 2009 at 11:24:38AM -0700, Wilton Helm wrote: New to Aserisk 1.6 and find the 'installation tutorials' seem low to non existent. I first looked at * about four months ago and rapidly came to the same conclusion. Even with the O-Reilly book, which I purchased in paper, although it is freely downloadable, I feel there is a huge dearth of information. As I have become a bit involved, I find there is more than meets the eye, but it is spread across the entire internet! So far I am not aware of anything that fits any of three categories I feel are essential: 1. A good tutorial with enough detail to allow a person with a CS degree, years of telephony experience and limited Linux experience (myself) to install and configure a reasonable * system (something more complex that an FXO or two and a couple of SIP phones. Occasionally someone writes such a HOWTO. It varies between versions and by your setup. 2. A reference guide that lists all commands and options with explanations of why they are useful and how to use them. Even the book doesn't attempt to touch this one. Such a reference needs to include things like Dahdi and other pieces that aren't strictly part of * but without which few installations could exist. Checked for 1.6.0 and on. See asterisk.pdf (and later also an HTML copy) under doc/ . As for DAHDI: a starting point would be the README files included in dahdi-linux and dahdi-tools . See them also in: http://docs.tzafrir.org.il/dahdi-linux/ http://docs.tzafrir.org.il/dahdi-tools/ Or use 'make docs' -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
At 09:30 AM 1/27/2009, you wrote: People are always going to ask stupid questions. For me it's not so much the stupid questions as the expectations that we're here to solve their problems according to their needs. If that continues to happen and the noise level gets high enough those that have the most to offer will leave and all will be lost. Maybe there needs to be a beginner list and posting on this becomes invite only from people who participate on that list. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
Wilton Helm wrote: [snip] My conclusion after installing a worthless * demo (that actually does allow two SIPs to talk to each other) is that Asterisk is not of any value to anyone other than a person who makes a full time career out of running Asterisk systems. I've installed and maintained several traditional PABXs and even wrote the control firmware (in 6502 assembly) for one, with sizes from 6 stations to 300 stations, including things like DID. It was kindergarten compared to Asterisk, and primarily because of the huge information vacuum. My conclusion, after installing an interesting * demo (that actually does allow two SIPs to talk to each other) in late 2004 is that Asterisk is of immense value to me and the company that I work for, having saved us well over $100,000 (if not more) over the last four years. I am not a person who makes a full time career out of running Asterisk - quite the opposite. I am employed to write business applications, not maintain * systems. I had zero knowledge of PABX's and telecoms before I implemented *, and the vacuum that you refer to provided me with everything I needed to implement a system that today is making over 300,000 call attempts per month with 30 agents and 140 extensions, with call monitoring, recording and voicemail. YMMV. Mine certainly did. For the better. Julian. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
Ira wrote: At 09:30 AM 1/27/2009, you wrote: People are always going to ask stupid questions. For me it's not so much the stupid questions as the expectations that we're here to solve their problems according to their needs. If that continues to happen and the noise level gets high enough those that have the most to offer will leave and all will be lost. Maybe there needs to be a beginner list and posting on this becomes invite only from people who participate on that list. Ira And which kind soul is going to post on the beginner list to help beginners, but still be annoyed to the point that he'd leave the non-beginner list because of all the beginner questions? And who does the inviting? Suddenly, I see poor John Todd having wy too much to do. ;) N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
Wilton Helm wrote: [snip] My conclusion after installing a worthless * demo (that actually does allow two SIPs to talk to each other) is that Asterisk is not of any value to anyone other than a person who makes a full time career out of running Asterisk systems. I've installed and maintained several traditional PABXs and even wrote the control firmware (in 6502 assembly) for one, with sizes from 6 stations to 300 stations, including things like DID. It was kindergarten compared to Asterisk, and primarily because of the huge information vacuum. I'm impressed that you picked up 6502 assembly out of an even larger vaccum considering there was no 'net back then to help at all. Did you install a PBX on an Atari? :) There is an immense amount of information about Asterisk and Linux in general on the net, and it just requires diligence, patience, and an open mind to find and utilize. Even better, read the source if you have questions. Lots of READMEs, comments, and other lists to help specifically with development. It certainly helps to be Unix inclined, and if you have no interest in moving in that direction, you are probably better off with traditional systems. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
To the best of my understanding, latest Asterisk should support it through chan_dahdi . No need for extra bristuff or whatever. But this needs some testing. Any chance I could get some information on how to set it up and use it (keeping in mind that I have limited Asterisk experience and no experience with zaptel or mISDN or BRIstuff)? I have an NT1 BRI here that is up and running and an HFC card and an F9 Linux with *1.6. The BRI can easily be moved to the HFC card for testing. I installed 1.6 from an F9 Yum distro, but I could probably figure out how to download and make the latest from source if necessary. I did the F9 installation and run it, but I'm not real strong on Linux. I mostly do 80186 embedded development in C and assembly. BRI there behaves very much like PRI. No extra ptmp complications as in the rest of the world. I never dealt with PRI, so I can't say for sure, but you are correct that it lacks ptmp, etc. It seems, though you may need to do some custom wiring. AFAIK, only that it is delivered in U form and some cards (including mine) require S/T, but I already have a converter for that. I am anxious to get my BRI into Asterisk, I just have been overwhelmed at the terse mISDN documentation and haven't jumped in. Having it in dahdi would at least mean one less separate piece to mess with. If I can get some documentation or a bit of handholding, I will take the plunge, report any problems and document the end result (assuming it is successful and worth documenting). Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
Thanks for the reply. I have looked at the links you provided and I think they will be useful. I may have some issues with drivers for the HFC, but I guess I won't know until I try it. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
YMMV. Mine certainly did. For the better. My comments were more negative than I intended. My installation is worthless at this point because it is only a cookbook example and I haven't tried to modify it to meet my needs. I didn't intend to imply that Asterisk is worthless, just that I've only gotten to the point of a trivial demo. My main concern is that the documentation isn't for the faint of heart. If one doesn't devote many hours, on a regular, ongoing basis, they may never get to the point of understanding it enough to apply it to a real-world situation. The more I explore and the more feedback I get, the more I find is there. I just got a very nice posting from Tzafir showing me a web domain I didn't even know existed. Not surprising, it is a lot like Linux--everyone has there own idea of what is needed and how it should be done, so it becomes a monster that is hard to get a handle on. From what I've seen so far, the commands far exceed any commercial PABX I've ever used or evaluated. It is very powerful, but the learning curve is immense, and I'm both a CS professional and a telephony professional. I'm not abandoning it by any means, but am frustrated at even where to jump in. I excitedly bought the O-Reily book, only to find that for all 1000 pages, it never provided anything that could be considered a reference manual and that its tutorials weren't even a good fit to my needs. It did get me two SIP phones talking to each other and to a softphone, but only after hours of experimenting with SIP phone settings and contacts with the manufacturers (who knew even less about VoIP). I think part of the problem is that the only people who know enough about * to address the documentation problems are busy either developing hardware and software for it or using it to run their businesses and don't have time to address the documentation problem, which is understandable. Also, once a person gets to that level of knowledge, its easy to forget how little a newcomer knows and leave out a lot of necessary details. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
I wonder if BRI would have gotten traction if it offered PRI functionality (DID's and aggregation of multiple spans). Even TODAY I would drop many of my sip trunks for such hypothetical BRI trunks for locations where a full PRI is too much capacity. That's the bane of the PRI: Welcome to Joe's coffee shop... Oh, I'm sorry we only sell coffee in 40 gallon drums. You have to buy a whole drum even if you only want a cup--or a sip -Karl - Original Message - From: Jerry Jones jjo...@danrj.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 27, 2009 12:27 PM Subject: Re: [asterisk-users] USA BRI -- any hope at all? Instead you could always get a SIP/IAX provider. On Jan 27, 2009, at 11:56 AM, Jon Pounder wrote: Michael Higgins wrote: At least here in Canada - DSL just seems to have killed BRI - you practically have to know the secret handshake to even be allowed to provision one any more. It killed it as an internet transport which was its most widespread use, however its many benefits as a digital phone line are being largely ignored. I barked up the same tree you are barking for a while and just gave up - lots of you could buy this and try it, but no proven solution. Kind of expensive to get a line put in and buy hardware for a maybe. Years ago we had tons of BRI circuits around I could have tried this on, but thats long gone. Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have found. What I did find left me with the impression that USA 'BRI', uh, '2B1Q' protocol(?) is not supported by *any* hardware vendor, at all, period, nor is it tested and proved in the software... stack(?), in one related branch or another on the OS side. A couple of direct inquiries to card vendors have dead-ended with a flat no, or requests for development funds(!) -- apparently there is code for one card, one vendor, that runs against 'bristuff', or did at one time, but wasn't maintained through several Asterisk releases (if the code was even released to the community... IDK). Is this common, that someone codes to their chip on their card and sells it to one or two consumers, then lets it drop and never gives the code up for continued development? (It seems contrary to GNU/ Linux licensing conventions, but, again, I'm not paid as a software developer. I just think they might have sold more cards with a less proprietary approach.) Anyway, can I, with confidence, state (to the $employer) that Asterisk on linux via USA 'BRI' digital lines simply isn't possible? (In that, obviously, I can't pay for development nor do beta testing, each with vague hope that it might work okay someday...) If this is the case, then I must use multiple analog lines to access PSTN, or pay premium for 'PRI' pipes (80% of which we will never need)... is that about correct? Thanks in advance for any pointers, specific RTFM suggestions, any help appreciated. If there is a different list to post this query to, I'm not (yet) aware of it. Cheers, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
On Tue, 2009-01-27 at 10:13 -0700, Robert Broyles wrote: I'm still pretty new to the mailing lists myself. I don't consider myself a novice Asterisk user, but one of my biggest 'complaints' is the lack of a well documented FAQ or Manual for Asterisk. Asterisk is truly an open-source community, and that pertains to documentation as well. The quality and quantity of the documentation depends heavily on contribution from the community at large. Digium has and will continue to put resources towards Asterisk documentation, but every contribution from the community at large helps. (Unless one is willing to buy or read O'Reilly's Book - http://www.asteriskdocs.org - which quickly will be outdated again.) Alas, you've mentioned the one thing that both makes me happy and sad at the same time. Happy that people find it useful, and that O'Reilly was kind enough to let us publish it under a Creative Commons license (and put the PDF on the web for free!)... and sad that it takes so much time and effort to keep up to date. (And just for the record, the time that the other authors and I spend on writing the O'Reilly book is our own personal time -- I'm not working on it during company time!) I have made it a personal aim to document all my findings in a blog, so that it's at least searchable by others through Google, in hopes that others might find it useful. But if we had a REGULARLY updated FAQ/Manual ... I think that would greatly cut down on the clutter posts. If you're interested and serious about writing, join the asterisk-docs mailing list and let's try to get something started. I've been beating the documentation drum for almost seven years now, and I'd love to see the -docs mailing list come back to life. -- Jared Smith Digium, Inc. | Training Manager ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
On Jan 27, 2009, at 10:50 PM, Wilton Helm wrote: YMMV. Mine certainly did. For the better. My comments were more negative than I intended. My installation is worthless at this point because it is only a cookbook example and I haven't tried to modify it to meet my needs. I didn't intend to imply that Asterisk is worthless, just that I've only gotten to the point of a trivial demo. My main concern is that the documentation isn't for the faint of heart. If one doesn't devote many hours, on a regular, ongoing basis, they may never get to the point of understanding it enough to apply it to a real-world situation. The more I explore and the more feedback I get, the more I find is there. I just got a very nice posting from Tzafir showing me a web domain I didn't even know existed. Not surprising, it is a lot like Linux--everyone has there own idea of what is needed and how it should be done, so it becomes a monster that is hard to get a handle on. From what I've seen so far, the commands far exceed any commercial PABX I've ever used or evaluated. It is very powerful, but the learning curve is immense, and I'm both a CS professional and a telephony professional. I'm not abandoning it by any means, but am frustrated at even where to jump in. I excitedly bought the O-Reily book, only to find that for all 1000 pages, it never provided anything that could be considered a reference manual and that its tutorials weren't even a good fit to my needs. It did get me two SIP phones talking to each other and to a softphone, but only after hours of experimenting with SIP phone settings and contacts with the manufacturers (who knew even less about VoIP). I think part of the problem is that the only people who know enough about * to address the documentation problems are busy either developing hardware and software for it or using it to run their businesses and don't have time to address the documentation problem, which is understandable. Also, once a person gets to that level of knowledge, its easy to forget how little a newcomer knows and leave out a lot of necessary details. Very true, but what you have to understand is that we all started out with zero knowledge about *. One thing i have learned over the years with * ( linux/unix), that there is plenty of info readily available as long as you know where to look. Best place to start is always the README file or man pages.Also, apart from google I found voip-info.org to be an excellent online recourse. If you are not comfortable with using linux, then i would suggest using something like trixbox (www.trixbox.org ) which can be configured via a webinterface. userfriendly Wilton n___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailinglists
On 1/27/09, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: I would have said Queen's English, but that evokes Freddy Mercury. ...and Freddy Mercury evokes Kevin Fleming. Perfect - we're back on topic! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
How pompous are we now? What happened to the 'open source community'? There's a give and take involved; you answer questions you know how to answer in the hopes that someone with greater experience and knowledge of the software will answer your questions. Yikes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira Sent: Tuesday, January 27, 2009 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RFC -- Improving the quality of the mailing lists At 09:30 AM 1/27/2009, you wrote: People are always going to ask stupid questions. For me it's not so much the stupid questions as the expectations that we're here to solve their problems according to their needs. If that continues to happen and the noise level gets high enough those that have the most to offer will leave and all will be lost. Maybe there needs to be a beginner list and posting on this becomes invite only from people who participate on that list. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
I'm impressed that you picked up 6502 assembly out of an even larger vaccum considering there was no 'net back then to help at all. Did you install a PBX on an Atari? No, I interfaced a Rockwell AIM to a 300 station Philips electromechanical PABX (designed and built about 100 interface cards, including DTMF receivers) and then wrote all the call processing code. The Rockwell AIM did come with manuals that completely documented both the hardware interface and the instruction set. In the days before the 'net, such paperwork was mandatory. it just requires diligence, patience, I'm trying. It certainly helps to be Unix inclined, Unix was barely out of Bell Labs when I got my CS degree and we never saw it, so I am at a disadvantage. I have worked a bit with a couple of Unix installations since and do have a computer running Fedora 9 and one that is supposed to be running Fedora 10 64 bit if I can ever get past a kernel bug, so I am trying to come up to speed. I am a lot more familiar with what to do after the reset vector on an 80186, or the inner workings of a protocol stack. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
On Tue, Jan 27, 2009 at 12:50:42PM -0700, Wilton Helm wrote: I just got a very nice posting from Tzafir showing me a web domain I didn't even know existed. It only includes documentation generated by 'make docs' . And is actually linked from the README itself. I'm not abandoning it by any means, but am frustrated at even where to jump in. I excitedly bought the O-Reily book, only to find that for all 1000 pages, it never provided anything that could be considered a reference manual It actually does contain references of all applicaitons, CLI commands, and such. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
I wonder if BRI would have gotten traction if it offered PRI functionality I can't say for sure, and don't even know the differences in functionality, but you may be right. When I last ordered DID I couldn't justify PRI so brought it in as analog. At that point in time and with that LEC PRI wasn't cost effective, even if I filled it up. I could have filled a 24B but it would have left my entire facility at the mercy of a single circuit, which isn't very smart. To me the thing that did the most to insure the demise of BRI in the US was the insane pricing. Most LECs saw it as a large cash cow and priced it with large margins (keep in mind it is cheaper for a CO to export two LDNs on a BRI than on two POTS, but most priced a BRI at about 4x a POTS). Most LECs only offered it as measured service with stiff per minute charges even on local calls. Many charged stiff rates for connect time in data mode (in fact some would get around this by treating data as voice). Their greed backfired, and of course when DSL offered more bandwidth for considerably less money, the bottom fell out of the data side. The only LEC I knew that didn't go down this path was QWEST. They priced it flat rate at rates that were competitive with POTS and I leased them for every site I managed. They offered higher quality and more features than POTS. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, Jan 27, 2009 at 12:49 PM, Michael Higgins li...@evolone.org wrote: Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have found. What I did find left me with the impression that USA 'BRI', uh, '2B1Q' protocol(?) is not supported by *any* hardware vendor, at all, period, nor is it tested and proved in the software... stack(?), in one related branch or another on the OS side. A couple of direct inquiries to card vendors have dead-ended with a flat no, or requests for development funds(!) -- apparently there is code for one card, one vendor, that runs against 'bristuff', or did at one time, but wasn't maintained through several Asterisk releases (if the code was even released to the community... IDK). Is this common, that someone codes to their chip on their card and sells it to one or two consumers, then lets it drop and never gives the code up for continued development? (It seems contrary to GNU/Linux licensing conventions, but, again, I'm not paid as a software developer. I just think they might have sold more cards with a less proprietary approach.) Anyway, can I, with confidence, state (to the $employer) that Asterisk on linux via USA 'BRI' digital lines simply isn't possible? (In that, obviously, I can't pay for development nor do beta testing, each with vague hope that it might work okay someday...) If this is the case, then I must use multiple analog lines to access PSTN, or pay premium for 'PRI' pipes (80% of which we will never need)... is that about correct? Thanks in advance for any pointers, specific RTFM suggestions, any help appreciated. If there is a different list to post this query to, I'm not (yet) aware of it. Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org Get a hold of Marcin Pyco (Former Digium Employee and extrememly smart guy. He has code/patches for zaptel to US BRIs work that include SPID as a variable in zap confs. Just have to be careful if you decide to upgrade at some point Thanks, Steve Totaro -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
It actually does contain references of all applicaitons, CLI commands, and such. Where? I saw some examples, but I've never found an organized list of commands. I'd love it. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, Jan 27, 2009 at 03:45:34PM -0500, Steve Totaro wrote: Get a hold of Marcin Pyco (Former Digium Employee and extrememly smart guy. He has code/patches for zaptel to US BRIs work that include SPID as a variable in zap confs. Could you please expand on that point? Why should such patches remain secret? And are those patches really necessary now that chan_dahdi knows that there is such a thing called BRI? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
Steve Totaro wrote: On Tue, Jan 27, 2009 at 12:49 PM, Michael Higgins li...@evolone.org wrote: Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have found. What I did find left me with the impression that USA 'BRI', uh, '2B1Q' protocol(?) is not supported by *any* hardware vendor, at all, period, nor is it tested and proved in the software... stack(?), in one related branch or another on the OS side. A couple of direct inquiries to card vendors have dead-ended with a flat no, or requests for development funds(!) -- apparently there is code for one card, one vendor, that runs against 'bristuff', or did at one time, but wasn't maintained through several Asterisk releases (if the code was even released to the community... IDK). Is this common, that someone codes to their chip on their card and sells it to one or two consumers, then lets it drop and never gives the code up for continued development? (It seems contrary to GNU/Linux licensing conventions, but, again, I'm not paid as a software developer. I just think they might have sold more cards with a less proprietary approach.) Anyway, can I, with confidence, state (to the $employer) that Asterisk on linux via USA 'BRI' digital lines simply isn't possible? (In that, obviously, I can't pay for development nor do beta testing, each with vague hope that it might work okay someday...) If this is the case, then I must use multiple analog lines to access PSTN, or pay premium for 'PRI' pipes (80% of which we will never need)... is that about correct? Thanks in advance for any pointers, specific RTFM suggestions, any help appreciated. If there is a different list to post this query to, I'm not (yet) aware of it. Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org Get a hold of Marcin Pyco (Former Digium Employee and extrememly smart guy. He has code/patches for zaptel to US BRIs work that include SPID as a variable in zap confs. Just have to be careful if you decide to upgrade at some point for what hardware ? Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
Jared Smith wrote: On Tue, 2009-01-27 at 10:13 -0700, Robert Broyles wrote: I'm still pretty new to the mailing lists myself. I don't consider myself a novice Asterisk user, but one of my biggest 'complaints' is the lack of a well documented FAQ or Manual for Asterisk. Asterisk is truly an open-source community, and that pertains to documentation as well. The quality and quantity of the documentation depends heavily on contribution from the community at large. Digium has and will continue to put resources towards Asterisk documentation, but every contribution from the community at large helps. I understand. As someone else already mentioned, Voip-Info.org is for more than just Asterisk. Perhaps if we created a single source that was just for Asterisk...where everyone could contribute towards making the documentation better. I would be very interested in helping sponsoring such a project, just so long as we have enough contributors. (Unless one is willing to buy or read O'Reilly's Book - http://www.asteriskdocs.org - which quickly will be outdated again.) Alas, you've mentioned the one thing that both makes me happy and sad at the same time. Happy that people find it useful, and that O'Reilly was kind enough to let us publish it under a Creative Commons license (and put the PDF on the web for free!)... and sad that it takes so much time and effort to keep up to date. (And just for the record, the time that the other authors and I spend on writing the O'Reilly book is our own personal time -- I'm not working on it during company time!) This was an excellent read. I'm sad to say that I was one that didn't purchase the book, but made good use of the PDF. I was hoping to win one of the books during your sessions at AstriCon this past year. Too bad. :-( I have made it a personal aim to document all my findings in a blog, so that it's at least searchable by others through Google, in hopes that others might find it useful. But if we had a REGULARLY updated FAQ/Manual ... I think that would greatly cut down on the clutter posts. If you're interested and serious about writing, join the asterisk-docs mailing list and let's try to get something started. I've been beating the documentation drum for almost seven years now, and I'd love to see the -docs mailing list come back to life. I'll be checking this out. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, 27 Jan 2009 20:43:30 +0200 Tzafrir Cohen tzafrir.co...@xorcom.com wrote: To the best of my understanding, latest Asterisk should support it through chan_dahdi . Cool. Got any links to any related informations, so I can go by more than anonymous hearsay? '-) No need for extra bristuff or whatever. But this needs some testing. Ah. But I can't replace the business telephone system with a testing system for the company that pays me to implement working stuff, can I? Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, 27 Jan 2009 11:44:12 -0700 Wilton Helm wh...@compuserve.com wrote: I'm in the same boat and have been looking at this for several months, but haven't actually jumped in, hands-on, yet. No, I don't think the situation is as dismal as you paint it, although the lack of appropriate marketing for BRI in the US has all but killed it here, making it relatively unattractive to vendors. Right. This is what I've come to understand from the archives. I have been advised of several cards that support it, but they were two or four port cards that were serious overkill for my application, and I haven't found ONE card that supports it, so maybe we could compare notes? Rather, can you tell me who claims to support what hardware, so I can confirm? '-) expectedly pricy as a result. This isn't my $$ so I don't care about initial cost. I do care that it is proven, though. I am trying to use an HFC card, and have been advised that it is possible, but will take some digging and experimenting. I have no time for digging and experimenting, unfortunately. I expect that time will be taken up entirely trying to install and configure asterisk. As near as I have been able to figure out, mISDN is the appropriate place to begin, but I haven't tried to sort it out, yet. There are a couple of other paths that might lead to a solution, but they seem to be less supported than mISDN at present (if that is even possible). It sounds like Dahdi is moving towards that, but I don't think its quite there, yet. Okay, these are various driver protocols in the kernel/user space, right? Since I haven't purchased anything, should I be analysing the rate and support for development of these different protocols and informing my decision that way? Qwest is one of the few providers I know of that prices BRI reasonably. I have one up and running on a TA and use it for my voice service. As soon as I can figure out mISDN, I plan to move it to Asterisk. (I'll keep a TA around for backup). Hmm. Yes, Qwest 'BRI' was the intent... the problem is, any solution that doesn't lease equipment from the provider conflicts to their bottom-line interest, so I don't expect any help from them. Am I wrong in that? At any rate, if you could send along some info about cards that are purported to work here, I'd *really* appreciate it. '-) Thanks, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Muted sound on a Linksys 962
Date: Tue, 27 Jan 2009 12:50:36 -0600 From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] Muted sound on a Linksys 962 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: a8d2ef23f63b4a42b264bb151c5d6...@db0005 Content-Type: text/plain; charset=us-ascii This worked for me Exten = s,1,Answer() Exten = s,n,Dial(Zap/g1/w5551212) What happens is that * doesn't go full duplex until it does a Native Bridge. The Answer Command creates a temporary bridge until the real one can take effect. I'm not sure how that would help in this case. The call is answered by the remote end and then the caller can hear the audio of the IVR menus. Or am I missing something here? -- James -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Tuesday, January 27, 2009 12:38 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Muted sound on a Linksys 962 Hi, One of our customers has an issue with the callee not being able to hear them. It seems to happen very frequently on one number in particular where there are about 3 IVR menus to dial through before getting to a live person. However, this does not happen on every call. Running tcpdump on the RTP packets, I can see that RTP is setting sent, but the values in the packet are all very close to 0xFF or 0x7F (which is 0 or -1 once you translate it using G.711). Could this be some issue with the phone muting audio because it's stuck sending DTMF? DTMFMode is rfc2833 on the Asterisk side and Auto on the phone side. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, 27 Jan 2009 12:56:46 -0500 Jon Pounder j...@inline.net wrote: I barked up the same tree you are barking for a while and just gave up - lots of you could buy this and try it, but no proven solution. That's exactly what I've come up with. Thanks for your reply. I don't see anything truly contradicting this in the other posts, so for now, it's doubtlessly a dead-end avenue... I do think it really is that bad, but I'd still like to be dissuaded of that. Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
On Tuesday 27 January 2009 15:05:57 Wilton Helm wrote: It actually does contain references of all applicaitons, CLI commands, and such. Where? I saw some examples, but I've never found an organized list of commands. I'd love it. For applications, Appendix B, and for dialplan functions, Appendix F. I put a great deal of effort to be sure that both of these appendices not only documented every single application and function, but also that every single one of them contained example code that illustrated their use and documented every single option available. If anything is missing from these two appendices (other than old deprecated stuff that we'd prefer that people forget about), I'd love to hear about it. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi Init script for Suse?
Doesn't look like SuSE is that evolved just yet. I poked at a few other init scripts in /etc/init.d, and they're all pretty much in the format of: echo -n Starting something ... command rc_status -v Some of the init scripts are downright horrific in their design because of this. I would imagine the newer SLES stuff might have cleaned things up, but Open Enterprise Server is based on SuSE 10.1/10.2. I would imagine that, whenever Novell gets around to it, OES3 will probably be based on SuSE 11 (if they even have that version -- I haven't checked yet). I think they're still ironing out kinks and trying to force stubborn holdouts (like myself) off of NetWare onto OES2. --J -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Monday, January 26, 2009 4:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dahdi Init script for Suse? On Sat, Jan 24, 2009 at 04:08:53PM -0500, Joshua Kinard wrote: Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10 box that'll work right? The one included by default only deals with debian and redhat, and the changes between the old zaptel script I have that works are far too invasive. Notably in the use of this action command that's probably redhat specific. Anything equivalent in SuSE? The Debian equivalent of action Starting FooBar foobar , if you source /etc/lsb/init-functions is log_daemon_msg Starting FooBar foobar log_end_msg $? Though it has an atvantage of making it easier to redirect output of the command. (Those functions don't seem to be part of the standard LSB init functions, sadly) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, 27 Jan 2009 12:27:04 -0600 Jerry Jones jjo...@danrj.com wrote: Instead you could always get a SIP/IAX provider. Can you please elaborate as to how this answers my question? Would getting a SIP/IAX provider be the same as getting a working 'BRI' ISDN line coming into the machine and properly handled in the chip or software stack? I know I can avoid all of this by paying a provider for services (which is what we do now). Is this proposed solution somehow different (other than cheaper) than that? I have COLO with fat bandwidth included, no PSTN lines. So, is this a viable alternative? How do I reliably fax from my server over SIP/IAX? Why would I pay for services that I have oodles of bandwidth to handle for a nominal investment? Example, you spend hours trying to get a tech support individual on the phone with any vendor, time where you learn nothing and get nowhere. Often, they don't know the answer anyway. Sometimes I call them back to tell them their business. OTOH, I can generally track down the patch, or library, or clue I need to fix anything open source and free, within an hour or two. Plus, I'm learning about the issues related and technologies underlying in the process. Anyway, I just don't see how this reply is at all helpful... say, if I intend to beta test asterisk and skype (I'm on the list, if they ever move forward), will this be possible on this kind of system? Would it be as useful, as flexible? Perhaps you were just being sarcastic -- since the lack of BRI sales (and marketing) *seems* to allow the opening to sales of these halfway services (where they wouldn't likely have much chance to sell them otherwise) -- sarcasm would make some sense. Unfortunately, in email sarcasm/sarcasm is the only way I've seen to reliably indicate communication as such. The terseness and offhandedness of the comment does inform toward this, but I don't really know... you. Nonetheless, I've 'bitten'. Let's see what you have to say. '-) Thanks for your reply. Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
On Tuesday 27 January 2009 12:35:15 Wilton Helm wrote: It seems to me that everything one may want to know would be contained on voip-info.org My own experience is that it covers a very broad spectrum (far broader than Asterisk) and in a rather terse manner. I have spent an hour or two at a time pouring over a topic there and come away little more enlightened than when I started. Most people who know enough to create useful entries there, assume too much of the reader. They assume that everyone reading the post works with Linux 40 hours a week at the command line level, and only needs a few VoIP clues to take an idea and run with it. A better assumption would be that they know how to log on. It shouldn't even be assumed that they know the difference between suand su - Yes, but voip-info.org is not meant to be the end-all be-all for users who are new to Linux. There are far better resources out there for teaching Linux newbies. Instead, voip-info.org attempts to provide the sorts of information that is useful for those already familiar with Linux and need the push up to learn this particular application. You could certainly compare and contrast the documentation for other large daemon applications, such as MySQL, PostgreSQL, or BIND, to see what each large application considers worthy of its documentation, versus documentation for bringing a Linux newbie up to speed. Note that I specifically chose applications which are primarily daemons and do not contain a GUI, as those are most comparable to Asterisk. I realize that this is challenging because different distros do things different ways. That is another topic of its own, but is also one of the banes of Linux that is hurting its usability considerably. Actually, the diversity of the Linux ecosystem is considered to be one of its strengths. The friendly competition between projects ensures that each continues to strive for the best. Any project which stagnates quickly falls by the wayside. It's certainly instructive that the continuing advances in open source browser technology was what spurred Microsoft to once again invest time into its own browser (whose development had stagnated after the demise of its previous main competitor, Netscape). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
** I understand. As someone else already mentioned, Voip-Info.org is for more than just Asterisk. Perhaps if we created a single source that was just for Asterisk...where everyone could contribute towards making the documentation better. I would be very interested in helping sponsoring such a project, just so long as we have enough contributors. ** We have some documentation and I can contribute that. Also we can provide the physical resources (Domain, Web hosting, bandwidth, storage, database etc). Ofcourse need a team with designated responsibilities. -Jai Rangi www.didforsale.com On Tue, Jan 27, 2009 at 1:16 PM, Robert Broyles rob...@poornam.com wrote: Jared Smith wrote: On Tue, 2009-01-27 at 10:13 -0700, Robert Broyles wrote: I'm still pretty new to the mailing lists myself. I don't consider myself a novice Asterisk user, but one of my biggest 'complaints' is the lack of a well documented FAQ or Manual for Asterisk. Asterisk is truly an open-source community, and that pertains to documentation as well. The quality and quantity of the documentation depends heavily on contribution from the community at large. Digium has and will continue to put resources towards Asterisk documentation, but every contribution from the community at large helps. I understand. As someone else already mentioned, Voip-Info.org is for more than just Asterisk. Perhaps if we created a single source that was just for Asterisk...where everyone could contribute towards making the documentation better. I would be very interested in helping sponsoring such a project, just so long as we have enough contributors. (Unless one is willing to buy or read O'Reilly's Book -http://www.asteriskdocs.org - which quickly will be outdated again.) Alas, you've mentioned the one thing that both makes me happy and sad at the same time. Happy that people find it useful, and that O'Reilly was kind enough to let us publish it under a Creative Commons license (and put the PDF on the web for free!)... and sad that it takes so much time and effort to keep up to date. (And just for the record, the time that the other authors and I spend on writing the O'Reilly book is our own personal time -- I'm not working on it during company time!) This was an excellent read. I'm sad to say that I was one that didn't purchase the book, but made good use of the PDF. I was hoping to win one of the books during your sessions at AstriCon this past year. Too bad. :-( I have made it a personal aim to document all my findings in a blog, so that it's at least searchable by others through Google, in hopes that others might find it useful. But if we had a REGULARLY updated FAQ/Manual ... I think that would greatly cut down on the clutter posts. If you're interested and serious about writing, join the asterisk-docs mailing list and let's try to get something started. I've been beating the documentation drum for almost seven years now, and I'd love to see the -docs mailing list come back to life. I'll be checking this out. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
Don't over think this, guys. Again, the point of having a WIKI is to allow for customization. A landing page for Asterisk documentation within voip-info.org is all you need, not a whole new source of documentation. Jai Rangi wrote: ** I understand. As someone else already mentioned, Voip-Info.org is for more than just Asterisk. Perhaps if we created a single source that was just for Asterisk...where everyone could contribute towards making the documentation better. I would be very interested in helping sponsoring such a project, just so long as we have enough contributors. ** We have some documentation and I can contribute that. Also we can provide the physical resources (Domain, Web hosting, bandwidth, storage, database etc). Ofcourse need a team with designated responsibilities. -Jai Rangi www.didforsale.com http://www.didforsale.com On Tue, Jan 27, 2009 at 1:16 PM, Robert Broyles rob...@poornam.com mailto:rob...@poornam.com wrote: Jared Smith wrote: On Tue, 2009-01-27 at 10:13 -0700, Robert Broyles wrote: I'm still pretty new to the mailing lists myself. I don't consider myself a novice Asterisk user, but one of my biggest 'complaints' is the lack of a well documented FAQ or Manual for Asterisk. Asterisk is truly an open-source community, and that pertains to documentation as well. The quality and quantity of the documentation depends heavily on contribution from the community at large. Digium has and will continue to put resources towards Asterisk documentation, but every contribution from the community at large helps. I understand. As someone else already mentioned, Voip-Info.org is for more than just Asterisk. Perhaps if we created a single source that was just for Asterisk...where everyone could contribute towards making the documentation better. I would be very interested in helping sponsoring such a project, just so long as we have enough contributors. (Unless one is willing to buy or read O'Reilly's Book - http://www.asteriskdocs.org - which quickly will be outdated again.) Alas, you've mentioned the one thing that both makes me happy and sad at the same time. Happy that people find it useful, and that O'Reilly was kind enough to let us publish it under a Creative Commons license (and put the PDF on the web for free!)... and sad that it takes so much time and effort to keep up to date. (And just for the record, the time that the other authors and I spend on writing the O'Reilly book is our own personal time -- I'm not working on it during company time!) This was an excellent read. I'm sad to say that I was one that didn't purchase the book, but made good use of the PDF. I was hoping to win one of the books during your sessions at AstriCon this past year. Too bad. :-( I have made it a personal aim to document all my findings in a blog, so that it's at least searchable by others through Google, in hopes that others might find it useful. But if we had a REGULARLY updated FAQ/Manual ... I think that would greatly cut down on the clutter posts. If you're interested and serious about writing, join the asterisk-docs mailing list and let's try to get something started. I've been beating the documentation drum for almost seven years now, and I'd love to see the -docs mailing list come back to life. I'll be checking this out. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, Jan 27, 2009 at 4:07 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Jan 27, 2009 at 03:45:34PM -0500, Steve Totaro wrote: Get a hold of Marcin Pyco (Former Digium Employee and extrememly smart guy. He has code/patches for zaptel to US BRIs work that include SPID as a variable in zap confs. Could you please expand on that point? Why should such patches remain secret? And are those patches really necessary now that chan_dahdi knows that there is such a thing called BRI? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir BRISTUFF never worked properly with US BRI. Chan_dahdi MIGHT work but I see a steady stream of Dahdi compaints, segfaults, not seeing things. Sound like Alpha code to me. I will not be installing anything called DHADI on a production system for another year at least. Besides that, can you specify in dahdi.conf or whatever what the SPIDS are? Does it give you an option for US vs other BRI? If not, it won't work reliably. If you don't mind missing inbound calls at random intervals, then it should work fine. From what I understand it Marcin's code works on all card's that BRIStuff would work on, but does not because US BRI is a bit different than other BRI around the world. About keeping secrets, which he doesn't, he has said that he had these patches for a long time now. At any rate, it is his code to whatever he wants. If he wants to charge for it, do consulting, or keep it a secret, it is not your business nor do you have any say what he wants to do with his code. I think you just want your hands on the code since you work for a competitor but I may be wrong -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialstatus through a call file
Hello, Is it possible to retrieve the DIALSTATUS variable when placing call through a call file. This variable is set when using the Dial() application from the dialplan, but I am using a call file for my current application and need to get the dialstatus. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On 1/27/09, Steve Totaro stot...@totarotechnologies.com wrote: On Tue, Jan 27, 2009 at 4:07 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Jan 27, 2009 at 03:45:34PM -0500, Steve Totaro wrote: Get a hold of Marcin Pyco (Former Digium Employee and extrememly smart guy. He has code/patches for zaptel to US BRIs work that include SPID as a variable in zap confs. Could you please expand on that point? Why should such patches remain secret? And are those patches really necessary now that chan_dahdi knows that there is such a thing called BRI? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir BRISTUFF never worked properly with US BRI. Chan_dahdi MIGHT work but I see a steady stream of Dahdi compaints, segfaults, not seeing things. Sound like Alpha code to me. I will not be installing anything called DHADI on a production system for another year at least. Besides that, can you specify in dahdi.conf or whatever what the SPIDS are? Does it give you an option for US vs other BRI? If not, it won't work reliably. If you don't mind missing inbound calls at random intervals, then it should work fine. From what I understand it Marcin's code works on all card's that BRIStuff would work on, but does not because US BRI is a bit different than other BRI around the world. About keeping secrets, which he doesn't, he has said that he had these patches for a long time now. At any rate, it is his code to whatever he wants. If he wants to charge for it, do consulting, or keep it a secret, it is not your business nor do you have any say what he wants to do with his code. I think you just want your hands on the code since you work for a competitor but I may be wrong -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Tue, 27 Jan 2009 09:49:41 -0800, Michael Higgins wrote: snip It seems to me that there a lot of it ought to work or could be made to to work associated with implementing US BRI into Asterisk. That being the case what's called for is someone to try it, just to prove the point. Over the past year or two at least a couple of hardware vendors have indicated a willingness to help with loan of interface hardware (Xorcom Pika) Perhaps a small group could take on the task of giving it a try? It's a pain to have a BRI provisioned from an ILEC just for sake of an experiment. I may know someone who has functional BRI already. They're a TV station using it to convey production audio. If they agreed would it be possible/practical to use their BRI for a short while just to have a quick test? Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Module res_odbc is not loading
Hi, I have remove the comment defor res_odbc.so and res_config_odbc.so in my modules.conf, but the module is still not loading when I do: module show like odbc I have o module returned anybody knows why? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
Rather, can you tell me who claims to support what hardware, so I can confirm? I don't have any notes on what I did. It was a bunch of Google searches. I seem to recall that Digium themselves made a two port BRI that would work. Eicon has some well respected products that I am pretty sure are supported for NT1 (They are based in Canada). Zyxel, I think does, also. I didn't go that way because most were two or more ports and I only needed one and couldn't justify the $200 to $400 cost. I have no time for digging and experimenting, unfortunately. I expect that time will be taken up entirely trying to install and configure asterisk. That translates fairly straightforwardly into, you have no time for Asterisk or Linux, its underlying operating system. I can't imagine going down this path in any form without setting aside weeks of time for experimentation. I've invested at least two person weeks to get Linux up and running and a small demo Asterisk install with no trunks and two SIP phones. I think the payoff will be worth it, but the learning curve is steep. If you are already capable of installing Linux from source code in your sleep, you mileage may very significantly. Okay, these are various driver protocols in the kernel/user space, right? Yes, Dahdi is tightly linked to Asterisk, the others are more loosely coupled. Since I haven't purchased anything, should I be analysing the rate and support for development of these different protocols and informing my decision that way? Based on the related posts today, it sounds like you should be looking at what dahdi can do and what drivers the board makers provide. The other options are possible paths to a solution but no where close to turn-key. any solution that doesn't lease equipment from the provider conflicts to their bottom-line interest, so I don't expect any help from them. Am I wrong in that? I've been getting ISDN from QWEST for about eight years now at a combination of four locations. The company has made a huge amount of progress in operations over that time. The last two installs went off pretty much without a hitch. The sales people tend to be weak on ISDN, and the I R people are variable. My best result has been with the guy in the CO that works ISDN circuits from the inside. In our office, he's sharp and quite willing to give out his phone number to anyone who doesn't come across as an idiot for follow-up questions. I wouldn't expect them to tell me how to configure Asterisk, but if they have a problem, or sales didn't write up the configuration correctly, he's been able to answer my question and change the CO programming on the spot for me. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
Thanks for engaging with me on this. I picked up the book and I see what you mean about Appendix B. I had under-appreciated it probably because of a paradigm shift I need to make. I think you meant Appendix E rather than F for dialplan. I still am not quite on the same page with you, though. There are a lot of commands that aren't function calls that go into various config files. The most basic and obvious one is exten There must be a hundred of these and I don't know where they are listed with all acceptable parameters and ranges and what they do and why. There are examples to get one started, but I don't think I can put my hands on even a definitive definition of exten. Am I making any sense? Maybe these are called variables or something. I'm scared to even look for the setting for an NT1 ISDN BRI, which is the mountain I have to climb next. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailinglists
There are far better resources out there for teaching Linux newbies. Instead, voip-info.org attempts to provide the sorts of information that is useful for those already familiar with Linux I can appreciate that. And I can appreciate being at the other end of the pipe, as I like to gloss over all obvious details when I have to write up something. I'm not suggesting that VoIP should become a Linux tutorial, but that, where possible, every line that must be typed to get to a desired end be explicitly included rather than assuming that a one sentence comment will empower the reader to type in a whole page of bash stuff. You could certainly compare and contrast the documentation for other large daemon applications I would concur with your thoughts here. The terse style is endemic to everything Linux. I bought a commercial Linux app recently and it didn't even have a single word about installation. Turns out there was no configuration, so you could just drop it into a directory and make a shortcut icon if you were using Gnome or something. But at least a line stating that would have saved the author an E-Mail exchange..Oh yes, and then there was the library it needed that wasn't in the distro. It would have been useful if the Readme mentioned that. Anyway, you get the point. It's certainly instructive that the continuing advances in open source browser technology was what spurred Microsoft to once again invest time into its own browser True, but I can open IE and use it and then open Firefox and intuitively know what to do. It doesn't say you can use Z or -M or --Query all to do the same thing, (possibly with identical parameters, or possibly with parameters formatted differently) but -M only works on Fedora and --Query only works on Suse (contrived by true to life examples). It is this sort of thing that makes the Linux learning curve steep and makes it challenging to provide detailed instructions for something like installing a package. Based on a number of conversations over the last year or two, I have become convinced that those for whom these command automatically flow off of their fingertips are mostly clueless as to how unintuitive some of this stuff is when first encountered. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
I'm with you on this. A VoIP trunking solution is never going to equal a LEC PSTN solution. It may be adequate for some purposes, but I'm not about to dump my BRI for a pair of IP numbers. The trade-offs aren't worth the small cost savings for me. Just the packetized delays (not to mention internet latency) are going to degrade things somewhat. Your example about FAX is a good one. Yes, there are work-arounds, but when you have customers to keep happy, meeting their expectations in an intuitive manner is critical. OTOH, I certainly like the idea of an open source PABX doing my internal routing. I've done some stuff with commercial PABXs that few people would attempt. I've also failed to do things because the PABX OS didn't support what I wanted to do. That's one reason I once wrote my own. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 dahdi only?
Wilton Helm wrote: Thanks for engaging with me on this. I picked up the book and I see what you mean about Appendix B. I had under-appreciated it probably because of a paradigm shift I need to make. I think you meant Appendix E rather than F for dialplan. I still am not quite on the same page with you, though. There are a lot of commands that aren't function calls that go into various config files. The most basic and obvious one is exten http://www.voip-info.org/wiki/view/Asterisk+howto+dial+plan There must be a hundred of these and I don't know where they are listed with all acceptable parameters and ranges and what they do and why. There are examples to get one started, but I don't think I can put my hands on even a definitive definition of exten. Am I making any sense? Maybe these are called variables or something. all the parameters as you call them are the dialplan commands and functions http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf exten = 123,1,Answer exten = 123,2,Playback(tt-weasels) exten = 123,3,Voicemail(44) exten = 123,4,Hangup if the number 123 is called, step1: answer the call step2: run the application called Playback, using the tt-weasels sound file (this plays tt-weasels to whoever called 123) step3: run the voicemail application for mailbox 44 step4: hangup the call iow, 123 is the called extension, and steps 1,2,3,4 are then executed Julian I'm scared to even look for the setting for an NT1 ISDN BRI, which is the mountain I have to climb next. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
