Re: [asterisk-users] TE110P Wildcard does not work with Ubuntu 12.04 server
Michel Verbraak michel at verbraak.org writes: Op 22-08-12 12:09, Shitian Long schreef: I am trying to setup TE110P wildcard on a PBX running ubuntu 12.04 server edition. I followed the procedure from http://docs.digium.com/misc/ADL_quickstart.pdf step by step. During the process of installing dahdi-linux-complete I got following warnings: root at ubuntu:/usr/local/src/dahdi-linux-complete-2.6.1+2.6.1# make perl: warning: Setting locale failed. perl: warning: Please check that your locale settings: LANGUAGE = en_US:en, LC_ALL = (unset), LC_CTYPE = UTF-8, LANG = en_US.UTF-8 are supported and installed on your system. perl: warning: Falling back to the standard locale (C). Frist of, I am wondering if this error matters? Second question, after installation process complete, and reboot the machine I got the following error, when machine boot up: Loading DAHDI hardware modules: wcte11xp: error I think the TE110P card is no properly loaded. I try to confirm my thought by using root at ubuntu:~# dahdi_tool There is no interface listed on the table. I am wondering if anyone got idea about this issue. Thanks. longst -- _ Having the same problems: /# dmesg -c /dev/null /# /etc/init.d/dahdi stop Unloading DAHDI hardware modules: done /# modprobe wte11xp FATAL: Module wte11xp not found. /# dmesg [ 2775.316252] dahdi: Telephony Interface Unloaded /# lspci | grep Dig /# however: # make config install -D dahdi.init /etc/init.d/dahdi /usr/sbin/update-rc.d dahdi defaults 15 30 System start/stop links for /etc/init.d/dahdi already exist. DAHDI has been configured. List of detected DAHDI devices: pci::03:02.0 wcte11xp-e159:0001 Digium Wildcard TE110P T1/E1 Board run 'dahdi_genconf modules' to load support for only the DAHDI hardware installed in this system. By default support for all DAHDI hardware is loaded at DAHDI start. any ideas? doesnt look like my system recognized the card at all. no lights on the card. card was working perfectly. i simply moved it from one box (gentoo) to another (ubuntu 12.04) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE110P Wildcard does not work with Ubuntu 12.04 server
Shaun Ruffell sruffell at digium.com writes: Best guess based on what you said that the driver really isn't installed for the current kernel. Cheers, Shaun thanks much! this lead me to find this: http://totalticketsystem.com/blog/technical-articles/how-to-install-asterisk-on-ubuntu-from-scratch/ which i followed and things are working now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue called by agi doesn't re-enter the script
http://www.voip-info.org/wiki/view/Asterisk+cmd+DeadAGI read the part at the bottom about ignoring sighup, if you're using a later version i think there is a option like agisighup that you can use in the dialplan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manual Call Transfer // Perl // Asterisk::AGI // MySQL
Hello Everyone!
I've hit a bit of a roadblock and I am hoping that someone might point
me in the right direction.
I am using Asterisk 1.2.4 - I do not have the option of updating it,
please do not waste your time telling me to =)
I am using PERL AGI scripts to maintain an active calls count field
for each phone in a mysql database table, for example (not actual
code, just trying to illustrate)
$SIG{HUP} = 'IGNORE';
mysql_update_call_count($user_id, ($count +1) );
$dialret = $agi-exec('Dial', $dialstring);
mysql_update_call_count($user_id, ($count -1) );
This works great, except when doing assisted transfers (or any
transfer for that matter).
We have Polycom IP550 Phones which can do the transfer with a button,
As an example of this process and the problem, and assuming these are
all internal phones dialing extensions...
phone A dials phone B
phone B presses transfer to transfer phone A to phone C
phone B hangs up
Because the Dial command in the AGI script executed when phone A
called phone B is still running the active call count remains at 1 for
phone B until the call between A and C ends (at which point they all
zero out).
I also tried using atxfer to resolve this problem and got a different behavior
phone A dials phone B
phone B presses *2 then phone C's extension to transfer phone A to phone C
phone B hangs up
an active call count remains at 1 for B and C but A drops to 0 count.
Might be worth mentioning the possibility that phone B is already on
the line when the call from phone A comes in.
I thought one possible solution might be creating an [applicationmap]
that essentially handles the assisted transfer manually. I've done a
great deal of reading on this matter and aside from the fact that I'm
still a bit fogy as to how i would even do that,.. it seems that there
is still no way for me to determine who is being transferred when the
second channel is opened (new uniqueid / agi script execution).
Is there perhaps something I am missing which would help resolve this?
I hope that I've explained my problem clearly. I have only been
tinkering with asterisk for about a week so I apologize if I'm not
using the appropriate vernacular.
Thank you!
-Ted
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[asterisk-users] Evaluating Asterisk
I am thinking of moving from a traditional PBX to an asterisk box. Many of my leadership group are skeptical of asterisk. So I was hoping to find a call center that is currently using this technology that would not mind spending some time on a conference call to address some concerns that my team has. Thanks Ted Foote Allied Business Services, Inc. 616-741-0437 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for Motel Instalation.
audio code MP124 - Original Message - From: Arturo Ochoa To: asterisk-users@lists.digium.com Subject: [asterisk-users] Recommendations for Motel Instalation. Date: Fri, 20 Jun 2008 15:42:28 -0600 Dear List, I have a customer who owns a little Motel, and he wants to upgrade to a Asterisk PBX. There is one analog phone per room (aprox 80), and the cable is CAT 3. Any recommendations on what card to use? TDM24XXP vs Channel Bank? Regards, Ing. Arturo Ochoa N Electrosystems S RL Tel. (656)-6230794 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- See Exclusive Videos: 10th Annual Young Hollywood Awards http://www.hollywoodlife.net/younghollywoodawards2008/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Deadlock problem with agents, queues and PRI (stop accepting incoming calls in PRI line)
Hi everyone, We have a Asterisk-based call center deployment with around 40 SIP users, attending incoming calls from two PRI lines (2xE1) using agents and queues. The problem is that Asterisk stops accepting new incoming calls to the PRI lines without reason, although there should be free channels to make room for new incoming calls, but Asterisk thinks these channels are being used. SIP calls can be placed without problemsbetween internal users. PRI lines shouldn't be the origin of the problem, as an old legacy PBX works perfectly with the same lines, so the problem seems to be related with agents or queues. After the crash, performing an zap show channels shows that all channels are busy, and calls seems that have been queued for a long time in different queues (and they are not really there - users don't wait 50 minutes to be attended listening to the music on hold). There is no other services running on the server, CDR is being stored to disk and we are not using any kind of AGI's or reporting tools. Currently the only solution is to reboot the machine, as rebooting Asterisk is not enough. Using any command on the CLI results in no output at all. The crash is not easily reproduceable, as it doesn't follow a clear pattern. Asterisk just seem to get blocked when it manages around 30-40 calls in the queues. During last week, we had 2-3 crashed each day. Based on users lists mails, it seems that other users have had a similar problem within the same scenario, at least with 1.2.x. More precisely, we have observed the same problem in bug ID 0006147, but it has been closed without a clear answer. Hardware and software specs: Platform: Suse Linux Enterprise Server 10 Machine: IBM xSeries 226, 1 GB RAM, Intel CPU PRI card: Digium TE212 with echo cancellation module Asterisk version: 1.2.18 Follows a list of the most relevant messages before and after the crash: DEBUG[28519] chan_sip.c: Stopping retransmission on 'NzNmZWM0ZDc0OTYyNWI5YWM2ZTBhZjY3NDM4N2RjNmQ.' of Response 12: Match Found (lots of messages like that) DEBUG[28511] chan_zap.c: Ring requested on channel 0/13 already in use or previously requested on span 1. Attempting to renegotiating channel. DEBUG[28511] chan_zap.c: Found empty available channel 0/9 DEBUG[29939] app_dial.c: Exiting with DIALSTATUS=CONGESTION. I would very appreciate any help on this. I can provide backtrace if needed. Best regards, Ted. -- Ted Brown tedbrown75 REMOVE_THIS gmail.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oddity with TDM400P / Asterisk setup
Hey, in case anybody has this problem. I went out to the location of the Asterisk box and found two problems. The default gateway, a sonicwall, was sharing an ip address with a wifi point (doh) and a patch cable plugged into his switch on both ends (doh again). --Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com On Aug 30, 2006, at 5:39 PM, Ted Wallingford wrote:Hi List,I am working with an Asterisk server running on Fedora Core 4. It has two TDM400P cards installed. There are 6 trunk ports and 2 (unused) analog line ports. There are 5 Polycom SoundPoint 501 SIP phones connected to the server, and a Linksys 24-port powered switch connecting everything. The * version running is 1.2.7.1. All of the ports on the switch with voice devices, including the server, have a service class of 5, while non-voice devices are connected to other ports that have a service class of best effort.The problem, which began this morning, is very elusive. Calls-in-progress from zap-to-sip or sip-to-zap or sip-to-Asterisk will drop at odd times during the call, anywhere from 2 minutes to 15 minutes into the call. At the same time the call drops, my SSH session to the server will hang. After 10 to 15 seconds, the output and input from ssh session appears on my terminal and I am able to resume working in the shell. Zap-to-Asterisk doens't seem to cause the problem. Only when I dial through to a SIP device does it seem to hang.Top reveals nothing out the ordinary, utilization wise, the disk has plenty of free space, and the arp cache doesn't ever indicate a duplicate IP address with the server's NIC, which I thought might have been the problem. I also attempted to move the server to another port on the switch. No improvement. Anybody have a problem like this?--Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com --Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com On Jul 13, 2006, at 3:02 PM, Warren (mailing lists) wrote:Ronald Wiplinger wrote: Kevin P. Fleming wrote: Can we please keep the discussions about carriers, money, jobs, work,etc. off of this list? This is not the place to discuss yourexperiences with _any_ company, it's a place to talk about Asteriskand using Asterisk.Please move flamewars and similar discussions to some other forum. I agree with you!Which place is in your opinion the right place?As long there is no other place, such messages will always pop up. How about the Asterisk-biz list?W___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] oddity with TDM400P / Asterisk setup
Hi List,I am working with an Asterisk server running on Fedora Core 4. It has two TDM400P cards installed. There are 6 trunk ports and 2 (unused) analog line ports. There are 5 Polycom SoundPoint 501 SIP phones connected to the server, and a Linksys 24-port powered switch connecting everything. The * version running is 1.2.7.1. All of the ports on the switch with voice devices, including the server, have a service class of 5, while non-voice devices are connected to other ports that have a service class of best effort.The problem, which began this morning, is very elusive. Calls-in-progress from zap-to-sip or sip-to-zap or sip-to-Asterisk will drop at odd times during the call, anywhere from 2 minutes to 15 minutes into the call. At the same time the call drops, my SSH session to the server will hang. After 10 to 15 seconds, the output and input from ssh session appears on my terminal and I am able to resume working in the shell. Zap-to-Asterisk doens't seem to cause the problem. Only when I dial through to a SIP device does it seem to hang.Top reveals nothing out the ordinary, utilization wise, the disk has plenty of free space, and the arp cache doesn't ever indicate a duplicate IP address with the server's NIC, which I thought might have been the problem. I also attempted to move the server to another port on the switch. No improvement. Anybody have a problem like this?--Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com --Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com On Jul 13, 2006, at 3:02 PM, Warren (mailing lists) wrote:Ronald Wiplinger wrote: Kevin P. Fleming wrote: Can we please keep the discussions about carriers, money, jobs, work,etc. off of this list? This is not the place to discuss yourexperiences with _any_ company, it's a place to talk about Asteriskand using Asterisk.Please move flamewars and similar discussions to some other forum. I agree with you!Which place is in your opinion the right place?As long there is no other place, such messages will always pop up. How about the Asterisk-biz list?W___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How many users on an asterisk box behind a dsl can you have
I have been trying to read up and understand Asterisk. I have a small office of 25 people growing to 50 and have a dedicated DSL for Asterisk and another DSL for computer use and was wondering using gsm primarily how many users I could put on the asterisk box on a single dsl. Average calls is probably going to be 25-35 at any given time. Any help or suggestions would be appreciated.Ted See the all-new, redesigned Yahoo.com. Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No ringing
We are having problems when people call us the caller hears no rings on the phone. We hear the phones ring in the office, but the caller thinks they have a dead call and just keeps calling back. It doesn't happen all the time but is consistent from the same numbers. If I call from one service provider (verizon), I hear the ring, but if I call from another (Cingular) it doesn't. Any ideas what could be going on? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Rugged VoIP phones for use with asterisk
you should use an analog made for ships with a ATA. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Monday, October 31, 2005 7:13 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Rugged VoIP phones for use with asterisk Is there anyone who knows where to find rugged IP phones? Rugged in this case means that need to be installed on a ship's deck, so it must be water resistant, anyway compliant with IP 65 specification (protected against dust and jets of water). Regards ++ | Francesco Pellegrini | | Frame Srl | | Via Antonio Cantore 62/10 | | 16149 Genova | | Tel. +39 010 8680570| | Fax. +39 010 6591413 | | Cell. +348 2237798 | ++ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.6/151 - Release Date: 10/28/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail/Record sending no RTP packets (CNG) back to caller when recording messages
In August and September of last year, there was some discussion of changing the Voicemail and Record applications to send back CNG RTP packets during recording to prevent inbound calls from dropping when they assumed a disconnect after 30 seconds of no RTP frames. Was there any resolution on this issue? I'm bringing up a new SIP provider, and am seeing the same behavior that was reported a year ago with BroadVoice, although on my system the disconnect happens after 10 seconds. -- Ted Cabeen Sr. Systems/Network Administrator Impulse Internet Services ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems dialing out TDM400P - latest CVS-HEAD
No dialpad key buttons seem to work dialing out. Latest CVS head. Sep 12 09:41:55 DEBUG[1941] chan_zap.c: Exception on 15, channel 1 Sep 12 09:41:55 DEBUG[1941] chan_zap.c: Got event On hook(1) on channel 1 (index 0) Sep 12 09:41:55 DEBUG[1941] chan_zap.c: disabled echo cancellation on channel 1 Sep 12 09:41:55 DEBUG[1941] chan_zap.c: waitfordigit returned 0... Sep 12 09:41:55 DEBUG[1941] chan_zap.c: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1 Sep 12 09:41:55 DEBUG[1941] chan_zap.c: disabled echo cancellation on channel 1 Sep 12 09:41:55 DEBUG[1941] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Sep 12 09:41:55 DEBUG[1941] chan_zap.c: Updated conferencing on 1, with 0 conference users Sep 12 09:41:55 VERBOSE[1941] logger.c: -- Hungup 'Zap/1-1' -- Ted Serreyn Phone:262-432-0260 Fax:262-432-0232 Serreyn Network Services, LLChttp://www.serreyn.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT Asterisk Peering
This is not a problem. I do this and a bit more. The IAX protocol helps quite a bit to go thru the NAT. -- Ted Serreyn Phone:262-432-0260 Fax:262-432-0232 Serreyn Network Services, LLChttp://www.serreyn.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Adams Sent: Thursday, July 14, 2005 10:13 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] NAT Asterisk Peering I was wondering and trying to figure out how the following idea would work and what might be needed to implement it, can anyone give any suggestions? What I am wanting to do is to setup an asterisk server, primarily so I can get voicemail, call files, and the idea of cheaper long distance by using the internet. What seems like the place that things could get tricky is that I am desiring to place the asterisk server behind a firewall that performs NAT operations on the packets. Any of the SIP phones or ATA devices that would connect directly to my box would also be behind the NAT location. The only passthru to the outside is in the idea of peering with other asterisk boxes, I do have one in mind, that are on the outside of my own NAT box. I am desiring to setup the peering with SIP rather than IAX at least on this main connection. Is there any examples of the VOIP-info website or other places that might be of use in this idea? Or does anyone have suggestions or knowledge of this working for them? -- Dan Adams - [EMAIL PROTECTED] http://www.infochi.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom configs?
Ditto, I only have a couple of the polycomm phones spent the better part of 1 day figuring out how to get them configured properly. -- Ted Serreyn Phone:262-432-0260 Fax:262-432-0232 Serreyn Network Services, LLChttp://www.serreyn.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Friday, July 15, 2005 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom configs? Chris, as I look over my stack of unopened Polycom 501s, I think to myself that I would enjoy seeing your provisioning script if you wouldn't mind sharing it. Chris Mason (Lists) wrote: Michael Graves wrote: I have a number of Polycom phones to setup with my * server. For my initial few phones I hand wrote configs. Does anyone here who uses Polycom phones have some form of management utility for automating their setup? I wrote myself a very simple script that makes provisioning the phone a one line command. Let me know if you would like it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two Zap Problems with 1.0.2 that appeared at the same time: choppyness and squealing
I've got an * system that is having some real problems with 1.0.2. The biggest problem is that calls going through my T100P get choppy for about 10 seconds every 1 or 2 minutes. Asterisk is running on a debian stable system with current packages. The T100P is plugged into a Adit Channelbank with 8 POTS lines hooked up to the Channelbank. I've watched the vritual memory and CPU status on the * box during the call and the system is totally idle. Looking at the verbose logging in *, there isn't any obvious activity on the console that corresponds with the choppyness. It sounds like this is the echo canceller having problems, but regular phones plugged into the lines sound fine. In early November, this system was running fine on 1.0, but even downgrading back to 1.0 now doesn't fix the problem. Any thoughts? The other problem is related to our legacy PBX system, which we route through * when the * is being tested. To do so, we run the 7 PBX lines into a FXS card on the Channelbank and configure * to bridge the PBX lines to the POTS lines as necessary. The problem is that when the employee talking through * on the PBX hangs up the line, the caller coming in through the POTS line hears a very loud squeal for a second until * hangs up the line. I've run the channelbank with a loopbacked T1 replicating the same problem, and it doesn't happen, so * is definitely involved somehow. Is there a setting to determine how asterisk detects hangups on FXS Zap channels? -- Ted Cabeen http://www.pobox.com/~secabeen[EMAIL PROTECTED] Check Website or Keyserver for PGP/GPG Key BA0349D2 [EMAIL PROTECTED] I have taken all knowledge to be my province. -F. Bacon [EMAIL PROTECTED] Human kind cannot bear very much reality.-T.S.Eliot [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GUI for Asterisk Configuration
Fellow Asterisk users, I want to make configuration of Asterisk less of a chore. I noticed that there are quite a few GUI based configuration tools posted on the wiki. I am curious if there is a consensus on which is the best, most stable, easiest to install etc. Thanks, Von Belser ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mac OS X installer for Asterisk
Benjamin, Is this package intended to mirror the directory structure of the linux builds? If so, I may have an issue: While /var/lib/asterisk is properly in place after running the installer, /usr/sbin/asterisk is not. I'm running on OS X 10.3.4 and downloaded the package on Sunday afternoon, if that's any help. Did I miss something? Thanks, Ted Wallingford -Original Message- From: Sunrise Ltd [mailto:[EMAIL PROTECTED] Sent: Saturday, July 17, 2004 2:09 PM To: astusr Subject: [Asterisk-Users] Mac OS X installer for Asterisk Hi I have created a Mac OS X installer package for installing Asterisk on OSX ver 10.2 and 10.3 Anyone who'd like to give this a try, please download the installer package from here ... http://www.astmasters.net/stuff/Asterisk.pkg.tgz to install Asterisk on OSX just double click the package file. please send any feedback to benjamin (at) sunrise (dash) tel (dot) com NOTE: this is a fairly old build but it's rock solid. We have run it on OSX Server 10.2.8 since October last year and it's been going like a Swiss clockwork. Rich Murphey has promised to fix the Makefile for the most recent CVS so it will build on OSX again. Once this is done, we'll make another installer package for the new version. Also, I am still working on extending the install package so that users can choose whether or not they want to install the sources. Anybody interested in this, please bare with me a few more days. regards benjamin -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __ Do You Yahoo!? http://bb.yahoo.co.jp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Wallingford, Ted.vcf Description: Binary data
[Asterisk-Users] Mac OS X installer: missing files fix
I've paraphrased the OS X installer developer's comments: there's a bug in Installer that is preventing the archive from working right. Below is the fix for the problem. First (obviously) run the installer. Since the executables are in the archive.pax.gz file in the installer package, first do a show package contents on the package file, then unstuff the enclosed archive.pax.gz file to the desktop... Then open up a shell, CD to the desktop, and run the following: cat Archive.pax | pax -r sudo cp -R usr/* /usr Anyway, hope this helps the Mac folks on the list. Thanks Benjamin for your efforts in this area. -Original Message- From: Wallingford, Ted [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 11:28 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Mac OS X installer for Asterisk Benjamin, Is this package intended to mirror the directory structure of the linux builds? If so, I may have an issue: While /var/lib/asterisk is properly in place after running the installer, /usr/sbin/asterisk is not. I'm running on OS X 10.3.4 and downloaded the package on Sunday afternoon, if that's any help. Did I miss something? Thanks, Ted Wallingford -Original Message- From: Sunrise Ltd [mailto:[EMAIL PROTECTED] Sent: Saturday, July 17, 2004 2:09 PM To: astusr Subject: [Asterisk-Users] Mac OS X installer for Asterisk Hi I have created a Mac OS X installer package for installing Asterisk on OSX ver 10.2 and 10.3 Anyone who'd like to give this a try, please download the installer package from here ... http://www.astmasters.net/stuff/Asterisk.pkg.tgz to install Asterisk on OSX just double click the package file. please send any feedback to benjamin (at) sunrise (dash) tel (dot) com NOTE: this is a fairly old build but it's rock solid. We have run it on OSX Server 10.2.8 since October last year and it's been going like a Swiss clockwork. Rich Murphey has promised to fix the Makefile for the most recent CVS so it will build on OSX again. Once this is done, we'll make another installer package for the new version. Also, I am still working on extending the install package so that users can choose whether or not they want to install the sources. Anybody interested in this, please bare with me a few more days. regards benjamin -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __ Do You Yahoo!? http://bb.yahoo.co.jp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Wallingford, Ted.vcf Description: Binary data
RE: [Asterisk-Users] Mac OS X installer: missing files fix
Benjamin, Did you try using the post-install script to do the cat and copy operations in lieu of the Mac package installer doing it? The package installer will run any shell command you put in the post-install script? I once used it to overcome a similar problem with PostgreSQL on Panther. That might do the trick. I am going to try out your * AppleScripts as soon as possible! Thanks, Ted -Original Message- From: Sunrise Ltd [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 3:39 PM To: astusr Subject: Re: [Asterisk-Users] Mac OS X installer: missing files fix Wallingford, Ted wrote: I've paraphrased the OS X installer developer's comments: there's a bug in Installer that is preventing the archive from working right. Below is the fix for the problem. Apple may have a reputation for attention to detail and perfectionism, but their PackageMaker utillity -- which is what you use to create those install packages -- does most definitely NOT share any of these virtues. It's one of the worst examples of sloppiness I have seen. PackageMaker simply refuses to include files targeted at /usr (and below) into the bills of materials file (Archive.bom). The files are all there, nicely shrinkwrapped into the archive itself (Archive.pax.gz), but no matter what you do, they won't show up in the BOM file. As a result, the installer will not install but ignore them. I am now going to change the target to /private/tmp and then run a postinstall script (luckily this feature actually works) to move the files into /usr. I will also provide a patching utility for those who have been hit by this. Folks, I am very sorry about the inconvenience. I have tested the installer on various systems beforehand, but I must have missed to wipe everything at some point. My sincerest apologies. First (obviously) run the installer. Since the executables are in the archive.pax.gz file in the installer package, first do a show package contents on the package file, then unstuff the enclosed archive.pax.gz file to the desktop... Then open up a shell, CD to the desktop, and run the following: cat Archive.pax | pax -r sudo cp -R usr/* /usr Thanks for your follow up. Yes, this will work, but I guess that it is a bit too crude for many Mac folks, they like to just click on things to make stuff work. That was the whole point of creating the install package in the first place. I have failed the Mac folks miserably in this regard. So, please hang on there for a little while, I'll get back to you with a new install package and a patching utility. Meanwhile, we've released a few clickable AppleScript script apps for basic control of Asterisk (start/stop/reload/show version). You can download them (as a zip archive) from http:/www.astmasters.net/stuff/AsteriskApplescripts.zip regards benjamin -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __ Do You Yahoo!? http://bb.yahoo.co.jp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Wallingford, Ted.vcf Description: Binary data
Re: [Asterisk-Users] ADSI phone vs. IP phone
Andrew Kohlsmith [EMAIL PROTECTED] writes: Why wouldn't you just use your existing Ethernet infrastructure putting the IP phones inline between the wall jack and the PC? There are a number of IP phones that have builtin switch/hub that allows the PC to daisy chain off the IP phone. To quote myself: True, but I don't have to retool my office and install POE switches to use ADSI phones, either. No, I will not put a hub/switch at every desk and then use wall-warts for every phone to get around retooling the office. :-) I'm not going to bastardize my network by placing the equivalent of a 3-port switch or hub at every desk to have the phone system compete with our heavy network users (CAD mostly), and I will fight tooth and nail against having to put a goddamned wall-wart at every station just to power the damned IP phones. :-) Do ADSI phones need wall-warts, or can they drive themselves from the line power? -- Ted Cabeen http://www.pobox.com/~secabeen[EMAIL PROTECTED] Check Website or Keyserver for PGP/GPG Key BA0349D2 [EMAIL PROTECTED] I have taken all knowledge to be my province. -F. Bacon [EMAIL PROTECTED] Human kind cannot bear very much reality.-T.S.Eliot[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] max queue time; newbie question (fwd)
Martin Pycko [EMAIL PROTECTED] writes: sure, use the 'n' option of the queue and put voicemail app as the next priority Will that work? From my read of the code, the timeout parameter is only checked while the call is being sent to an agent's phone (inside the try_calling function). The timeout doesn't seem to be checked while the user is waiting to get to the head of the queue (inside the wait_our_turn function). Unless the ast_waitfordigit function checks the timeout and I missed it, this solution won't work. Am I reading the code right? On Fri, 9 Jan 2004, Ken Alker wrote: I am just studying Asterisk now and have a question. Is it possible to force anyone who enters a queue into voice mail after they have been in the queue for 30 seconds? -- Ted Cabeen Sr. Systems/Network Administrator Impulse Internet Services ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sample configs / load module failure
Hi List, I am trying to locate some detailed documentation and sample configs. I downloaded and compiled Asterisk, and I haven't been able to find much detailed docs on the config files. The distribution I compiled and installed doesn't have any config files, and the handbook is good but doesn't cover all of the configs. Here's my specific problem, when launching Asterisk for the first time, it fails to launch with the following information: [res_parking.so]WARNING[1024]: File loader.c, Line 212 (ast_load_resource): /usr/lib/asterisk/modules/res_parking.so: undefined symbol: ast_moh_start WARNING[1024]: File loader.c, Line 368 (load_modules): Loading module res_parking.so failed! If this is merely a matter of not using the parking module, that's fine, but I can't find the docs on how to NOT use a specific module. Thanks, Ted ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #1133 - 18 msgs
Overhead paging is when somebody picks up an extension, speaks into the handset, and their voice is broadcast throughout a building or zone. What is overhead paging and how is it done with asterisk? You can do overhead paging and even stream mp3's using nbsd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
