Re: [asterisk-users] Learn Asterisk

I can definitely recommend the 'sit down and play with it' website. Worked for me. PaulH David @ULC wrote: > > What the best website and book to start learning asterisk ? > > > > > _

[asterisk-users] How to force TDM410 to alaw

Hi I am having some problem forcing my tdm410 to alaw over ulaw, I have 1.6.1.0 asterisk (debian i486) dahdi1:2.2.0 built with the hardware echo canceller firmware /etc/asterisk/chan_dahdi.conf alaw=1-4 but I have this in the general section, before any channel definition dahdi s

[asterisk-users] Minimizing downtime during updates

I was about to ask this question when I figured out the answer by combing through the makefile. I am posting this anyway because I think it's good to know, and I didn't find any threads that speak to it when I searched the list history. My Question was: When updating Asterisk, the sound tarballs

Re: [asterisk-users] Realtime extensions

Hi Many thanks for your help. I managed to solve the problem by rewriting the dialplan. I have split the line _06[069] into three different extensions 060, 066 and 069 and now asterisk is matching the numbers as I expect it to do. Maybe it is not the "clean" solution, but ;). Many th

Re: [asterisk-users] Asterisk + mySQL

[Jun 22 17:51:48] ERROR[13726]: cdr_addon_mysql.c:249 mysql_log: mysql_cdr: Failed to insert into database: (1054) Unknown column 'calldate' in 'field list' == Spawn extension I have the following columns (for billing) : accountcode src dst dcontextclidchannel dstchannel

[asterisk-users] Documenting configuration with Real Time

Hello, all. As we work through our design issues, we are very interested in moving immediately to real time database since we anticipate expanding our system in to a clustered system within a year or two. One of the biggest disadvantages we anticipate is the lack of configuration documentation.

Re: [asterisk-users] 400 calls at g711 how much cpu power

So what happened to the OP? Seems he would be eager to help us fight the swine flu... -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Coloc

Re: [asterisk-users] 400 calls at g711 how much cpu power

Let me also know, I just have a business with Tamiflu :) I need to contact 6.000.000.000 people to help them. No spam, I promise. Olivier ContactTel Business a écrit : Lol , simply lol, don’t forget the super duper, top secret patch ,everyone is hiding from you  that makes as

Re: [asterisk-users] 400 calls at g711 how much cpu power

Is this Project Eagle Eye ? Call every phone at once to tell them about H1N1 in their neighborhood From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ContactTel Business Sent: Monday, June 22, 2009 3:46 PM To: 'Asterisk Users Mailing

Re: [asterisk-users] 400 calls at g711 how much cpu power

Lol , simply lol, don't forget the super duper, top secret patch ,everyone is hiding from you that makes asterisk able to do 4000 calls on a p3, PS. don't tell anyone i said this . But yeah , since you need to blast 500 calls+, you should be aware that normal blasting even 4 seconds audio

Re: [asterisk-users] Asterisk on AVR32

When you run configure, you need to spec the "host" parameter for the architecture and environment you will be running under. For example, a 1.4 distro being built for a blackfin running uclinux would run ./configure host=bfin-uclinux This will imply that the compiler being used to build the

Re: [asterisk-users] Learn Asterisk

On Mon, 2009-06-22 at 23:22 +0530, David @ULC wrote: > I am from the eastern part of India and there is No institute to have > a formal training for Asterisk. Digium does have an authorized training partner in India, but since this is a non-commercial list, I kindly ask that you contact me directl

Re: [asterisk-users] Realtime extensions

On Mon, 2009-06-22 at 12:30 -0500, Danny Nicholas wrote: > Hey Jared, Do you have a FAQ reference for some commonly asked questions > like this? I've got an email archive that goes back almost a year, but > finding a reference in 4K+ emails is sometimes difficult, even if I can > remember some ke

Re: [asterisk-users] Learn Asterisk

I am from the eastern part of India and there is No institute to have a formal training for Asterisk. I would like to open One Institute dedicated for Training. ( Though I am also learning ) Any advice is highly appreciated. On Mon, Jun 22, 2009 at 10:50 PM, David @ULC wrote: > > What the best

Re: [asterisk-users] Learn Asterisk

On 22/06/09 18:20, David @ULC wrote: > > What the best website and book to start learning asterisk ? Website: Google, http://www.voip-info.org Book: TFOT (The future of Telephony) Google for it , it is freely/legally downloadable. ___ -- Bandwidth

Re: [asterisk-users] Learn Asterisk

www.voip-info.org is a good resource IMO; the O'Reilly online book would be considered the standard, although it is oriented toward a 1.2 or 1.4 user. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf

[asterisk-users] "internal_timing" not working (re: SIP silence suppression)

We are trying to get Asterisk to behave correctly when our SIP clients have "Silence Suppression" turn on, but are not having any luck. Basically, there are several apps in Asterisk which won't send any audio to the SIP client, unless the SIP client itself sends audio to Asterisk (which it

Re: [asterisk-users] Realtime extensions

Hey Jared, Do you have a FAQ reference for some commonly asked questions like this? I've got an email archive that goes back almost a year, but finding a reference in 4K+ emails is sometimes difficult, even if I can remember some keyword to help out. -Original Message- From: asterisk-use

Re: [asterisk-users] Learn Asterisk

On Mon, 2009-06-22 at 22:50 +0530, David @ULC wrote: > What the best website and book to start learning asterisk ? I'm obviously biased (as I'm co-author of the book), but I recommend O'Reilly Media's "Asterisk: The Future of Telephony", Second Edition. You can download a free PDF of the book at

Re: [asterisk-users] RTP/SIP traffic prioritization and Linux issues

John A. Sullivan III wrote: > Hello, all. I've stumbled across what seems to be a traffic > prioritization issue in a Linux environment and wonder if anyone else > has encountered or addressed this issue. > > We had planned to use expedited forwarding for our RTP and perhaps our > SIP packets. O

Re: [asterisk-users] Realtime extensions

On Mon, 2009-06-22 at 11:24 -0500, Danny Nicholas wrote: > You can prove this by switching the order of the two filters in > extensions.conf. The order that the extensions in extensions.conf appear has no bearing on the sort order. I've explained the sorting mechanism at length in this list bef

[asterisk-users] Learn Asterisk

What the best website and book to start learning asterisk ? ___ -- 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

Re: [asterisk-users] Crash process Asterisk

Hello, i'found this here: https://issues.asterisk.org/view.php?id=7176 normally you will find the best information from a gdb trace at the bottom, where the last function has been called. in your case this was #1 0x00295220 in expire_register (data=0xb677b180) at ../include/asterisk/lock.h:592

Re: [asterisk-users] IMAP voice mail storage

Philipp Kempgen wrote: > John A. Sullivan III schrieb: > >> By the way, how does one disable IMAP storage? > >> I did not see a module for IMAP storage. It would seem strange that I >> would have to recompile. > > Sadly you have to recompile. > Disable voicemail IMAP storage in `make menuselect

Re: [asterisk-users] Realtime extensions

You can prove this by switching the order of the two filters in extensions.conf. IMO these evaluate the same given the number you are testing with. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez

Re: [asterisk-users] Realtime extensions

On Mon, 2009-06-22 at 08:51 +0200, Andrew Nowrot wrote: > Hi > > I am having a problem with extension matching in asterisk (I am using > asterisk 1.6.0.6). Is there a difference between extensions matching > in realtime architecture and extensions matching in extensions.conf > file. > > For examp

[asterisk-users] RTP/SIP traffic prioritization and Linux issues

Hello, all. I've stumbled across what seems to be a traffic prioritization issue in a Linux environment and wonder if anyone else has encountered or addressed this issue. We had planned to use expedited forwarding for our RTP and perhaps our SIP packets. Our plan was to set DSCP to 101110 (by th

Re: [asterisk-users] Different inbound routes for each interface onaTDM800P card.

Presumably this would work on Zap as well since OP was on an earlier 1.4 rel. Taking that ball and running with, a cleaner piece of logic; exten => s,1,Set(IVR_CHAN=${CUT(CHANNEL,-,1)}) exten => s,n,Set(IVR_CHAN=${CUT(IVR_CHAN,/,2)}) exten => s,n,Gotoif($["${IVR_CHAN}" < "3"]?techsupp|s|1) exten

Re: [asterisk-users] Crash process Asterisk

Hi Stefan, Thanks for your answer. You mean that you're not sure if it's a bug, or you don't know the bug reference ? I'm interested to find the bug report but I don't know how to formulate my search. Regards, Adrien -Message d'origine- De : asterisk-users-boun...@lists.digium.com [ma

Re: [asterisk-users] Different inbound routes for each interface on aTDM800P card.

On Mon, Jun 22, 2009 at 09:56:39AM -0500, Danny Nicholas wrote: > Here is one way: > - exten => s,1,Answer > - exten => s,2,Gotoif($["${CHANNEL}" = "Zap-1"]?techsupp|s|1) Zap/1-1 ? Elsewhere I have: exten => s,1,Set(DAHDI_CHAN=${CUT(CHANNEL,-,1)}) exten => s,n,Set(DAHDI_CHAN=${CUT(DAHDI_CHAN,/,2

Re: [asterisk-users] Meetme Talker Optimization

On Sat, Jun 20, 2009 at 8:15 PM, John A. Sullivan III wrote: > Hello, all.  I've been playing with MeetMe and talker optimization > seemed like a great idea.  I activated it as follows: > exten => 201,1,MeetMe(100201,cTo) > No one can hear the talker.  It doesn't matter how loudly the talker > shou

Re: [asterisk-users] Meetme Talker Optimization

On Mon, 2009-06-22 at 10:58 -0400, David Backeberg wrote: > On Sat, Jun 20, 2009 at 8:15 PM, John A. Sullivan > III wrote: > > Hello, all. I've been playing with MeetMe and talker optimization > > seemed like a great idea. I activated it as follows: > > exten => 201,1,MeetMe(100201,cTo) > > No on

Re: [asterisk-users] Different inbound routes for each interface on aTDM800P card.

Here is one way: - exten => s,1,Answer - exten => s,2,Gotoif($["${CHANNEL}" = "Zap-1"]?techsupp|s|1) - exten => s,3,Gotoif($["${CHANNEL}" = "Zap-2"]?techsupp|s|1) - exten => s,4,Goto(regivr|s|1) - exten => s,5,hangup -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:

Re: [asterisk-users] Different inbound routes for each interface on a TDM800P card.

Start by creating two ZAP Groups, i.e. ZAP/G0 and ZAP/G1 then you can route the calls according to ZAP groups inbound and outbound. - /etc/zaptel.conf: Configuration of your hardware interfaces - /etc/asterisk/zapata.conf: A

Re: [asterisk-users] Crash process Asterisk

Adrien Lemoine schrieb: > Hi all, > > > > To remember, Asterisk runs in version 1.2.7.1 on RedHat AS 4. Hello, i am not sure which bug this may be, but i am sure that it has been fixed since the last 6 years since 1.2.7.1 was up2date. update to 1.2.31 or newer and you wount have the bug ag

[asterisk-users] Different inbound routes for each interface on a TDM800P card.

I'm new to Asterisk and inherited this project so I apologize if this question has been asked a hundred time before. I did start with Google but I may not be asking the right questions, because I wasn't finding any answers. I have Asterisk 1.4.24 and FreePBX 2.5 running and using a Digium TDM800P

Re: [asterisk-users] Realtime extensions

On Mon, 2009-06-22 at 08:51 +0200, Andrew Nowrot wrote: > For example when I have these two extensions: > > -- _0699[134]X > -- _06[069]XXX > > that are in the database and number 0699123123 comes in asterisk will > always choose exten _06[069]XXX > and when they are in the extensions

Re: [asterisk-users] Strange res_config_odbc error messages in 1.6.1.1

Tilghman Lesher writes: > Looks like you're using a widevarchar column, which is something I didn't plan > for. It should be in SVN shortly, however. On the regseconds, it's a typo in > the code (UINTEGER4 should be INTEGER4). I have: column type (-9) unrecognized for column 'name' but: `name

Re: [asterisk-users] Cisco 7941G & Auth

I had similar problems with the tftp service within Solaris. I installed the tftp-hpa server and used that instead of the Solaris one and now it works fine. Even in Linux/Unix some tftp servers return the wrong error to the phone if the file doesn't exist. You definitely don't need the tlv fil

[asterisk-users] Crash process Asterisk

Hi all, I posted here some days ago about a crash of my Asterisk process. To remember, Asterisk runs in version 1.2.7.1 on RedHat AS 4. The process crashed again but this time I have previously put the option -g. By following this : http://www.voip-info.org/wiki/view/Asterisk+debugg

Re: [asterisk-users] Cisco 7941G & Auth

Hey Sasa, I have templates of all the files you need here (SEP file, extension file): http://dave.vc/wordpress/wp-content/uploads/2008/11/phoneadd.zip If you need further assistance, let me know. Thanks Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:aster

Re: [asterisk-users] How run AsyncAGI commands in background

Hi Moy, many thanks for clarifying. I'll do some further investigations about it and I'll post the result here. Regards Jose Moises Silva escribió: On Fri, Jun 19, 2009 at 5:32 AM, Jose Arias wrote: Hi Moy, I'm using an asterisk 1.4.18 from scratch patched with the last AsyncAGI patch, w

Re: [asterisk-users] Cisco 7941G & Auth

"Jonathan Thurman" wrote: > What does your SEP.cnf.xml file look like? In my experience, > the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I > had > to specify the firmware version in each SEP file. I am using 8-4-4S, but > for you this would be something like this: > > >