Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Friday, April 21, 2017 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Hack attempt

Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Friday, April 21, 2017 10:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk

Re: [asterisk-users] Turn on SIP debugging from DialPlan

; > System(pkill tcpdump); > Hangup; > > Or whitout RTP: > > System(tcpdump -nq -s 0 -i eth0 -w /tmp/sip.pcap port 5060 > &); > Wait(1); > Dial(SIP/${EXTEN}); > Sys

Re: [asterisk-users] Turn on SIP debugging from DialPlan

ul in tracking down in and out of band DTMF > problems that we were having with various carriers. > > Tim > > On 2/17/17 3:07 PM, Derek Andrew wrote: > > The SIP trace will be adequate but this is on a remote system with > > limited disk space. > > > > I would love t

Re: [asterisk-users] Turn on SIP debugging from DialPlan

But how do you turn on the debugging from the dialplan? What would be cool is: same => n,TURN ON DEBUGGING On Fri, Feb 17, 2017 at 5:09 PM, Victor Villarreal <mefhigos...@gmail.com> wrote: > Hi Derek, > > SIP debug can be enabled via Asterisk CLI (console) with the command:

Re: [asterisk-users] Turn on SIP debugging from DialPlan

, Feb 17, 2017 at 4:56 PM, Tim Pozar <po...@lns.com> wrote: > Why not capture the packets with something like tcpdump and run it > through Wireshark? > > Tim > > On 2/17/17 2:43 PM, Derek Andrew wrote: > > I have some troublesome numbers that I would like to capture

[asterisk-users] Turn on SIP debugging from DialPlan

I have some troublesome numbers that I would like to capture the SIP dialogue when I am calling them. When I am about to dial the number, is there any way to turn on SIP debugging in the dial plan before I make the call? (and turn it off after the call is completed?) --

Re: [asterisk-users] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Maier Sent: Wednesday, November 30, 2016 12:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

[asterisk-users] queue show - extensions in call going from (in use) to (not in use)

is no longer in use? My thought process would be that if the agent is in a call, that it should always be 'in use', but perhaps I'm missing some pertinent information here. Any tips would be appreciated. Thanks, Derek B

Re: [asterisk-users] Calls are dropped after 15 minutes

Set session-timers=refuse in sip.conf and do a sip reload. We had this problem with a handful of devices and this ultimately stopped the issue. Thanks, Derek B. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf

Re: [asterisk-users] Registration server with PJSIP

the other PJSIP-related tables such as ps_endpoints, ps_aors, etc. I could be mistaken but perhaps `sippeers` is the table you’re looking for. Thanks Derek B. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: July 2

Re: [asterisk-users] Double queue calls being delivered to agents

Looks like it missed 13.9.0 ☹ Thanks, Derek B. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett Sent: Wednesday, May 04, 2016 10:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-us

Re: [asterisk-users] Double queue calls being delivered to agents

Awesome. Thanks again Richard. On May 4, 2016, at 10:59 PM, Richard Mudgett <rmudg...@digium.com<mailto:rmudg...@digium.com>> wrote: On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett <rmudg...@digium.com<mailto:rmudg...@digium.com>> wrote: On Tue, May 3, 2016 at 6:

[asterisk-users] Double queue calls being delivered to agents

Sorry for last post -- forgot to wipe out the digest contents :/ Derek B -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] Double queue calls being delivered to agents

I took a look through Asterisk 11 and 13 change logs but didn't see any mention of that patch/fix. Am I missing something? Derek B > On May 4, 2016, at 8:50 AM, "asterisk-users-requ...@lists.digium.com" > <asterisk-users-requ...@lists.digium.com> wrote: > > Send

[asterisk-users] Double queue calls being delivered to agents

to eliminate this issue. Thanks for taking the time to read this. -Derek B -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Script to Program Snom Phones

methods. jg -- Copyright 2015 Derek Andrew (excluding quotations) +1 306 966 4808 University of Saskatchewan Peterson 120; 54 Innovation Boulevard Saskatoon,Saskatchewan,Canada. S7N 2V3 Timezone GMT-6 Typed but not read

Re: [asterisk-users] PRI answer too fast

Derek Andrew (excluding quotations) +1 306 966 4808 Information and Communications Technology University of Saskatchewan Peterson 120; 54 Innovation Boulevard Saskatoon,Saskatchewan,Canada. S7N 2V3 Timezone GMT-6 Typed but not read

Re: [asterisk-users] is pattern matching inside macro valid?

:402 attempt_thread: Call completed to SIP/101/009871888729 Anurag Rana http://newbie42.blogspot.in/ -- Copyright 2014 Derek Andrew (excluding quotations) +1 306 966 4808 Information and Communications Technology University of Saskatchewan Peterson 120; 54 Innovation Boulevard Saskatoon

Re: [asterisk-users] asterisk multiple ip

asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Copyright 2014 Derek Andrew (excluding quotations) -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?

Does a reload (not a sip reload) reload everything or does it also require the sip.conf file to be modified? On Wed, Apr 30, 2014 at 5:00 AM, Administrator TOOTAI ad...@tootai.netwrote: Le 30/04/2014 12:39, Administrator TOOTAI a écrit : Le 30/04/2014 12:15, Administrator TOOTAI a écrit :

Re: [asterisk-users] ring groups with different caller id

PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ring groups with different caller id On Tue, 2009-11-03 at 14:02 -0500, Derek Belrose (OMEGABYTE) wrote: Is there a way to ring multiple phones simultaneously but use different caller id settings

[asterisk-users] ring groups with different caller id

on the type of phone that is being called? Thanks, Derek ___ -- 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

Re: [asterisk-users] Asterisk Wiki

bilal ghayyad wrote: Hi List; I am trying to use wiki via the link (http://www.voip-info.org/wiki/index.php?page=Asterisk) in effective way to find the needed resource for me, but still it is hard to arrive for the needed information. For example: what is the best (shortest) way to

[asterisk-users] Avaya SIP phones (4610SW) and MWI

Hi all, I'm new to the list, so I apologize in advance if I'm beating a dead horse by asking this, but I read somewhere that asterisk 1.4 has MWI working for Avaya and their rather troublesome SIP firmware. Can anyone verify this before I go changing phone systems around? Thanks Derek

Re: [asterisk-users] GotoIf Dialplan inquiry

Steve Finkelstein wrote: Hi all, I have the following in my extensions.conf: exten = s,4,GotoIf($[${CALLERID(number)} = 8585979857 | 8585970327]?15:5) The numbers listed above are known spammer numbers. However, when I call from any other CALLERID, it still directs me to s,15 which is

Re: [asterisk-users] Zaptel linux26

Khaled Chehab wrote: I am using centos 4.4 ,when I am compiling zapltel using l make linux26 ,error accrued ,what s missing [EMAIL PROTECTED] zaptel]# make linux26 grep: /include/linux/autoconf.h: No such file or directory make: *** No rule to make target `linux26'.

Re: [asterisk-users] Blacklist

Nitesh Divecha wrote: Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *Set(DB(blacklist/${blacknr})=1)* the number... Does it save in MySQL DB? hyperion*CLI show dialplan app-blacklist-add [ Context

Re: [asterisk-users] function_db_read: DB requires an argument, DB(family/key)

Per Jessen wrote: Lee Jenkins wrote: OK, so I tried this: exten = _X.,1,Noop(CallerId is ${CALLERID(all)}) exten = _X.,n,Noop(blurp) exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) This now appears to execute the first Noop(), skip the second, and then issue the no

Re: [asterisk-users] allowing call every 15mins

Honestly though this is a strange request... Why bother offering tech support if you are only allowing calls for 1 minute every 15 minutes? Why not be honest about it and do this: exten = 1,1,Playback(sorry-we-dont-offer-support) exten = 1,n,Wait(30) exten = 1,n,Hangup ?? -A.

Re: [asterisk-users] asterisk answering machine

Cosmin Prund wrote: If you're learning Asterisk right now, you might try using basic Dialplan first, so less things may go wrong. There's a dialplan function that does what you want: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+GotoIfTime As for white-listing CallerId's, you may

Re: [asterisk-users] Cisco 30VIP Phone

Chris Nighswonger wrote: Is anyone else on the list using Cisco 30VIP phones with the chan_skinny driver? I have tried to catch the one of the developers on the chat relay, but cannot seem to get anywhere. I am trying to understand how the soft buttons are setup. They are apparently

Re: [asterisk-users] Cisco 30VIP Phone

Jason Parker wrote: - Derek Whitten [EMAIL PROTECTED] wrote: if i remember right, most of the buttons on those and the 12SP+ phones don't really work because there isn't a button template in * There is a button template, the problem is that most of the softkeys simply aren't

Re: [asterisk-users] FWD outgoing problem

Wilson Pickett wrote: Dial(Zap/1-1,IAX2/yy:[EMAIL PROTECTED]/xx|60|r) Looks right to me and the call seems to be accepted by FWD. What codecs are you using? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

Re: [asterisk-users] Re: Fedora + Linux Kernel 2.6 for Zaptel/Asterisk Installation

Benny Amorsen wrote: Avoiding html in email would be a good start. ROFLMAO! signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

[asterisk-users] 1.4 sample postgresql configs

Here are some 1.4.x sample basic postgresql configs w/sh1tl1sting, cid rewriting and 8xx # blocking http://www.kfuq.net/asterisk/cfgs/ I forgot to add these configs also have pgsql voicemail storage and uses ogg vorbis for emailed voicemail messages :-D signature.asc

Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions

5. Set(BLACKLIST=${BLACKLIST()}) [pbx_config] 6. GotoIf($[${BLACKLIST} = 1]?shitlisted|s|1:7) [pbx_config] Philipp Kempgen wrote: Thomas Kenyon wrote: Philipp Kempgen wrote: You might use the Blacklist() application in 1.2 (deprecated!).

Re: [asterisk-users] What means: Request to schedule in the past?!?!

Lacy Moore - Aspendora wrote: On 2/22/07, Frederico Madeira [EMAIL PROTECTED] wrote: My asterisk is show me some errors on line registration. This message appear on console: Request to schedule in the past?!?! I could be wrong here, but I think one of the symptoms of that could be not have

Re: [asterisk-users] Type of wake-up Call

Stefan Wintermeyer wrote: Hi, Am 07.02.2007 um 09:53 schrieb Pierre du Plessis: Is there a way to program asterisk to dial an extension Monday to Friday at a specific time and then read a specific string? eg: Kids, go to the bus stop now, you're about to miss the bus! Write a cronjob

Re: [asterisk-users] playing wav49/gsm files on linux?

Dr. Michael J. Chudobiak wrote: How can I play wav49 or gsm voicemail files on FC6? Nothing seems to play them (hxplay, xine, mplayer, etc). I think I have all the normal codec packages installed. I can play regular wav files, but they're too big. - Mike

Re: [asterisk-users] playing wav49/gsm files on linux?

Dr. Michael J. Chudobiak wrote: Derek Whitten wrote: switch voicemail to .ogg format voicemail.conf: format=ogg but you can't actually do that, can you? WARNING[9933]: file.c:984 ast_writefile: No such format 'ogg' mp3 would be better, but it doesn't work either. WARNING[9879

Re: [asterisk-users] windows SIP Softphones ?

Dennis Kavadas wrote: hi all what do must win32 clients use as a free or OSS sip softphone ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] LookupCIDName / LookupBlacklist syntax

WARNING[8384]: app_lookupcidname.c:70 lookupcidname_exec: LookupCIDName is deprecated. Please use ${DB(cidname/${CALLERID(num)})} instead. [WARNING[8384]: app_lookupblacklist.c:104 lookupblacklist_exec: LookupBlacklist is deprecated. Please use ${BLACKLIST()} instead. I seem to be unable to

Re: [asterisk-users] Does X100P decode caller ID?

Leo Ann Boon wrote: It is, and is identified by wcfxo as a Wildcard FXO: Wildcard X100P. So much for The DigitNetworks X100P is detected as an actual X101P card. IIRC, there were 2 Digium single FXO cards - the X100P using the Motorola SM56 and the X101P with Intel/Ambient 537. The X101Ps

Re: [asterisk-users] Does X100P decode caller ID?

Yuan LIU wrote: From: Nilesh Londhe [EMAIL PROTECTED] On ebay, I have seen x100p (or clone) with two different chipsets; 1) has motorols chip 2) has something else I dont call. My experience says that the x100p/clone with motorola chipset shows caller id with default * settings. This one

Re: [asterisk-users] How to exit from console?

Rudolf Ladyzhenskii wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I

Re: [asterisk-users] Re: Voicemail IMAP

From: http://www.oldskoolphreak.com/tfiles/voip/tts-imap.agi #!/usr/bin/perl # # AGI Script that reads back e-mail from an IMAP account. # # Requires the Asterisk::AGI, Net::IMAP::Simple, and Email::Simple modules. # # Written by: Black Rathchet ([EMAIL PROTECTED]) # #

Re: Fwd: [asterisk-users] Some queries on g729 license.

C F wrote: I knew I was doing the right thing, here is the proof, enjoy when you read it, and have a good laugh. -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Jan 8, 2007 8:22 PM Subject: Re: [asterisk-users] Some queries on g729 license. To: [EMAIL

Re: Fwd: [asterisk-users] Some queries on g729 license.

Al Bochter wrote: Derek Whitten Messages like this SHOULD NOT be posted to the list I have been trying to block you from my servers do to your abuse I will add this email address to the list also and contract your service provider. You are not doing the right thing you are acting like

Re: [asterisk-users] Best Hardware for Asterisk Server?

joe a. wrote: Mark Greene[EMAIL PROTECTED] Wrote on: 1/2/2007 12:58 PM: I believe I am going to start out with some refurbished Dell Poweredge servers. They have had a high success rate with a friend. I was going to go that route as well. But, depends on the model. I have several of the

Re: [asterisk-users] Best Hardware for Asterisk Server?

Colin Anderson wrote: ASUS motherboards, in particular, have worked for me perfectly, everytime with both Digium and Sangoma cards. They are also easy to work with and well documented. -Original Message- From: Doug [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 02, 2007 1:04 PM

Re: [asterisk-users] Best Hardware for Asterisk Server?

Steve Edwards wrote: On Wed, 3 Jan 2007, Derek Whitten wrote: no problems on my proliant DL580 Nothing but problems with my DL380's until I ran a non-SMP kernel. Thanks in advance, Steve Edwards [EMAIL

Re: [asterisk-users] How accurate is show translation?

James Harper wrote: Just to give you another (relative) comparison... This is from a VIA processor running at 533MHz, (64KB Cache) asterisk compiled as i586 as it's missing some of the nicer MMX instructions: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc

Re: [asterisk-users] Iptables rule help

Mail list wrote: Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060

Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?

John Novack wrote: Carla Schroder wrote: On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote: On 12/6/06, John Novack [EMAIL PROTECTED] wrote: Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run into some gotcha down the road where there is some

[asterisk-users] (REPOST DUE TO NO ANSWER) translate.c:88 powerof: Powerof 0: No power?? / translate.c:133 ast_translator_build_path: No translator

Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power?? Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power?? Dec 2 17:45:19 WARNING[64722]: translate.c:133 ast_translator_build_path: No translator path from gsm to unknown Dec 2 17:45:19

[asterisk-users] translate.c:88 powerof: Powerof 0: No power?? / translate.c:133 ast_translator_build_path: No translator

Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power?? Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power?? Dec 2 17:45:19 WARNING[64722]: translate.c:133 ast_translator_build_path: No translator path from gsm to unknown Dec 2 17:45:19

Re: [asterisk-users] Setting RTP ports for Asterisk?

Vincent Delporte wrote: Hello When I make calls from home to the PSTN by going through the Net - Asterisk - the Net - VoIP provider - PSTN, I get no sound either way. I assume it's because I must tell Asterisk to use fixed ranges of UDP ports and map ports accordingly on the NAT firewall

Re: [asterisk-users] Voicemail, SQL ODBC

Norbert Zawodsky wrote: RR wrote: Norbert, mate, I don't know why you're having so much problems. Do you wanna post your extconfig.conf here? just to humour us? I have it running with MSSQLServer a more complicated prospect than mySQL which has a dedicated driver for it, and it still works.

Re: [asterisk-users] IAX access to FWD broken?

jason wrote: last I had heard, pretty much all FWD accounts that were created in the past year or so no longer work with IAX. Still don't know why. Timothy Parez wrote: I've got the same problem here. It can't register anymore -- timeout Brian Capouch schreef: I hadn't used FWD for

Re: [asterisk-users] Voicemail, SQL ODBC

Norbert Zawodsky wrote: RR wrote: snip Mate, I can't say it with authority but I'm almost certain that the only DB that a specific driver was written for is MySQL. I think if you use res_mysql.o you should be able to talk to mySql directly without needing ODBC. /snip O.k., Nice to

Re: [asterisk-users] Zaptel drivers for Solaris?

Andrew Joakimsen wrote: Solaris has poor support for anything in general. maybe you are looking in the wrong places for support... SOLARIS IS NOT LINUX signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided

Re: [asterisk-users] random one way audio and noise between SIP phoneson same LAN

Scott Scecina wrote: In all cases, the called party cannot hear the calling party. do you have the RTP ports open? signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

Re: [asterisk-users] Why is this happening?

Brian Candler wrote: On Wed, Oct 18, 2006 at 09:11:15AM -0400, Matt wrote: In the case of you example the IAX2 registration came in from the source port on the far device of 1207. Connections don't just move between ports. I understand all this. However, here is my question. MY on 4569

[asterisk-users] help

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, October 04, 2006 11:03 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 27, Issue 16 Send asterisk-users mailing list submissions to

Re: [asterisk-users] Asterisk calling through FWD?

Nick Ellson wrote: Hi Michael, I tried what you had said and then tried calling you, and it worked. Then I called my brother and while I did not get the error, I still got the busy message i was getting before I borked my config trying too many ideas ;) So, any other 6 digit FWD users

Re: [asterisk-users] Re: Re: 911 versus 9.911

Jim Rice wrote: Pardon me for jumping into the middle of the thread, but how does one actually test 911 (or 9911)? I called Verizon, and after speaking with four different departments and supervisors, they said to just call 911 and tell them that I am going to be calling them to test.

Re: [asterisk-users] Idiot questions

Nilesh Londhe wrote: I would suggest buying a very low price FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping. On 8/24/06, *Adam Collard* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You will need a TDM400 with an FXO module for each line

[asterisk-users] Weird E1 problem

fix it). Its really strange how the other 3 lines work perfectly. Does anyone have any ideas? thanks, Derek -- Derek Conniffe Rivertower Ltd DID Number: 01 440 1806 (International: 00 353 1 440 1806) Ireland: (Local) 01 440 1800 United Kingdom: 0870 068 2368 International: 00 353 1 440 1800

Re: [asterisk-users] need a pointer regarding scripting asterisk

shawn bright wrote: Hello there, i am a newbie here, but i have managed to get asterisk up and running with one zap channel, and various tutorials with dial plans. Cool. What i need to do is a little more complex though. We monitor field equipment for farmers, right now we store their info

Re: [asterisk-users] zaptel on dual processor, How?

Zeeshan Zakaria wrote: How to install kernel sources? On 7/17/06, *Dennis Gilmore* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Monday 17 July 2006 12:05 am, Zeeshan Zakaria wrote: I am trying to install zaptel on dual Xeon processor but it gives error, saying

[asterisk-users] comcast info -- somewhat offtopic

A comcast representative told me the other day they are planning on doubling their internet speed from 8Mb to 16Mb at the end of this year. :-D signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by

Re: [asterisk-users] comcast info -- somewhat offtopic

Dovid Bender wrote: and cable vision now has 30/2 and they will have 50/50 real soon - Original Message - From: Derek Whitten [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 12, 2006 10:45 AM Subject

Re: [asterisk-users] comcast info -- somewhat offtopic

Martin Joseph wrote: On Jul 12, 2006, at 7:45 AM, Derek Whitten wrote: A comcast representative told me the other day they are planning on doubling their internet speed from 8Mb to 16Mb at the end of this year. They certainly don't deliver anywhere near 8Mbits per second here... So I

Re: [asterisk-users] Re: $3,000 server

Don wrote: You know I like this list for questions and answers...not spam about other people's personal problems... No one gives a shit about anything but asterisk herego away... - Original Message - From: Michael Workman [EMAIL PROTECTED] To: 'Asterisk Users Mailing List -

Re: [asterisk-users] NuFone, please send the log file

Ronald Wiplinger wrote: Dear NuFone, Without misunderstanding I ask you again, please send the log file and pay back my money! Not following this request results in the assumption that NuFone is cheating and I will post this info every hour on more Internet places. This should help that

Re: [asterisk-users] NuFone suggests to use Vonage!!!!

Joe Baptista wrote: On Sun, 9 Jul 2006, Andrew D Kirch wrote: To some extent I see your point and have been on the receiving end of one of Jeremy's tirades. I've since decided that NuFone is an interesting study in whether your business can survive with only clueful customers. Some

Re: [Asterisk-Users] HP Proliant server?

[EMAIL PROTECTED] wrote: Has anyone had any experience running asterisk on a dual-xeon HP Proliant server. Have you had any experience setting up digium cards on this? We've only used Dell before and are thinking about upgrading to a hp ProLiant ML350 G4p. ANY comments

Re: [Asterisk-Users] open source sip softphone (Window OS version )

Asterisk guy wrote: are there any open source sip softphone (Window OS version )? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] No ringing being played to remote caller?

defaultzone=us zapata.conf: [channels] group = 0 switchtype=national signalling=pri_cpe channel=1-23 group = 1 context=from-trunk emdigitwait=2500 signalling=em_w rxwink=300 usecallerid=yes immediate=no overlapdial=yes channel=25-72 Any help is appreciated. -- Derek Fedel

[Asterisk-Users] Phone recommendations?

Hi All, I'm looking for a good voip hardphone that has a decent set of the regular features (conference, 2 lines, etc) thats reliable, has decent quality, and isn't too pricey. Does anyone have any suggestions? Thanks in advance. Derek -- Derek Fedel

[Asterisk-Users] Asterisk not waiting for EM Wink (I think)

and a Linksys PAP2 device, with the same results. (Both SIP) Any help would be appreciated, Derek zaptel.conf: span=1,1,0,esf,b8zs em=1-24 loadzone = us defaultzone=us zapata.conf: [channels] language=en context=from-pstn signalling = em_w rxgain=2 group = 0 channel = 1-24 /var/log/asterisk/full (snip

Re: [Asterisk-Users] PHP UnixODBC MS SQl 2000

://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:44870975154201497913098! -- Derek Fedel Director of Network Development ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit

Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P

___ --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 !DSPAM:447f25ce303401551342201! -- Derek Fedel Director of Network

[Asterisk-Users] Problem with tor2 driver and Zapata Tormenta 2 Quad T1/PRI Card

. Thanks in advance, Derek ___ --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] Calls connected, but no audio

Miles Scruggs wrote: Hmm all your questions are covered in this email, but I'll summarize it again in this reply: Server: 1.2.7.1 direct connection to the Internet config settings: [pap2] type=friend secret=something qualify=yes nat=yes host=dynamic canreinvite=no context=private

Re: [Asterisk-Users] # key

Akpome Akpoguma wrote: When I use the # key to interrupt an application it does work. Pls is there any idea on what could be wrong? Rgds, _ Don't just search. Find. Check out the new MSN Search!

[Asterisk-Users] Are my expectations too high?

I'm trying to use Asterisk with a TDM400P and 2 analog lines, but I'm having a hard time getting the kind of audio quality that I'd like. I'm hoping to be able to use SIP phones to make calls through Asterisk and have the same quality as a regular analog phone connected to the PSTN. Are my

Re: [Asterisk-Users] Are my expectations too high?

Get an FXO card with hardware echo cancellation. I use the Sangoma A20002D (four FXO ports with echo cancellation). It definitely costs more, but the hardware echo cancellation makes a huge difference in call quality! Software echo cancellation doesn't really work... With this card, would you

Re: [Asterisk-Users] Are my expectations too high?

there that might benefit you in a situation like this. Go look through your zaptel source tree for fxotune and see if it cant possibly correct some of the problem you're having. Thanks for this suggestion. I ran the test and activated the settings with fxotune -sand I'll see how it works. I

Re: [Asterisk-Users] Are my expectations too high?

No it's not. There will be artifacts in any TDMXXX TigerJet Digium analog card, IMO. These artifacts are mitigated through the black art and dumb luck of different chassis, local RF interference, different handsets, different Asterisk version, etc. But you will most likely never get the exact

Re: [Asterisk-Users] Are my expectations too high?

Welcome to computer telephony. :-) I'm actually working in the computer telephony field and have been for the last 10 years, but I deal mainly with T1s and trunk adapters on RS/6000s. I'm a software person so I don't do a huge amount telephony configuration, but I have done my share over

Re: They are? Re: [Asterisk-Users] Now that Nufone is dead...

Jens Vagelpohl wrote: On 23 May 2006, at 16:35, Andrew Kohlsmith wrote: On Tuesday 23 May 2006 10:48, Carlos Chavez wrote: Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? That's news to me; I terminate about 5kmin/month through them, except

Re: [Asterisk-Users] Are my expectations too high?

that is affecting the TDM400P itself? I am using the phone in a home office with 4 computers running, a19 CRT, 5 LCDs, a wireless access point, router, cablemodem, switch and a million and one cables and wires behind my desk. Derek ___ --Bandwidth and Colocation

Re: [Asterisk-Users] Delay when ringing internal extensions on incoming zap call

Going to AMP, Setup - General - Extension of fax machine for receiving faxes = disabled *should* disable fax detection by causing it to use a different branch of the AMP macro's... I did set it to disabled, but it still called NVFaxDetect() with a parameter of zero.

Re: [Asterisk-Users] Audio problems 50% of the time.

of download bandwidth available, but they have a hard time hearing me and I tend to break up a lot. Derek On 5/17/06, kurt x [EMAIL PROTECTED] wrote: I have an Asterisk server that I use at work. I have a phone that is at home that logs into the Asterisk server at work. My home phone is hooked up via

[Asterisk-Users] Delay when ringing internal extensions on incoming zap call

I have a TDM400P with 2 FXO cards and I'm using [EMAIL PROTECTED] 2.8 I noticed that when I place a call to the analog lines from outside, Asterisk takes a while to actually ring the extension the call is being sen to. I've been doing some tests, calling from my cellphone and here is what I

Re: [Asterisk-Users] Delay when ringing internal extensions on incoming zap call

- After the first ring on my cell, Asterisk logs to the CLI that is has an incoming call - After the second ring, it kicks off part of the incoming call context I fixed this by setting: usecallerid=no in zapata.conf I made this change and it helped in that it reduced the number of rings.

Re: [Asterisk-Users] Delay when ringing internal extensions on incoming zap call

to remove the NVFaxDetect() by only editing *_custom.conf files? - According to the documentation I was able to find, the 0 in NVFaxDetect(0) means to not wait, but obviously it still waits at least one ring. Derek ___ --Bandwidth and Colocation provided

[Asterisk-Users] TDM400P static on call

I just got a TDM400P with 2 FXO cards. I got it all configured and I can place and receive calls. I seem to be getting static on the call, mainly when I speak. E.g, if I call someone, I can hear them just fine, but they would hear static. Not a lot...more like a constant background hissing

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