[xubuntu-users] xubuntu-users-boun...@lists.ubuntu.com

please unsubscribe from this jason walker -- xubuntu-users mailing list xubuntu-users@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/xubuntu-users

[docker-dev] Re: BUY CLONE CARD NEW YORK ONLINE

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[dolphin] [Bug 368121] Change in sorting mode only has effect after Dolphin to be restarted

https://bugs.kde.org/show_bug.cgi?id=368121 Jason Walker changed: What|Removed |Added CC||jason...@gmail.com --- Comment #5 from Jason

[dolphin] [Bug 369399] No "What's this" help in Dolphin > Share > Share with Samba

https://bugs.kde.org/show_bug.cgi?id=369399 Jason Walker changed: What|Removed |Added CC||jason...@gmail.com --- Comment #4 from Jason

[frameworks-kio] [Bug 154499] Add a little eject symbol next to mounted volumes/disks/CDs/etc in Dolphin's Places panel

https://bugs.kde.org/show_bug.cgi?id=154499 Jason Walker changed: What|Removed |Added CC||jason...@gmail.com --- Comment #13 from Jason

Re: [Discussion][Internals] Remove the Interface suffix from PSR naming conventions

AM Niklas Keller wrote: > On Tuesday, August 16, 2016 at 3:14:14 AM UTC+2, Jason Walker wrote: >> >> I disagree with removing the -Interface suffix. >> >> Not so much that I think it has to be there, but more along the lines of >> if it isn't going to be

Re: [Discussion][Internals] Remove the Interface suffix from PSR naming conventions

I disagree with removing the -Interface suffix. Not so much that I think it has to be there, but more along the lines of if it isn't going to be there then the FIG should select a new, perhaps less verbose, naming convention instead. I feel the suffix gives clarity and informs consumers/implement

Re: [Internal] [Discussion] Paul M Jones

Hi Michael, First off, I wanted to give you a well deserved hat tip for what you've been doing in this thread (and as a secretary) in helping keep this topic in the proper direction, that is, to its conclusion. I'm sure you've had to listen/read to more than your ears/eyes wanted to handle and

Re: [PSR-13] "With-er" methods

How do the prefixes "Configurable" or "Amendable" sound? E.g. ConfigurableLinkInterface ConfigurableLinkCollectionInterface ...or... AmendableLinkInterface AmendableLinkCollectionInterface -- You received this message because you are subscribed to the Google Groups "PHP Framework Interoperab

[Yahoo-eng-team] [Bug 1268728] [NEW] Add 2-leg configuration to Radware LBaaS Driver

Public bug reported: Radware's LBaaS driver currently supports a configuration where: * the load balancer's VIP and PIP are identical and on the same network as the application servers * the load balancer's HA backup (if used) is on a dedicated private network The driver needs to support a ne

[Perldl] Non-zero coordinates/ranges

Using which, I get the non-zero index from my original PDL. What I then want to do, is identify the consecutive indices and boil it down to ranges or coordinates rather than an index. Original PDL: [0 10 20 10 0 0 0 0 1 2 3 2 1 0 0 0 0 0 1 2 3 4 3 2 1 0 0 0] Which Index: [1 2 3 8 9 10 11 12 1

[Perldl] Length of a series of zero values

HI- I am new to PerlDL. I've attempted to read all the documentation I can but I have yet to find a solution to my question. Say I have a 1-D piddle that looks like: [1 2 3 2 1 0 0 0 0 0 1 2 3 4 3 2 1 0 0 0] I can use which_both to determine the index of the non-zero and zero positions like

[RCU] Tiny MCE issue

Error: tinyMCE is not defined Source File: /roundcubemail-0.4-beta/program/js/editor.js?s=1270216758 Line: 20 /* +--- + | RoundCube editor js library | | | | This file is part of the RoundCube web development suit

[asterisk-users] REALTIME in 1.2

I am trying to change a 1.6 realtime statement into a 1.2 realtime statement and I know much has changed. I wish I could just upgrade, but alas not right now. exten =>x,n,Set(NULL1="${REALTIME(schedules,id,${SCHEDULE})}") comes back with pbx.c:1371 ast_func_read: Function REALTIME not regis

[asterisk-users] Recording music in Queue

I know that this is a "feature" but I would like to have the hold music recorded while a person is on hold. So I know the agent put them on hold and not just muted. I have monitor-join=yes monitor-format=wav in my queues.conf any ideas? Per http://www.asteriskguru.com/tutorials/queue

[asterisk-users] D-Channel Span Up without Down

I am getting a bunch of Primary D-Channel on span 1 up but there was not a down message before that. Is this normal? Confidentiality Statement & Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or

[asterisk-users] Realtime Issue

It seems that my realtime is not assigning channel variables correctly. INFO Asterisk 1.6.0.26 Exten.conf exten => _X.,1,NoOp() exten => _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)}) exten => _X.,3,Set(NULL="${REALTIME(agents,device,${DEVICE})}") exten => _X.,4,NoOp(DEVICE is ${DEVICE}) exten

CRASH: 1.6.3 svn editing shortcuts

Dear LyX developers, In LyX 1.6.3 svn (my build is not completely up-to -date though), LyX crashes if you do the following: 1) Tools->Preferences->Editing->Shortcut. 2) Choose an action *with* a shortcut. Press "Modify". Click "Delete Key" until you clear the shortcut completely. If you click "

[magick-users] Help with ASP (COM) use of ImageMagick needed

Hi, I'm a new user of ImageMagick and I'm using it with ASP I'm struggling through how to address IM but I'm struggling with converting this to ASP composite -dissolve 25% -gravity south \ wmark_image.png logo.jpg wmark_dissolve.jpg If anyone could help it would be apprec

[asterisk-users] sip issue with one way audio

I am getting this error [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our c

Re: [asterisk-users] 2 Digit Issue

alplan it is catching the invalid extension and > should be passing it to the i (invalid) handler to loop back into your > attendant. > > > > On 8/1/07, *Jason Walker* <[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]>> wrote: > > I had to switch quickly t

Re: [asterisk-users] 2 Digit Issue

ng (8XXX) -- > Are you sure you didn't have those extensions in another context that > you forgot to include? > > According to the dialplan it is catching the invalid extension and > should be passing it to the i (invalid) handler to loop back into your > attendant. &

[asterisk-users] 2 Digit Issue

I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2 digits into the dialplan. error -- Invalid extension '81' in context 'impact' on SIP/207.174.111.34-b77167f8 I pressed 8107 and ideas my dial plan is (part of it) [impact] exten=>s,1,Answer() exten=>s,n,Set(CALLERID(name)=Im

Debian, urgent and important debt - which are you paying?

p doo-doo my friend! Often we pay those who "shout the loudest", to keep them "off our backs" BUT do you know if that is wise? We know. Contact us NOW for a free consultation: http://www.how-to-beat-debt.co.uk/freeconsultation/ Best Regards, Jason Walker. "How To Beat

Debian, can you just 'walk away' from debt?

Hi Debian, I'm often asked if it's possible to simply "walk away" from debt... and the answer is - "In certain cases yes!" Do you want to know if this applies to you? Then follow this link and find out http://www.how-to-beat-debt.co.uk/freeconsultation/ Best R

[asterisk-users] Web User control

I am looking to allow some users to login to a website and change where their ext is forwarded to. any ideas? It can be very simple or I can install a full package and then allow certain users certain access. Thanks in advance Jason ___ --Bandwidth

Re: [asterisk-users] Linksys not Ringing

I do not have any answer int he dialplan. what I mean is that when I call any other SIP phone is does the answer in the CLI. Even if I put and answer() in the dialplan still no ringing Jason Luki wrote: shouldn't there be an answer in there somewhere?... like... No... you can (and probably

[asterisk-users] Linksys not Ringing

I have 2 linksys SIP phones SPA-942 I have a dialplan of exten => 144,1,Wait(1) exten => 144,2,Dial(Sip/phil,20) exten => 144,3,Voicemail([EMAIL PROTECTED],u) The CLI looks like this when I dial 144 -- Executing Wait("IAX2/JASONSERVER-9", "1") in new stack -- Executing Dial("IAX2/JASONSERVE

Re: [asterisk-users] RE: Polycom reject button

Good Idea, but when the user has to do nothing is better for my users! Thanks JAson Mojo with Horan & Company, LLC wrote: Another option is to have the user hit the forward button on their phone and manually type in their cellphone number when they're going to be out of the office

Re: [asterisk-users] RE: Polycom reject button

exten => 111,1,Wait(1) exten => 111,2,Playback(Randy) exten => 111,3,Dial(Sip/Randy,20) exten => 111,4,Goto(111-${DIALSTATUS},1) exten => 111-BUSY,1,Voicemail([EMAIL PROTECTED],u) exten => 111-NOANSWER,1,Dial(IAX2/${TELIAX_OUT}/212551212) works GREAT Thanks a lot Jason Doug Lytle wrote: Mike

[asterisk-users] 2 Call locations

I have a SIP user and a remote IAX device I want both to ring 3 times then if neiter pick up it to go to the next thing in the dialplan. Can you do this from the dialplan or do I need to set a hunt group up? Thanks Jason ___ --Bandwidth and Colocat

[asterisk-users] Polycom reject button

I have users in my dialplan that go from SIP to Cell When they are at their desk and they hit reject call, it goes to the next thing in the dialplan, thus transferring to their cell. Not what they want. Is it possible to change the reject button to make it go to voice mail or a new ext? Tha

Re: [asterisk-users] Polycom Provistioning Issue

.cfg file look like? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker Sent: Friday, January 26, 2007 12:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom Provistioning Issue From what I know this log show

Re: [asterisk-users] Polycom Provistioning Issue

Fixed that issue but it does not change the error 0126204105|cfg |3|00|Image sip.ld has not changed 0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 0126204105|cfg |3|00|Downloaded application image is identical to current version 0126204105|cfg |3|00|Phone suc

[asterisk-users] Polycom Provistioning Issue

From what I know this log show everything working Perfect. Then it goes to the Welcome screen then after a long time of processing, it errors out with a 0x1 error Any Ideas? 1005195711|so |4|00|-- Initial log entry -- 1005195711|so |4|00|+++ Note that bootrom log time

[asterisk-users] DTMF Tone Issues

I have 1.2.12.1 Voicepulse using IAX I get about 30-40% issues with not having the DTMF tones work. I have 3 questions #1. Voicepulse says they are sending them, Is there some setting I can adjust to make sure my end is working? #2. I have set the Dialplan to play a sound "Operator" then go to

[asterisk-users] Voicemail issues

I put my voicemail groups into different contexts so that I can use Dial by name and escape. I had set ext 500 as exten => 500,1,VoiceMailMain(${CALLERID(number)[EMAIL PROTECTED]|s) but now that the contexts are different. this does not work #1 how do I have everyone use an ext to get the voicem

Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

Ken, Also stay away from Swissvoice phones    I have found several ways to do the second thing. http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers It works great. Jason Tom Vile wrote: I tend to stay away from the Grandstream phones for business use because they simply break to ea

[asterisk-users] DTMF over IAX

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions

[asterisk-users] DTMF Tones

I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct DTMF tones 25% of the time. I have to call several times to enter an extension. I have a router and a packet shaper and some other stuff. Anyone have any other ideas why this might happen. I do not have any Zap channels bu

[asterisk-users] Escape from Voicemail

I used to have fonality and I could press * when I got to someones voice mail to go back to the menu. I assume I add that to the dialplan but how? Thanks BTW I went back to 1.2.12 and transfer works and DTMF works and it seems to be much better for now. Thanks for you help Jason ___

[asterisk-users] DTMF / Silence issues

I am now running 1.4 beta3 I have an ongoing issue that it does not recognize my DTMF key press. I will call and press as many numbers and the background message still plays. I am also having an issue with transfers NOTICE[30930]: chan_sip.c:13289 handle_request_invite: Unable to create/find SIP c

[asterisk-users] 1.4 downgrade

I am having a bunch of issues with 1.4 and want to go back to 1.2 any ideas on the best way I saw someone say "apt-get remove" will this work for asterisk or do I need to do it for each libpri, addons, zaptel and asterisk? Thanks Jason ___ --Bandwid

[asterisk-users] Issues with Asterisk 1.4 Beta

quot; on my Swiss phones Any help would be great. I am a little new to asterisk and so if I posted this incorrectly please let me know Jason Walker ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNS

Re: [Efw-user] VMWare appliance in ESX

With ESX server, you do not have the option of IDE drives; it's SCSI only. I can't find any documentation as to what platform the appliance is geared towards. Any clues? Jason Walker On 9/4/06 7:20 PM, "MorfiusX" <[EMAIL PROTECTED]> wrote: > Try creating an ID

[Efw-user] EFW Release 2 on ESX 2.5

I have installed EFW on Release 2 on ESX server and it appears that I'm having a kernel timing issue. I've run into this with ESX before, and the solution was to patch the kernel, among other things. The easiest way to see this problem manifest is with pings. From the outside world, ping the firewa

[Efw-user] VMWare appliance in ESX

I tried to install the EFW VMWare appliance in ESX v2.5 and was not able to get the disk recognized. Has anyone gotten this working? This e-mail is intended solely for the use of the individual to whom it is addressed and may contain information that is privileged, confidential or otherwise exe

RE: [Asterisk-Users] Dial String Questions

Some phones do not send DTMF automatically. What soft phone are you using? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander LopezSent: Wednesday, January 25, 2006 9:23 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Dial S

RE: [Asterisk-Users] Can't create iax channel

The statement of zaptel being required is strange...I use IX trunking exclusively for my servers. Two of them have no zaptel/Digium hardware and the trunk calls are fine. Based on your post, seems that you have an issue with codecs more than creating an IAX trunk. What version of Asterisk are you

RE: [Asterisk-Users] CentOS vs. Vanilla Kernel

Julian - What hardware are you using? Proc, RAM, SCSI or IDE, etc. The reason I ask is that I have multiple hardware platforms, all on FC1 or FC4, and none of them hit 100% for each IRQ. I am usually in the high 98% with the occasional 100% on P3 servers (give or take 1 Gig RAM, 1 Gig CPU). Two

[Asterisk-Users] HDLC errors on PRI

    I have looked through other postings to the user group for HDLC errors, went through what worked for other people, and still can not seem to get past this issue.   For 3 days, I have been getting HDLC abort(6) errors in *. Prior to Tuesday, the circuits were clean...I had maybe 10 HDLC

RE: [Asterisk-Users] I give up - Help with TE410P

eal T1 attached until I can fix this. Swapping card does not seem to follow issues. Maybe I'll give support another :) Bart - Original Message - From: "Jason Walker" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

RE: [Asterisk-Users] I give up - Help with TE410P

My 2 cents: If you are running kudzu on RH or FC, new and remove hardware should be detected...in most cases. I assume other distros have something similar...? If 2 of 8 T1s are not coming up - sounds like you may have a wiring issue. Can you swap cables from a "bad" circuit to a "good" circuit

RE: [Asterisk-Users] Red Alarms, No D-Channels, and Crazy People

I have not read through the rest of your posts, but try some of the other variations of switchtype: ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: AT&T 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ;

RE: [Asterisk-Users] Red Alarms, No D-Channels, and Crazy People

One thing to consider is if there were alarms on the T1 to SBC, they may have something in place to take the circuit down. Even if you get your configs right, the T1 just might not come up clean. MCI does this to us sometimes. Please post your /etc/zaptel.conf and your /etc/asterisk/zapata.conf

RE: [Asterisk-Users] iax softphone

t: Re: [Asterisk-Users] iax softphone I'm running it on sp2 myself, never had a crash with it so far. Jason Walker wrote: > Are you running on XP SP2just curious? How about the version of *? > > -- > -- >

RE: [Asterisk-Users] iax softphone

Nope,  I do not have that issue. On 10/23/05, Jason Walker <[EMAIL PROTECTED]> wrote: Tom - do you end up with that phone shutting down with an error on Windows XP? I downloaded the latest. After about 3 minutes on a call, the other end can no longer hear me and then the phone jus

RE: [Asterisk-Users] iax softphone

it happen with one specific version of asterisk ? Whatever the problem is, it should not be there. Please help us find the bug. Joachim. Jason Walker wrote: > Tom - do you end up with that phone shutting down with an error on > Windows XP? I downloaded the latest. After about 3 minutes o

RE: [Asterisk-Users] iax softphone

Tom - do you end up with that phone shutting down with an error on Windows XP? I downloaded the latest. After about 3 minutes on a call, the other end can no longer hear me and then the phone just dies. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom VileSent: Saturday,

RE: [Asterisk-Users] Unable to negotiate codec???

9 as their main codec.   My box rejects connections from my provider due to incompatible codecs and vice versa.   I'm waiting for them to get back to me on this.   Clint.     - Original Message - From: Jason Walker To: 'Asterisk Users Mailing List - Non-Commercial

RE: [Asterisk-Users] Unable to negotiate codec???

OZTell, my provider, use G729 as their main codec.   My box rejects connections from my provider due to incompatible codecs and vice versa.   I'm waiting for them to get back to me on this.   Clint.     - Original Message - From: Jason Walker To: 'Asterisk Users Ma

RE: [Asterisk-Users] Unable to negotiate codec???

What codec are you using on the client and the server? From my understanding, you have to have a license for both ends of the G.729 call. Are you passing this through one server to another and the call is being rejected at the server level? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]

[Asterisk-Users] Just a test...

    I have not seen any posts for awhile. Just testing.   thanks     ___ --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 UN

[Asterisk-Users] Multiple instances of asterisk showing from 'ps aux'

      When I run 'ps aux' I get this:   root   964  0.0  0.4 47836 8280 ?    S    00:02   0:00 asterisk -vvvg -croot   965  0.0  0.4 47836 8280 ?    S    00:02   0:00 asterisk -vvvg -croot   967  0.0  0.4 47836 8280 ?    S    00:02   0:00 asterisk -vvvg -croot 

RE: [Asterisk-Users] Digium TDM400P (11B) problems

As an FYI - here is the output of my TDM400P: Module 0: Installed -- AUTO FXS/DPO Module 1: Not installed Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) I do not have newt installed on this machine, so zttool bombs. Just sending this out as an example. He

RE: [Asterisk-Users] Dial 2 channels at onece: Not working anymore atCVS?

What if you force a hangup between the two steps? I have multiple destinations specified when my internal number is called at work using similar syntax. All of the SIP and SCCP extensions dial based on my setup - which again, is very similar to yours. I do not use CVS HEAD on the production box

RE: [Asterisk-Users] ACD calls to busy agents

Have you tried the "incominglimit" parameter (or did she)? I have found this to work pretty well when limiting the number of calls. After monitoring the "full" log, I saw that incoming calls where incrementing or decrementing the active call parameter for SIP agents. By limiting the number of call

RE: [Asterisk-Users] Voicemail 2

Correct - but is the context defined in voicemail.conf? As mickeymouse? Or whatever...? ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Saturday, October 15, 2005 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subje

RE: [Asterisk-Users] Perplexed - IAX trunk == jitterbuffer or nojitterbuffer

2 Oct 2005, Jason Walker wrote: > > I have 4 * servers interconnected with IAX trunks. Three are on a > local LAN, one is accessible over a VPN tunnel out of the office. The > IAX peer status > (iax2 show peers from the CLI) will sometimes show upwards of 300ms. > Considering the

[Asterisk-Users] Perplexed - IAX trunk == jitterbuffer or no jitterbuffer

I have 4 * servers interconnected with IAX trunks. Three are on a local LAN, one is accessible over a VPN tunnel out of the office. The IAX peer status (iax2 show peers from the CLI) will sometimes show upwards of 300ms. Considering the lag and distance, I am not entirely surprised. Anyway - my

RE: [Asterisk-Users] Asterisk and Mitel SX 200 Slip and Frame Errorscausing Major Ala rms

You may have already tried this, but in the past whenever slips come into the picture on my T1s, crimping a new end for the CAT5 cable seems to help. We run T1s to a 110 block. Every once in awhile, the 110 needs to be repunched. I have found that slips can clear up when we rerun the cable...st

[Asterisk-Users] Sangoma DS3 cards + Asterisk

Has anyone used the DS3 card from Sangoma with Asterisk? I have read many posts from users that the Sangoma cards have better echo canceling and so forth. I guess I am just wondering if there are more benefits to using this brand. I currently am responsible for multiple Asterisk servers all wit

RE: :SPAM: Re: [Asterisk-Users] RE: faxing to/from asterisk - newscripts

I would appreciate seeing the scripts as well. Nice job! Desktophero at gmail.com Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Troy Swaine Sent: Friday, October 07, 2005 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion S

RE: [Asterisk-Users] Revieving some fax problems

I have run into a similar situation. One of our older faxes at the office seems to not work with spandsp module. The newer faxes work just fine. When I watch the logs, there appears to be communication from * requesting the fax to "slow down". When the fax machine does not respond, * seems to s

RE: [Asterisk-Users] is a dual 1.5Ghz server better than a single 3Ghz for a 100 Iax users asterisk server

One key that I have found is the more RAM the better. I am not discounting the CPU by any means and with the number of registrations you are talking about, I have not set up a system for that many concurrent users. I do have a 2x1.266 PIII w/ 2 Gigs of RAM that handles 75-85 concurrent SIP (GSM)

RE: [Asterisk-Users] Don't call

It looks like your * server is not able to see the destination (presumably sip.uni.it).No route to destination     -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Fabio MontemaggioreSent: Friday, September 30, 2005 2:34 AMTo: asteriskSubject: [Aster

RE: [Asterisk-Users] Who is going to AstriCon (TheAsteriskConference)?

This would be super-fantastical!!! With all of the other conferences going on, I can only get away so much. I love the idea of a webcast... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Saturday, September 17, 2005 8:37 PM To: Asteris

RE: [Asterisk-Users] Grandstream

That's what I have used...works until you change it. ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rene Kluwen Sent: Friday, September 16, 2005 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Grands

RE: [Asterisk-Users] SIP reinvite asterisk and NAT

I am curious...are you saying to use SIP locally and IAX from point to point (over a WAN or VPN tunnel)? With that in mind, do you think that using a lesser compressed codec over the IAX trunk would give an okay amount of bandwidth savings? Thanks. -Original Message- From: [EMAIL PROTECT

RE: [Asterisk-Users] Is digium supporting new te405p and te406pinstall?

couldn't see any interrupt from /proc/interrupt. My email server has no spam filter. --- Jason Walker <[EMAIL PROTECTED]> wrote: > I have not been able to get * 1.0.9 on a FC4 box...I have an older IBM > server just waiting and try it every so often. When I am using a card > for ti

RE: [Asterisk-Users] Is digium supporting new te405p and te406p install?

I have not been able to get * 1.0.9 on a FC4 box...I have an older IBM server just waiting and try it every so often. When I am using a card for timing (TE405P is what we pretty much use), I feel pretty comfortable with FC1 and 1.0.9. Are you using 1.0.9? Have you tried 1.2 beta? -Original M

RE: [Asterisk-Users] SIP Connection Problems

5000-600?   Do you mean 5060? That is the port for 5060. 1-2 is for RTP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B. Asterisk UsersSent: Sunday, September 11, 2005 12:46 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP Connection Probl

RE: [Asterisk-Users] TE110P reset

? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, September 10, 2005 5:00 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] TE110P reset On Saturday 10 September 2005 19:40, Jason Walker wrote: > PRI chann

RE: [Asterisk-Users] TE110P reset

PRI channels will reset when not in use throughout the day. A reset on a channel should not happen when that channel is in use. This happens all the time on my PRI circuits (TE110P and TE410P). From what I gather, it's somewhat like a handshake for the D chan between the cpe and net sides. -Or

RE: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd

RL within the Queues cmd Jason Walker a écrit : > Now I don't feel so inadequate ;) > > This is exactly what I am doing. Perhaps there is more to this particular > option. > > Here is more information - > > I am testing this on * ver. 1.0.7 (I have another box with

RE: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd

ckman Sent: Wednesday, August 31, 2005 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd Jason Walker wrote: > I am testing this on * ver. 1.0.7 (I have another box with 1.0.9 and another > one with CVS

RE: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd

great and I would encourage anyone to use it. As a side note, Michael is a great guy to work with and is extremely reliable in supporting this software. Thanks, Waldo On Aug 31, 2005, at 10:47 AM, Jason Walker wrote: > I installed/ran both MozPhone and DIAX but did not see in the de

RE: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd

Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Denis Girard Sent: Wednesday, August 31, 2005 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd Jason W

RE: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd

I installed/ran both MozPhone and DIAX but did not see in the debug any information of the URL I "sent". Perhaps the real question is: if optionalurl is used, how is the url "sent" to the device(s)? Has anyone applied this within a solution and is willing to share their experience? Thanks! Jaso

[Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd

From voip-info.org:     Queue(queuename|options|optionalurl|announceoverride|timeout)     'optionalurl' allows you to send a URL to devices that support it.   Does anyone have details on the “devices” that support the optionalurl method of the Queue application? I am wondering if th

RE: [Asterisk-Users] asterisk -rx (or remote connections in general)

Do you have 5 or 6 scripts running against the interface for one instance of an outside script? Or, do you have multiple connections (outside users) attempting to run multiple instances of a script that are pulling 5-6 CLI scripts? This would exponentially increase the real number of scripts being

RE: [Asterisk-Users] asterisk -rx (or remote connections in general)

Try setting your logger.conf to allow full output (uncomment the "full" section) and see if there is something specific to the CLI crash. Be careful though and do not let the logging get out of control, especially on a big system. The file can get huge. -Original Message- From: [EMAIL PRO

RE: [Asterisk-Users] X100P dial out problem

Shot in the dark Do you have to dial '9' on your outside line? Perhaps if you changed your Dial command to this: [outgoing] exten => _9X.,1,NoOp("Call for "${EXTEN}) exten => _9X.,2,Dial(Zap/1/${EXTEN:1}) The :1 will drop the leading '9' when it hits the outside. If this is a regular line,

RE: [Asterisk-Users] Cisco IP Phone 30 VIP

VIP those phones don't use .xml like the 7960s http://voip-info.org/tiki-index.php?page=Configuring%20Cisco%2012SP%20phones %20with%20Asterisk On Wed, 2005-08-10 at 16:49, Jason Walker wrote: > The SEP file should be > > SEP.cnf.xml > > You can also use XMLDefault.cnf

RE: [Asterisk-Users] ZAP bchan and dchan HELP!!

Where are the d chans in the trunk group? Which chan? Here is the example from the zapata.conf.sample ; ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ;group => ,[,...] ; ;trunkgroup is the numerical trunk group to create ;dcha

RE: [Asterisk-Users] Cisco IP Phone 30 VIP

The SEP file should be SEP.cnf.xml You can also use XMLDefault.cnf.xml These have worked for me w/ 7960. What phone are you using? Here is some more information for reference: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx -Original Message- From: [EMAIL PR

RE: [Asterisk-Users] ZAP bchan and dchan HELP!!

Did you setup your T1s as trunk groups? What channels are set up as d chans from the carrier? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Wednesday, August 10, 2005 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion S

RE: [Asterisk-Users] dialplan defenition

For ZAP cards, you can tell Asterisk to answer calls immediately across trunks. Does CAPI have the same type of setting? I am not familiar with Asterisk and CAPI so I am not sure of the options. In Zapata.conf, setting immediate=yes will make the call drop into the 's' extension of the context.

RE: [Asterisk-Users] ip phones

Soft phones or hard phones? There are many free VOIP soft phones out there. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, August 04, 2005 9:57 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ip phones

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