Re: [asterisk-users] Remove ANSI colour trings from log files only

You can post process the logs with something like sed. See: https://superuser.com/questions/380772/removing-ansi-color-codes-from-text-stream On 7/23/20 5:10 PM, Andrew Yager wrote: Hi, Is there a way to drop the ANSI colour strings from log files? In particular, I've got JSON logging throwi

[asterisk-users] Hardening FreePBX installs...

I did a quick and dirty blog entry called "Protecting your Asterisk PBX…" as I wanted some checklist for going through and locking down exposed services, etc. for FreePBX rollouts. Folks can see it here: https://blog.lns.com/?p=177 Feedback encouraged. Tha

Re: [asterisk-users] Iridium integration / gateway

n their network as a result. I can try to answer questions on my use case. Best, -Tim On Tue, Apr 3, 2018 at 10:01 PM, Jean-Denis Girard wrote: > Hi list, > > I have a request to integrate Iridium in a Asterisk system. A quick > search didn't return much: I expected to find product

Re: [asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?

ally you'd need the functionality of the monitor split to only stream the remote user's voice, then you'd need to pipe that to a Google Speech API tunnel. That's probably not something you can hack away at with simple Asterisk dialplan applications. Cheers, -Tim On Sat, Ja

Re: [asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?

PLAYBACK "I didn't understand that" GOTO to Anchor-point ELSE Goto next step using valid decoded DTMF data Catch-all, should never get here. /Pseudo-code Don't forget to filter your user sourced data against your white-list, always assume users are hos

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

to mitigate your exposure. Be aware it's not *JUST* you, it's the whole internet getting probed for this stuff (and your IP address is guaranteed to be a valid output of some random number generator), and that means the government too. They have systems in place to handle those attacke

Re: [asterisk-users] Asterisk Voicemail changes

-users] Asterisk Voicemail changes On Friday 01 Sep 2017, Tim Turpin wrote: > Is there a way that I can modify the source code for the voicemail > application? I need to change some of the options in the user’s > interface to make it work like an existing system that I’m replacing. $ vi

Re: [asterisk-users] Asterisk Voicemail changes

Is there a way that I can modify the source code for the voicemail application? I need to change some of the options in the user’s interface to make it work like an existing system that I’m replacing. Thanks. Tim From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users

Re: [asterisk-users] Asterisk Voicemail changes

onfig? On 31 August 2017 at 23:07, Tim Turpin wrote: Thanks for the info, but not really what I’m looking for. If possible, I’d like to modify the source and re-compile the existing voicemail to make it match what I have today. Thanks. From: asterisk-users-boun...@lists.digiu

Re: [asterisk-users] Asterisk Voicemail changes

/Config_minivm_examples.html That said, I don't know if it's actually actively developed or stable (docs last updated 2015 - Asterisk team?) Also make sure your Asterisk is up to date because of https://issues.asterisk.org/jira/browse/ASTERISK-27103 On 31 August 2017 at 16:12, Tim Tur

[asterisk-users] Asterisk Voicemail changes

Is there a way that I can modify the source code for the voicemail application? I need to change some of the options in the user's interface to make it work like an existing system that I'm replacing. Th

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

plain text... (facepalm). Probably why spam has been a problem on this board. -Tim On Fri, Jun 16, 2017 at 5:51 AM, John Novack wrote: > > Jonathan H wrote: > > On 16 June 2017 at 08:38, J Montoya or A J > Stiles wrote: > > > It's hardly Digium's fault, if Goog

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

fectively. At this point I'd like to see some response from the mailing list admin about any root-cause efforts, AFAIC this is starting to smear the Digium/Asterisk brand's ability to handle IT related issues... No response = no confidence vote. -Tim On Tue, Jun 13, 2017 at 7:04

Re: [asterisk-users] How to detect fake CallerID? (8xx?)

t specified way-points for a read to occur and be logged, and the patrol or janitor turns in the reader after every shift for download and auditing. -Tim On Wed, May 10, 2017 at 8:11 AM, Steve Edwards wrote: > I have a 'time and attendance' application. Think janitorial or securit

Re: [asterisk-users] Call does not go to voicemail

When a user calls an extension, the number they enter can feature a "Goto" with a text entry in the dial plan. This makes it harder for those at a phone to go places in your phone system they shouldn't. -Tim On Mon, May 8, 2017 at 4:51 PM, wrote: > On 05/08/2017 04:37 PM, Tim

Re: [asterisk-users] Call does not go voicemail

e bigger one is causing the annoying unexpected behavior in your dial plan Try doing the extension 4 without the Line_1 and see what happens: "... exten => 4,1,Dial(${FD_L2},20,trw) exten => 4,n(vmail),Voicemail(4) exten => 4,n,Hangup() ..." I'll bet Line_2 never ring

Re: [asterisk-users] Call does not go voicemail

sical extension is buggy, and 2) you need to label the voicemail entry point and jump to it if the FD_L1 was any other state but BUSY. "... exten => 4,1,Dial(${FD_L1},25,trw) exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2:vmail) exten => 4,n(line2),Dial(${FD_L2}

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

port 80 and 443. It's a good idea to review these resources often (once a quarter, once a month, once a week - your choice), especially after any software or hardware changes on your network. You may find something has been misconfigured at installation, that you would need to address. Be

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

ned (and unless you run a public FTP site, THERE IS NO GOOD REASON). Fail2Ban is a BLACKLIST method, blacklists are most effective after good network hygiene is implemented, as you drastically limit the pool of potential bad actors with a whitelist. Best, -Tim On Fri, Apr 21, 2017 at 9:38 A

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

does the right thing in putting audio back together. Very helpful 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

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

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 the SIP > dialogue when I am calling them. When I am about to dial the number, is > there a

Re: [asterisk-users] RS485 Audio device

ake a complete audio circuit. Just a thought... -Tim On Wed, Nov 2, 2016 at 1:30 PM, Sylvain Rochet wrote: > Hi, > > On Wed, Nov 02, 2016 at 04:21:07PM -0400, Eric Wieling wrote: > > If cable can be pulled , you have a couple of options. > > > > Long Reach Ethernet fr

Re: [asterisk-users] cloud solution?

will be an option for Asterisk directly from Digium, but for now you can do that yourself ;-) -Tim On Tue, Sep 27, 2016 at 11:58 AM, Jonathan H wrote: > Something like this? > https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/ > blob/master/Asterisk-13-on-Ubuntu.md

Re: [asterisk-users] SIP trunk down. Wireshark shows ICMP Communication administratively filtered

those ports from that(those) address(es). That's your first line of defence on toll fraud. If the firewall has a VoIP helper function that opens up UDP ports as needed during a session start, that should reduce your attack surface too. -Tim On Wed, Sep 21, 2016 at 3:15 AM, Olivier wrote: &g

Re: [asterisk-users] Blacklist callers from file

x27;s what I'll be doing for my aging parents, who feel helpless to the effect of the bothersome calls, and don't want a complicated solution to the issue. -Tim -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Blacklist callers from file

the Macro, as John hasn't responded yet anyway. Kevin what phones are you using? Depending on how your phones work, there are various way to get this to happen. If your phones don't have hot keys, a quick and dirty way to "kiss-off" unwanted callers is to create an extension tha

[asterisk-users] Blacklist callers from file

gives my users a bit of power as they don't have to be polite, or think of an evasion to the sales person, they just need to push a button and hang up the phone when they realize is a sales call. I didn't want to publish code for this since it's a bit long winded, but it should give you

Re: [asterisk-users] Toll free pattern matching

Don't forget to handle the other special extension cases in a [context]. What I find is a good practice is to write a root "catch-all" (template), then I tag that onto any new [context](template). [Special-Extensions] exten

Re: [asterisk-users] VoiceMail and SMS

the best practice way that emerges after a few years. Gotta keep maintaining your work. -Tim On Fri, Jul 15, 2016 at 8:29 AM, Joaquin Alzola wrote: > Hi Guys > > I am asking too many questions because we would like to use Asterisk first > as a proof of Concept and check from there we

[asterisk-users] Tim's band DEEPFALL NOT SPAM!!

First of all I apologize for emailing everyone in one mass email like this, but it is the only logical way to get this done. We have restarted the Kickstarter campaign in hopes of raising the funds needed to get us into the studio with a national producer. PLEASE DONATE IF YOU CAN! No Donation is

Re: [asterisk-users] small homebrew pbx

>> >> >> Once the FXS is set up, it's just a matter of adding a ring group/pickup >> group: >> http://edoceo.com/exemplar/asterisk-call-groups > >Again, I guess you meant the FXO port of the device. Yes, I

Re: [asterisk-users] small homebrew pbx

FXS is set up, it's just a matter of adding a ring group/pickup group: http://edoceo.com/exemplar/asterisk-call-groups Regards, Tim On Mon, 15 Jun 2015 16:46:13 +1000 wrote >Hello all, > >I'm new here and I'm interested in building a small PBX with asteri

Re: [asterisk-users] Strange Issue: asterisk deleted

ribe (also on an Asterisk box) via a third party recently, but don't have any real specifics other than it being Asterisk 1.4.x on Debian (5 or 6), running on a local LAN, no outside access. Curious if there are any commonalities to the two compromised systems. --Tim -- __

Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

- Original Message - > > > On 22/10/2014 11:23 AM, Tim Nelson wrote: > > Greetings- > > > > Working with the T.38 gateway functionality that is sparsely > > documented > > [1], I'm attempting to get the following functional: > >

Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

- Original Message - > > > On 23/10/2014 10:07 PM, Larry Moore wrote: > > > > > > On 22/10/2014 11:23 AM, Tim Nelson wrote: > >> Greetings- > >> > >> Working with the T.38 gateway functionality that is sparsely > >> docu

Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

- Original Message - > > > On 23/10/2014 3:55 AM, Tim Nelson wrote: > > - Original Message - > > > >> Greetings- > > > >> Working with the T.38 gateway functionality that is sparsely > >> documented [1], I'm attempting

Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

- Original Message - > On 10/22/2014 03:55 PM, Tim Nelson wrote: > > - Original Message - > > > >> Greetings- > > > >> Working with the T.38 gateway functionality that is sparsely > >> documented [1], I'm attempting to get the f

Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

ind a solution contained within > the responses [2]. > Thank you, > --Tim > [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway > [2] > http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T

[asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

ts with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2] . Thank you, --Tim [1] https://w

Re: [asterisk-users] Polycom DND + Intercom/Paging Override?

- Original Message - > Tim, > I THINK but I'm not sure that you can do this with the Polycom > multicast page function. Have you attempted this yet? > Thanks > david Given the odd nature of multicast paging with Polycom, I was hoping to avoid such a setup. My re

[asterisk-users] Polycom DND + Intercom/Paging Override?

ties, but I've had no such luck yet. Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.as

[asterisk-users] Asterisk Call Redirection

Hi Guys, I am able to divert a incoming phone call from asterisk to a sip softphone. Is it possible to redirect a call to a serial port? If so how would I do it? I don't mind a brief explanation. There is a ppp/dialup server listening on serial port. Thanks,

Re: [asterisk-users] SIP OPTIONS "storm"?

- Original Message - > On 13 Feb 2014, at 18:10, Tim Nelson wrote: > > I recently experienced an odd situation. I have an Asterisk 11.5.0 > > system (Box A) with a SIP peering to another Asterisk 1.8.23.0 > > system (Box B). At some point, Box A started sending o

Re: [asterisk-users] SIP OPTIONS "storm"?

equests. The issue is why was Asterisk sending out 65Mbps worth of them to one peer? I did get a capture of the traffic, but nothing appears to explain *why* the traffic was there to begin with. --Tim -- _ -- Bandwidth an

[asterisk-users] SIP OPTIONS "storm"?

11.x branch. Thoughts/suggestions? Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asteris

Re: [asterisk-users] CTI

, operation, and support has been. --Tim -- _ -- 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/hell

Re: [asterisk-users] AstDB Partial Replication?

er. I was about 90% there, but didn't think to clear the existing entries before adding the entries from the other host. Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asteri

[asterisk-users] Multi-Voicemail Message?

calling Voicemail(), but how to get input for multiple extensions/voicemails, and delimit them properly for passing to Voicemail()? All ideas welcome. Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] AstDB Partial Replication?

n system A, but > before that removal can sync to system B, we've already imported > that to system A again. > > Does this make sense? > > TLDR; How do I sync AstDB entries between two hosts,

[asterisk-users] AstDB Partial Replication?

ed that to system A again. Does this make sense? TLDR; How do I sync AstDB entries between two hosts, in both directions, while maintaining data integrity? Thanks --Tim -- _ -- Bandwidth and Colocation Provided by http://www.

[asterisk-users] 回复: Fw： OpenVox G400P network registration problems

Hi, This is tech-support from OpenVox, would you mind to send email to tim.j...@openvox.cn for more details about G400P issue? Or contact me via IM below for better communication. Regards, MSN: tim.j...@msn.cn Gtalk: tim.june...@gmail.com Skype: tim.jjune OpenVox Communication Co. L

Re: [asterisk-users] Strange issues with newly rebooted machine

sk. I'm guessing you'll be able to see some errors flow by, but more importantly, maybe the log will stop, showing you exactly what is hanging. Good luck! --Tim -- _ -- Bandwidth and Colocation Provided by http://

[asterisk-users] DAHDI - Tickless Kernel?

, but newer ones are 250. Thoughts? --Tim -- _ -- 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/

Re: [asterisk-users] Asterisk - Soundcard - Recording?

rrect, ALSA is included or at least fully supported whereas other sound systems are 'less than stellar'. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a li

Re: [asterisk-users] Asterisk - Soundcard - Recording?

t; http://oreka.sourceforge.net/about/ > http://www.ubuntugeek.com/how-to-recording-internal-audio-in-ubuntu.html > > What else do you want it to do? No idea at this point. I'm just doing some preemptive legwork for a project coming down the pipe. --Tim -- ___

Re: [asterisk-users] Asterisk - Soundcard - Recording?

- Original Message - > On Tuesday 28 May 2013, Tim Nelson wrote: > > Greetings- > > > > I've got a curious project that I could use some input on. I'd like > > to use > > Asterisk to record some audio channels via USB 'soundcard'. When

Re: [asterisk-users] Asterisk - Soundcard - Recording?

s to be automated on a small system with no GUI, only console/SSH access. As such, running a full featured audio recording/mixing application in realtime (with user control) is not an option. :/ --Tim -- _ -- Bandwidth and Colo

Re: [asterisk-users] Asterisk - Soundcard - Recording?

n omnidirectional room mike, phone handset, etc). --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asteri

[asterisk-users] Asterisk - Soundcard - Recording?

Asterisk these days (Asterisk 11 presumably) Any thoughts? Or, do you have any alternative ideas that would work better than using Asterisk for this? Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-

Re: [asterisk-users] Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2

First thing to *ALWAYS* check is if you have any Asterisk version specific modules (Fax for Asterisk, G.729, etc). Ensure these are not loaded (noload in modules.conf, or simply move them out of the asterisk modules dir). Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105

[asterisk-users] Asterisk 1.8.19.0 - "[2012-12-18 19:19:51] ERROR[24485]: astobj2.c:115 INTERNAL_OBJ: user_data is NULL"

presents, where it's coming from, and of course, if there is a fix for it. All info appreciated, thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] [OT] Polycom IP450 Firmware Issues

- Original Message - > Tim, > > What version are you on? There is a specific upgrade path for pre > 3.3. > Yes, that was the issue. I needed to upgrade to version 3.3 first, then upgrade to latest 4.x was no problem.

[asterisk-users] [OT] Polycom IP450 Firmware Issues

oot, and *ONLY* the new firmware placed there. So, is the Polycom firmware matrix wrong about this phone/firmware compatibility, or am I missing something? The bootrom has also been upgraded to the latest without any problems. Thoughts? My head is getting sore from banging it on my desk... :/ Tim

Re: [asterisk-users] watchdog like functions

'll want to use 'qualify=yes' for your IAX2 peers which keeps registrations active by sending a 'ping' every 60 seconds (by default). Quite a bit of detail available here: http://www.voip-info.o

Re: [asterisk-users] watchdog like functions

our IAX experience that would lead you to discount it as 'buggy'. --Tim -- _ -- 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] SIP - Authenticated vs Unauthenticated Calls

- Original Message - > Tim Nelson wrote: > > Greetings- > > Hola, > > > I'm running into an issue as follows, in simplified form: > > > > A remote Asterisk box, when registered/peered via SIP to a central > > server, and mak

[asterisk-users] SIP - Authenticated vs Unauthenticated Calls

y rambling above... why is this the case? In what circumstances would Asterisk respond to the same peer differently, seemingly at random? I'm happy to provide any details required, but I'm having a brain freeze on what would be relevant at this point. Thanks for any point

Re: [asterisk-users] Odd Sangoma Card Issues

A bad Sangoma card Their warranty is 5 years I believe, no better time than the present to request a new one under warranty. :) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asteri

Re: [asterisk-users] Peer blocking CDR and recording?

- Original Message - > No idea? ): How about showing your dialplan, and the log or console output from when you make the call? I have a hard time believing this number is special in any way... --

Re: [asterisk-users] Call Termination Provider Madness

ve call will normally take one > RTP channel incoming and one RTP channel outgoing. > I have mine set to for example: rtpstart=1 and rtpend=10100. This > should be enough for 100 simultanious calls. 2 RTP ports per session (inbound/outbound media)... that would mean 50 simultane

[asterisk-users] SIP DTMF Flash Event

VoIP phone on the planet... Asterisk sees the Flash event (via the logs), but does not act upon it. Thoughts? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

Re: [asterisk-users] 1.4.43 lost part of dialplan

the opening post... Are you running FreePBX by any chance? It will overwrite your extensions.conf along with several others anytime you 'Apply Changes'. Maybe that's the case? > Judging from 7+ years lurking on this list and never seeing anyth

Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk

ereabouts)... I just stumbled upon "Homer SIP Capture". It's 100% open source, and looks to be what you're in search of. Have a look: http://www.sipcapture.org/ --Tim -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Easy to install CDR-Viewer?

d still be nice, though. Have a look here: http://www.areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54 This is for the old CDR stats package, which still works wonderfully well. --Tim -- _

Re: [asterisk-users] Easy to install CDR-Viewer?

x27;old' CDR-stats which was simple PHP based, not the 'new fangled' CDR-stats these young punks are using... :D --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] Easy to install CDR-Viewer?

simple (productive :p) solution? CDR-stat is about as easy as it gets, assuming you can setup a basic LAMP stack, and edit a config file or two (database parameters for CDRs). What issues are you having with that installation? --Tim --

Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem

rt Hylafax and faxgetty, then retry. That will allow 120 seconds on the dial before hanging up and assuming no carrier. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a liv

Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem

- Original Message - > Thanks Tim, > I tried your suggestion below the logs: > > -- Accepting AUTHENTICATED call from xxx.xx.xx.xx: >> requested format = ulaw, >> requested prefs = (), >> actual format = ulaw, >> host

Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem

all between your IAXmodem peer and the DAHDI channel. You could be experiencing some digit loss when dialing, causing the calls to *sometimes* go through or not. Try adding a 'ww' to your dialstring to allow the POTS line to settle before dialing: Dial(DAHDI/g3/ww${EXTEN}) --Tim --

Re: [asterisk-users] CAS T1 - No Ringback

- Original Message - > On Friday 27 Jul 2012, Tim Nelson wrote: > > Another mystery for the list, hopefully someone has ideas on a > > fix... > > :) > > > > I've got an Asterisk 1.8.12.0 system connected to a CAS T1 > > (ESF/B8ZS, > > frac

[asterisk-users] CAS T1 - No Ringback

#x27; The odd part is, you can see above the dialed number was XXX, but the actual sequence on the trunk as performed was to dial XXXYYY, then some 'waits', then the last digit Y. Is this normal? --Tim -- ___

Re: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0

ad, the presence (status) > updates you are seeing appear on your phones are the statuses that > Asterisk itself generates based on the phones' activity. > Ah, I was suspecting that to be the case. Thanks for the info! --Tim --

[asterisk-users] Polycom Presence with Asterisk 1.8.12.0

stle' background from a Polycom phone I'd be especially delighted. It was set by a user on a device, and there is no option to remove it, or replace it with the blank background which is the default. :/ --Tim -- _ -- Bandwidt

Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash

, you can set it to 'Device and User' mode instead of the default extensions mode so users can 'hotdesk' between phones. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -

Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash

- Original Message - > Thanks Tim. > > One of the problem that I am facing is the complicated generated > configuration for the FreePBX, is it the same thing in the Elastix? > > To understand this complicated generated commands, is there a > documentation to explai

Re: [asterisk-users] Trixbox or FreePBX?

- Original Message - > Hi Tim, > How about AsteriskNow? > > Thanks and BR, > Anam > > On 7/7/12, Tim Nelson wrote: > > - Original Message - > >> Hi All; > >> > >> Based on what I have to use Trixbox or FreePBX? > >&g

Re: [asterisk-users] Trixbox or FreePBX?

ion in some time. At this point, for a standard PBX installation, my recommendations would be (in this order): 1. Elastix 3. AsteriskNOW 2. PBX In a Flash --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-d

Re: [asterisk-users] Can I install Asterisk normally and then installing the GUI

er' than from the tar.gz on downloads.digium.com. But again, that was some time ago, YMMV... --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] DAHDI DTMF problem?

rt to you for determining the DTMF problems. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.ast

Re: [asterisk-users] DAHDI DTMF problem?

e > configured the > same. I can make outgoing calls on the T1 from Asterisk. > > Can someone give me a clue as to what could be causing this? > What kind of hardware are you using to interface with the T1? Is there any chance a HWEC or other DSP is interfering with DTMF? --Tim

Re: [asterisk-users] Can I install Asterisk normally and then installing the GUI "asterisk now"

reePBX in a nutshell. Of course there is better package management (RPM repos from Digium vs source installs or building your own packages), etc. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] sip and extensions

iately. Just FYI, you've broadcast it to (tens or hundreds of) thousands of list readers. I have to believe some are of the nefarious type that would love to use your account for free calling at your expense. :/ --Tim -- __

Re: [asterisk-users] PRI trunk between Asterisk servers does not work.

- Original Message - > Quoting Tim Nelson : > > > - Original Message - > >> > >> Curiously enough, I can't do that at all on Voip3. Not span 3 of > >> course, because only span 1 should exist, but I can't execute "pri > >

Re: [asterisk-users] PRI trunk between Asterisk servers does not work.

on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) --Tim -- __

Re: [asterisk-users] .lock file issue

files once per minute. If this isn't a new bug, it could also be a full filesystem, or maybe the system lost power during an event where a lock was created but not removed? --Tim -- _ -- Bandwidth and Colocation Provide

Re: [asterisk-users] Digium IP Phones D40

ones were "ear-splittingly" loud, both ringers and handset. --Tim -- _ -- 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] Proactive problem monitoring on SIP on Asterisk

> > It works very well. > ...it worked well when you could buy it. Apparently it is EOL now [1]. Sorry for the noise. These aren't the droids you're looking for. --Tim [1] http://www.manageengine.com/products/vqmanager/eol.html --

Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk

ems and (speech) quality. Have a look at VQmonitor: http://www.manageengine.com/products/vqmanager/ It works very well. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to As

Re: [asterisk-users] SCCP Questions

driver does not give you satisfactory results. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] G729 and voice mail

ow > licenses then Asterisk is not licensed for G729 codec. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King > Sent: Tuesday, June 05, 2012 2:32 PM > To: Asterisk Users Mailing Lis

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