similar to: debug logs

I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s -- Abra?os Luis Claudio Mobile + 55 21 9215 2888 Mobile +55 15 9141 8402 Office +55 15 2102 5859 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/b8593cb1/attachment.htm

Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call

Sorry for disturbing, but I sent some messages today and I am not seeing them on this list. Can sombody tell me, in case this message appear on the list. Thank you

I've installed Sven Slezak's Notify module. He gives the follow as an example line to put into extensions.conf exten => s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/ sunnybook) I understand what is going on with this line but I don't know where in the extensions.conf file to put it? Thanks, Chris Griffin cgriffin@33keys.com

I`ve been noticing alot of those messages in the CLI lately: Apr 5 11:18:02 NOTICE[25593]: chan_sip.c:6444 check_auth: stale nonce received from '<sip:reg-1@pbx.domain.com> I haven't changed my configuration in ages. What could be the cause of this suddent appearance? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to get an IAX trunk to failover to a local trunk it the trunk is down. This is what I've been working on: [macro-forward1]; exten => s,1,Dial(IAX2/192.168.1.1/${ARG1},20) exten => s,2,Goto(call-${DIALSTATUS},1) exten => s-CONGESTION,1,Dial(LOCAL/${ARG2},20) exten => s-CHANUNAVAIL,1,Dial(LOCAL/${ARG2},20 ;end macro-forward1 exten =>

Hello, I created a new field named pre_dst of type varchar(80) exactly like dst field in cdr table. In the dialplan I put: exten => _7.,1,Set(CDR(pre_dst)=${EXTEN:1}) and when I call, all goes fine except that pre_dst has always NULL value in cdr. Do you know why? Is something wrong I did? I know that original fields in cdr are only readable, but in this cas pre_dst is one I created

Has anyone ever been party to an integration of ACT CRM platform with Asterisk? Thanks Cory Andrews

We have the latest version of asterisk, on a xeon dell server (2gb ram), with 6 snom320's(latest firmware) and 3 grandstream gxp 2000's (latest stable firmware) and are having a few problems. We have a basic menu that transfers calls to different extensions. The problems can be found on all extensions. We have 2 different incoming providers and the problem happens on both providers. I

I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco, Span 2 sends to my existing phone system(Nortel). My Span1 gets sent to the context from-pri, detailed here: [from-pri] exten => _49XX,1,Set(CALLERID(all)=${CALLERID(all)}) exten => _49XX,2,Dial(Zap/g2/${EXTEN},,twk) exten => _49XX,3,Congestion() exten => _49XX,4,Set(CALLERID(all)="") exten =>

Originally posted by: mailto: Hi all Now I'm making IVR sequance that is customised [mainmanu]. I wish to notify invaid command like a following exten => i,1,playback('your command is ...') exten => i,2,playback(${EXTEN}) ; <---- Say 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I

We have decided to allow our tech's to do support for non-clients of voicemeup.com You can head to http://support.voicemeup.com/ and one will be in touch 8 to 6pm business hours. 3 levels of support are offered for Asterisk/compiling Trixbox , Ivr's etc. -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -------------- next part

Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___________________________________________________________ Yahoo! Answers - Got

Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI> core show version Asterisk

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I

Please note that we are available to fix the current REMOTE crash that affects Asterisk/openpbx/trixbox and crashes these systems via a malformed packet please contacts use if you need a hand to patch your systems. -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -------------- next part -------------- An HTML attachment was scrubbed... URL:

https://bugzilla.mindrot.org/show_bug.cgi?id=2173 Bug ID: 2173 Summary: configure fails to find libedit in non-standard location Product: Portable OpenSSH Version: 6.2p1 Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: Build system

Good morning, On Mon, 2023-06-26 at 16:29 +0200, Michel van Dop wrote: > Hello, > > I use/test now a log wrapper. > He checks which log has already been sent. In this way it only sends > the new data it is fast and almost realtime. > > This was the link that not work about log rotation in old mailing i > found: > https://trac.xiph.org/ticket/2173 This is most

Hi Philipp, Thank you, thats good news that the log on date rotation is in the new version 2.5. And can you indicate what you are waiting for before making it the stable 2.5 release? And in which version of openssl does this problem no longer occur? Sorry for straying from the original question :-) Best regards, Michel > Op 26-06-2023 17:11 CEST schreef Philipp Schafft <phschafft at