similar to: Bug or Not?

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. I can SIP into all 4 machines and life is great. When I try to IAX from the live machine to

Hi gang, Can someone shed some light on the pros and cons of working with the SVN branches of Asterisk vs working in the 1.4 or 1.6 branches? What branch does the SVN release roughly equate to? TIA Danny Nicholas -------------- next part -------------- An HTML attachment was scrubbed... URL:

OK, last showstopper that I just can't puzzle my way through - parking calls with the snom phones. I get the two phones connected, I hit transfer on one, the other phone goes to MOH and the first phone gives me DT, so I dial 700 and hit the OK button. Call transferred, the SNOM hangs up before I have a chance to hear which extension it parked to. Is there a way to make the SNOM phones

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

Hi Gang, I'm testing 1.8.0 on one of my machines and this snippet "chokes" on line 7 (works fine with 1.4.30) [tb-account-balance] exten => s,1,Set(BALCOUNT=0) exten => s,n,NoOp(Verbose(acct ${digitacc} pwd ${digitpwd} )) exten => s,n(runagi),Set(TEST_RETURN="NONE") exten =>

Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant

I tried to boot a Centos 8.2 install CD, one burned with Centos-8-2-2004-x86_64-boot . In the setup, it persisted in telling me that ethernet thing enp0s25 was disconnected. Nyet. 'Twas working several seconds previous and is working now. This is a showstopper. How do I debug it? Also, whatever else it did, I now have environment variable

Hi Gang, I'm working on a lumenvox app and am having "fun" with the Gotoif's on speech/DTMF recognition. If you're using DTMF to enter a number instead of speech to enter a numeric value, the engine will often return a "confidence score" of 1000 instead of 1-999. Therefore this Gotoif fails: exten => s,n,GotoIf($["${SPEECH_SCORE(0)}"

Hi Gang, I've got the latest SVN branch of 1.4 downloaded onto SUSE 11.0. Everything is happy EXCEPT, I can't get fax to be recognized by make menuselect. I tried copying app_rxfax.c and app_txfax.c to the apps directory and starting again from ./configure, but no joy. Any suggestions? Danny Nicholas -------------- next part -------------- An HTML attachment was

Hi gang, Since I'm getting no joy from device_Status or SIPPEER in 1.4.26-rc1, I thought I would do an AGI to read my hints and check for line in use that way. The AGI works fine from a prompt, but returns the dreaded "utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I try to run it from the dialplan. Here is my dialplan snippet;

> Date: Wed, 28 Jan 2009 13:11:19 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005> >

Hi gang, We are moving our computers from a cluster of physical machines to a VMWARE server with virtual machines. We investigated and are looking to replace our TDM400P/TDM410P with AEX410P cards. Can we run asterisk with the DAHDI drivers from one of the Virtual machines or is DAHDI going to have to be a native process on the "REAL" machine? Thanks Danny Nicholas

Hi gang, In 500 words or less (if possible), please explain what is a legal music-on-hold file? My boss hates the stuff provided with the distribution and I figure that I'm asking for trouble if I take my Les Mis tracks and run them through Audacity and SOX to make new files. Thanks in advance Danny Nicholas -------------- next part -------------- An HTML attachment was

On 8/27/10, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body

Hi List, Another dumb conversion question (I hope). I installed 10.0 and copied my 1.4 configuration files over. With a few tweaks everything works great except for 1 feature that I specifically went to 10.0 for. When I do an attended transfer, I still get the receptionists caller ID on the transferred phone instead of the incoming callerID. My assumption is that there is some

Hi Gang, Running Asterisk 1.4SVN using Polycom 501 phones. Just enabled CallerID and for the most part it works as good as you'd expect anything to from the phone company to. Except: on about 1 out of 10 transfers, instead of getting a callerid of "joe cool <100>" or "abc company <205-555-1212>" I just get "asterisk". Here is what

Date: Fri, 1 Oct 2010 18:40:40 -0300 From: Rodrigo Lang <rodrigoferreiralang at gmail.com> Subject: Re: [asterisk-users] AMI Originate To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <AANLkTikV+32vKVSkAFmkDciOPn+rO=k3jYJmsZLNj1QS at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" 3

Thanks for the speedy response, Danny. So you recommend I run something like: asterisk -vvvvr | tee ast-help.txt Then when I need help on a command I request it on the command line, exit to the shell, edit (or whatever) the .txt file to find the command syntax I am looking for, then re-enter the asterisk cli? Kind of defeats the purpose of 'online help' doesn't it? Not trying

Hello list, This may or may not be Asterisk related, but if I had hair I'd pull it out over this. I have a TDM400P card in a Dell POWEREDGE 1550 running Asterisk 1.4.30. Everything works great except that every time it rains, I get flooded with this CLI message - == Starting post polarity CID detection on channel 1 -- Starting simple switch on 'DAHDI/1-1'