similar to: So many Asterisk Patches - Which do I choose anduse?

Ok, I usually use the latest stable CVS, with no patches or modifications. If figured if there was a worthwhile patch, Mark would have already included it. However, there was that neat patch about being able to press a certain key and it'd begin recording in mid-stream, that was an awesome feature and I patched my latest features.c file with that patch. But I keep seeing mentions of other

Before I go and try to write something myself, I'm curious if anyone has a script that they're using for setting and clearing the MWI on a legacy PBX. I need to pick up a Zap channel and dial #63XXX to set the MWI, or #64XXX to clear it, where XXX is the extension number. One complication is that I've got a couple dozen extensions to handle the MWI for, and only four channels to work

I have it running on my Windows 2000 mahcine using STI products and don't have much of a problem. I would guess that it might be something on the workstations instead of the AstTAPI. Also, might be a little faster, easier, cheaper to just upgrade your existing workstations? Just a thought. Feel free to contact me off list if you would like. Thanks, Dustin -----Original Message-----

Hello all, I have a nice dual Opteron server with a lot of disk space I'd like to let Windows ADS groups use. I am running FreeBSD (AMD64) 5.4-RELEASE-p1 with samba-3.0.20,1 I joined the ADS domain. Smbclient works perfectly. Server shows up in "My Network Places" When I click on it, I get a login box and no credentials will authenticate me. Read some of the samba docs, and found

I am currently in the process of testing out realtime support for sip.conf. I have followed all of the directions that are listed in the Wiki, but for some reason this does not work. When utilizing a flat file, I am able to register endpoints without any problems, and calls can proceed. One interesting side effect that I have noticed is that when I am using realtime for sip, I am unable to see

I've got a Zultys WIP2 and Zultys 2x2 both of which support encryption. I have patched my asterisk with srtp (srtp.sourceforce.net) as well as with the patch found at http://bugs.digium.com/view.php?id=5413. I'm trying to utilizing the encryption feature of the two Zultys phones to create an encrypted call, but am having extreme difficulty. I keep getting the following error message

Hi, I still have some trouble with the AGI interface: - I can use EXEC now, but it never gives me the error returned by the executed application, if an error occurs - I can use ANSWER, but I have to put something else behind "ANSWER". If I say "ANSWER", I get "510 Invalid or unknown command". If I say "ANSWER ''" or "ANSWER ." or

> -----Original Message----- > From: Paul Rodan [mailto:asterisk@glitch.cc] > I'll ask a stupid question, how does a user hit an alpha > letter from his touchtone? > > I know that the Cisco 7960's support entering alpha letters, > and it could > potentially do it (maybe), but how does the average end user > enter an a b c or d from their touchtone phone?

Hi, this was my fault, you are right, i tried with a X-lite Professional and the conference (3-way call) is working now, i guess the phone BT-100 doesnt support it, i dont have a BT102D, so i can tell if it works too, bye -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of mmiranda@americatel.com.sv Sent: Friday,

I dont have a cisco acount yet can some bady hel me with the ata18x-v2-16-030401a-1.zip file. thanks in advance Rodney Acosta Coya. Dpto. Tecnologia de la Informacion. racosta@moanickel.com.cu Tel:(53)(24) 62 611 -----Mensaje original----- De: Paul Rodan [mailto:asterisk@glitch.cc] Enviado el: Martes, 23 de Noviembre de 2004 11:24 a.m. Para: 'Asterisk Users Mailing List - Non-Commercial

I am also running a pretty recent version albeit not today's CVS, but CVS-HEAD-11/20/04-11:29:52. D you have problems b/w the Ciscos or only when going out to the PSTN?? I have 35 7960s with a PRI and no problems that you speak of. I do get an occational dropped call but that may be the DHC server lease running out on the phones. Can you tell us a little more about your setup???

ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! ClueCon was put together by Asterlink, the same team of people who helped shape Asterisk into what it is today by writing features, fixing bugs, offering IRC support and assisting with the management of the development effort. We have produced several real-world solutions based

We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried

Hi, Anyone know anything about asterlink? I've heard good things about them. So I thought I'd give them a try. I went to sign-up.... however, I haven't heard from them, I've called them several times between 9am and 5pm EST and no responce. Called today again after 9am EST, no responce. Tried Sales, Billing, and CS, and got the same person's voicemail! Is this a oneman

My PHP script is using AMI's Originate command to make two-way calls. The originate connects the first leg of the calls, plays a file to the first called party, and then uses Dial() from the dialplan to dial the other leg of the call. I'm noticing that only about 30% of the calls make it through successfully. In looking at Asterisk's CDR's, I have noticed a lot of "NO

If your remotes are not reporting any trouble. This may be farfetched but power may be to blame. I have had the ciscos 'freak out' with unstable power. It looks like the load on the power cubes cannot keep the caps loaded to deal with fluctuations. Or you may have a ground loop somewhere. Are the phones plugged into UPSs? I had flaky 7960 work fine after pluged into a Cisco POE switch.

Are IVR and "Auto Attendant" interchangeable terms? They both do the "Press 1 for" thing. Sales is asking me how to word it and I've always used both terms interchangeably.

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

Hello all, Tonights updates seriously broke Evolution as it no longer starts on two 4.4 up-to-date boxes. When I run "evolution" from the cli I get this: $ evolution (evolution:4865): evolution-smime-WARNING **: Failed all methods for initializing NSS The updates included firefox, seamonkey, seamonkey-nspr, and seamonkey-nss. A reboot causes no change, Any ideas? Marvin Eberly

c,a`ll.ed -----Original Message----- From: darryl schindler [mailto:syslinux at zytor.com] To: caleb peres; jake shepard; malik guyer; edward egbert; jasper amin; jacinto malusky Sent: Friday, April, 2004 4:01 PM Subject: choose the expert,choose the quality The sale is on n_.ow ! For Vi.ox`x, Tramadol, V.al',ium, , Xe`ni_cal and 200 m~or~e other pr_esc`rip~t~ion m`,ed's o-n.line. The