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...uot;Unauthorized". I have stopped, restarted, unloaded & loaded sip, and erased astdb to start from scratch... no dice. None of the config files have changed, and, as I said, they all appeared to work last night. Can anyone give me a clue here? Yours, Yaakov Menken -- Yaakov Menken Capalon Communications, Inc. Ask us about Voice over IP for Business! http://www.capalon.com 888-CAPALON (227-2566) 410-358-9800 x120 410-510-1053 fax 443-413-1042 cell menken@capalon.com

...he is a human being rather than cellphone voicemaill by pressing a digit. This is somewhat similar to the 2nd macro example found at http://www.voip-info.org/wiki-Asterisk+cmd+Dial Is there a queues.conf option that I'm missing here? Thanks for any advice, Yaakov Menken -- Yaakov Menken Capalon Communications, Inc. Ask us about Voice over IP for Business! http://www.capalon.com 888-CAPALON (227-2566) 410-358-9800 x120 410-510-1053 fax 443-413-1042 cell menken@capalon.com

...he use of a meetme room. Does anyone have a good solution for this? What I'd like to implement, ideally, is that once an incoming call is transferred to a particular operator, the system also calls a manager who can monitor silently. Any help is much appreciated! Yaakov -- Yaakov Menken Capalon Communications, Inc. Ask us about Voice over IP for Business! http://www.capalon.com 888-CAPALON (227-2566) 410-358-9800 x120 410-510-1053 fax 443-413-1042 cell menken@capalon.com

...t and play over the phone -- and this will require pause & resume, as well as fast forward / reverse (jump forward / jump back). It doesn't seem like mpg123 can do this. Is there any application that can, that is also compatible with Asterisk? Thank you for all your help! Leah Newmark Capalon Hosting Solutions

...D-Link wireless router (but of course the adaptor is not through the wireless part); 802.11g/2.4GHz Does anyone know of anything that could be triggering this odd behavior or have more detailed questions I can ask her to help pinpoint the problem? Any assistance is much appreciated. Leah Newmark Capalon VoIP

...se on my agi debug. I know for a fact it's running the script I've edited: Launched AGI Script /var/lib/asterisk/agi-bin/myscript.php and it's not in some other directory. Any input: obvious or not is requested...a few people here are stumped! Thank you! Leah Newmark VoIP Programmer Capalon Communications

One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine !!!! What indicates that there is no fault on his Internet connection!!! He is using his

...hat I'll need to wait 23 > > > seconds for Dial() to timeout before I can do anything. I'd like to > > > be immediately able to enter an extension (if possible, which maybe > > > it's not...) > > > > > > On 6/19/06, Leah Newmark <lnewmark@capalon.com> wrote: > > > > Using the Background command, you will be able to play the voicemail > > > > while still being allowed to enter digits. > > > > > > > > exten => s,1,Wait(2) > > > > exten => > 108,2,Background(voicemail/defa...

...ase, I think Vonage is to blame. I found this thread: http://forums.digium.com/viewtopic.php?p=49236&highlight=&sid=d3888f3bb90e5c96b5c0432bd632a2d4 but it doesn't help much. All incoming calls are using IAX. Did anyone have a similar problem and resolve it? Thank you. Leah Newmark Capalon VoIP asterisk-users-request at lists.digium.com wrote: >Message: 8 >Date: Thu, 19 Jul 2007 10:41:44 -0400 >From: Leah Newmark <lnewmark at capalon.com> >Subject: [asterisk-users] Blank Voicemails >To: asterisk-users at lists.digium.com >Message-ID: <469F7828.8000403 at...

...llers (which made me turn on the record silence option), but my users tell me it's not only those callers, and sometimes those callers do successfully leave messages; I only hear when it doesn't work. What can I do?! I'm stumped, and the situation is intolerable. Thanks! Leah Newmark Capalon VoIP

Hi, I am using Polycom 501 and I came across a problem. As soon as I have incominglimit=1 in sip.conf, which is necessary for buddy watching, I cannot transfer calls. On the console it tells me: Call from user '3052' rejected due to usage limit of 1. Can someone please tell me how to get around this problem? (I don't know if this is relevant, but in the phone.cfg file, I have

...to have a hard time solving it - the procedure is to talk with her ISP, who if they are on a good mood, will talk to *their* upstream ISP about stabilizing the route for you. I have done this before and it worked out for me, but YMMV. -----Original Message----- From: Leah Newmark [mailto:lnewmark@capalon.com] Sent: Wednesday, June 21, 2006 2:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls We've tried 2 different ATAs, the same hardware, same setup as everyone else and both have this problem. This problem is for any number she d...

Hello, > I am setting-up a system to place outgoing calls for a certain > number of minutes (as allowed per the customer's account). I would > like to "break into" the long distance channel to announce "1 minute > left", etc. What asterisk command can I use to do this? > > Thank you in advance for your help. > > Chaya Rosenberg >

I have a very, very simple Asterisk setup in my house. I have a Sipura 3000 with a PSTN line connected and one analog phone connected. The [incoming] context looks like this: exten => s,1,Dial(SIP/50,23,r) exten => s,2,VoiceMail(u50@default) exten => s,3,Playback(vm-goodbye) exten => s,4,Hangup As you can see, when somebody calls in if I don't answer in 23 seconds then they are

I am using Trixbox 1.0(Asterisk 1.2.7.1)at a customer site. They whishes to change the default Voicemail behavior. Standard behavior No answer/Busy -> send to Voicemail Requested behavior No answer/Busy -> message that if you press 9 you will instead be cent to reception -> send to Voicemail or Reception if 9 pressed. I want this to always happen when Voicemail is invoked. How

Hi, I'm having trouble setting the ringtone on my Polycom 501. The relevant entry in extensions.conf is: exten => 801,hint,SIP/creative1 exten => 801,1,SetVar(ALERT_INFO="Test") exten => 801,2,Dial(SIP/creative1,20,Ttr) In the sip.cfg: <alertInfo voIpProt.SIP.alertInfo.1.value="Test" voIpProt.SIP.alertInfo.1.class="13"/> and <TEST

...uld potentially work, but it won't save password changes. Nothing generated from voicemail is showing up in the asterisk logs, nor does the console show any error after changing a password. Any assistance on this strange behavior is much appreciated! Thank you, Leah Newmark VoIP Programmer Capalon Communications

I am having a couple of (unrelated) problems with my polycom 501. 1. The buddy watch is just not working. It tells me that everyone is online, whether or not they are. Here is an example directory entry for one of the peers (whose phone is not registered). <item> <ln>F</ln> <fn>J</fn>

The script runs fine command line. I have edited in the past to try as /usr/bin/php -q and it didn't help. Right now, it's not even reading the changes. I must be missing something very obvious... LN asterisk-users-request at lists.digium.com wrote: > Message: 16 > Date: Wed, 24 Jun 2009 17:17:59 -0400 > From: "Juan E. Rodr?guez" <jerdguez at gmail.com> >