similar to: call to a particular 800 number never showsanswered on Zap channel

Whenever we call IBM, the call counter on the phone never starts and in the CLI the zap channel never gets the answered signal from the PRI. See below. -- Executing Dial("SIP/5933-645d", "Zap/g1/18004267378") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/18004267378 At this point, I am in IBM's menu system. However the call never

> Watch the output of 'pri debug span 1' on the Asterisk server while > placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) > might be relevant. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

Watch the output of 'pri debug span 1' on the Asterisk server while placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) might be relevant. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Andy Goss > Sent: Monday, October 10, 2005 5:58 AM > To: Asterisk Users

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on

I can not figure out what the heck is going on. I went back to my old version and I still get errors when the voicemail system tries to load any of the greetings, unavail messages, etc. the normal voicemail prompts work, but any user recording don't work. Leaving a new message appears to work, but the system wont replay them, it throws errors. Here is an example of the errors: Oct 11

If anyone could tell me what this error is all about, I would be very grateful. Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not permitted Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216

I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is possible to customize the email message sent to the pager email address. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 agoss@ntad.com

I need to find a way to have the Polycom phones automatically park calls. Right now my users hit #70# (I know the last # is optional but it speeds it up.) to park a call. Personally I think this is easy, but my users would like one button to do this for them. The Polycom has buttons we don't need (Transfer & Services), it would be nice if I could remap one of those buttons to dial

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?

Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is

Hi all! I've got a machine setup to be an "RSYNC Server", i.e. running rsync in daemon mode waiting for connections from various other machines on my network. This machine is running Debian Sarge and rsync 2.6.3. For the past several days, I've been getting notices like this in my backup logs: ============================================================================ =====

Question for all: Right now i have a Table in a mySQL DB that has a row called Ingredients. When the data is entered into the DB its enter like so from a text area: 1 1/2 lbs. beef top sirloin, thinly sliced 1/3 cup white sugar 1/3 cup rice wine vinegar 2 tablespoons frozen OJ concentrate 1 teaspoon salt 1 tablespoon soy sauce 1 cup long grain rice 2 cups water 1/4 cup cornstarch 2 teaspoons

Hi all, I am new to R. I downloaded and installed R 2.15.1 I tried typing R.exe --help at the DOS Command line C:\", but I keep receiving: [quote] R.exe is not recognized as an internal or external command [/quote] I tried many variations of R.exe --help, but roughly the same response Any ideas? thx w -- View this message in context:

After months of BroadVoice ignoring my trouble tickets for dropped calls, delayed termination, etc., I'm throwing in the towel. While they have credited $19.95 to my account, they refuse to credit anything more, despite ALL of the problems I've had. I feel the least they could do is credit the remaining $8.61 to my account, yet they won't. I haven't really been following up on

Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex

Hi, When is start one vpn i get the following result: tinc10703003005 Link encap:Ethernet HWaddr C2:F7:7B:75:47:1A inet addr:192.168.3.20 Bcast:192.168.3.255 Mask:255.255.255.0 inet6 addr: fe80::c0f7:7bff:fe75:471a/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX

Hello All, Right now I have several providers. Voipjet, Teliax, and more recently Broadvoice. Broadvoice gives me unlimited to europe, but what I want to do is determine the best way to setup a dialplan so for example, certain countries will go through the cheapest route. I am really only interested in Poland, Russia and Turkey. Poland is free on broadvoice, but not for cellular, which I may

Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond....) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line

I just signed up for Broadvoice, and used a similar network configuration that I have on stanaphone, voipjet, and others. My asterisk box is behind a vanilla Linux masquerade (netfilter/ipchains) firewall. The SIP and IAX services have been working fine in both directions for the other SIP termination services. The Broadvoice inbound service worked immediately. (which to me is odd, inbound