similar to: Zaptel warning

is it possible to register asterisk in a sip proxy as if it were a terminal (like a cisco ATA)? how? Thanx Jenna ;) ___________________________________________________________ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar

OK, i've got asterisk @ home 1.3 up and running with Broadvoice. BUT I have one nagging problem to sort out. When you call my BV # the calling party gets no ring indication, just silence until either I answer the phone, or the call bounces over to voicemail. below is the console output when a call is recieved. what am i missing here? thanks Bernie -- Executing

I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above,

Tie line is type em_w (old school stuff) to TE110P. Phone rings on the asterisk side, but the calling party does not hear the ring through sound. If I pick it up within the first two rings it goes through and I can talk otherwise our old switch drops the call. Anyhow...here is my config if anyone can shed some light on it. It used to work with HEAD a few weeks ago. -Matt

Hi * users, I want to know if STUN suport is available with Asterisk. Kindly let me know. I have posted this also in DEV list but none replied to me. thanks, Somesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050808/e26855c9/attachment.htm

Hello, I am trying to let someone know that is being called from a specified location. For that, the command: exten => _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm)) should let the called person hear Anounce.gsm as soon as he/she answers. (Only calls with prefix 107 are given this notice). The call proceeds fine, but no one hears AnounceSPF.gsm. I tried putting this file in every

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

I just bough a Sipura SPA-2100 to use with Asterisk. When I use the analog handset plugged into the SPA-2100, the person on the other end can hardly hear me. I check the SPA-2100 setup and their is no mic/spk gain control. Is this a problem with the SPA-2100 or with Asterisk? Any way for asterisk to compensate for the poor audio level (if the problem is the SPA-2100)? Thanks, Mike

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date:

I've just gotten off a skype conference call and it pisses me off that the quality of skype is higher than my asterisk calls. Is there such a thing as a super high bandwidth codec? In a situation that you have the bandwidth to share is there something that I can use for important calls when the situation warrants it? TIA, Dean -------------- next part -------------- An

I'm trying to network boot a thin client using pxelinux. I had this working about a year ago but have since reinstalled my OS (went from RH9 to FC2). I'm using atftpd. The thin client gets the dhcp address fine and requests pxelinux.0. Eventually, it just times out. Here's the relevant info from /var/log/messages: Oct 7 23:14:36 home atftpd[4717]: Advanced Trivial FTP server started

Hello, i have two xen hosts servers configured to support migration of domus, but when i try to migrate, the command never ends, i can see on the remote server the machine with xm list but the machine is not accesible, and on the own server i see migrating-{machine_name} and i have access to it. Every server have one nic, with public ip (that is using with xenbr0), and i have a virbr0 configured

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

We have been running into problems here, we have 2 PRI's when they fillup, All channels in use, and we dial more calls asterisk becomes unstable and crashes alot. We are currently on: Asterisk CVS-v1-0-10/26/04-09:35:31 built by root@localhost on a i686 running Linux I know I need to upgrade. Is this a know issue?? Kyle

I'm trying to get this zap dial to work. I want to send DNIS and ANI to other system (ZAP/g2) at answer, while the caller hears ring (RBT). I hear the RBT, but callerid and exten is not sent to other T1 - The ZAP/g2 T1 is standard D4, SF, E&M Wink Start. - At ZAP/g2 wink, asterisk should send DTMF "*ANI*DNIS*" exten => _XXXX,1,NoOp,${CALLERID} exten =>

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How come an outgoing call using my TDM400P immediately say the call is answered? I'd like to be able to detect when the call is actually picked up, is this possible? If this is normal with analog cards, does the same thing happen with T1 cards? -L ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football

hello i want to use SIPGetHeader application in asterisk-1.0.9. Jul 2 00:04:33 WARNING[19575]: pbx.c:1293 pbx_extension_helper: No application 'SIPGetHeader' for extension (default, 2000, 1) Any one using this __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hello. How can I check if the RTP traffic between two channels is bypassed? Some * console command? Thanks.

Hello All, I have an iaxy(new version), and while it does the job well, there is one thing I am looking for. I want to be able to specify a dns name on the config, not an ip. This does not seem to work if I try to set it as such. Has anyone come up with a workaround or solution to this? I want to be able to put it on the net, or travel with it, plug and go. The issue is that I am using a