similar to: Inform callers on recorded/monitored number.

Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten => _7., .... How do I get Asterisk to wait until the user is finished dialing instead of trying as soon as it gets the second digit? I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like to be able to dial others... Same problem for outside

Greetings all, My next step is to purchase a nice VoIP phone for my desk. I have a grandstream, and the sound is great, but I'm looking for more of an office style phone, preferably that can handle multiple lines, has a more flexible display (i.e. name as well as number). SIP would be preferable. Any suggestions? Thanks, Eric

Hello, i have a simple but annoying problem. I have the following entry in /etc/asterisk/externsions.conf file: ---<Cut Here>--- exten => 10100,1,Wait(4) exten => 10100,2,Playback(transfer,noanswer) exten => 10100,3,Dial(${PHONE30},30,t) exten => 10100,4,Background(extension) exten => 10100,5,Background(is-curntly-unavail) exten => 10100,6,Voicemail(9999) exten =>

Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend

Asterisk Project Security Advisory - AST-2011-001 Product Asterisk Summary Stack buffer overflow in SIP channel driver Nature of Advisory Exploitable Stack Buffer Overflow Susceptibility Remote Authenticated Sessions Severity Moderate

Hi, I''m working on a french website and I use the Form.serialize method to send the info through AJAX. The thing is that the accentuated letters (é,ê,à, etc.) don''t get replaced by their HTML entities and they get corrupted when retrieving the data. How could I fix that? thanks a lot, Blaise Bernie

Hi Everyone, I am currently running Trixbox 2.6 and I have a problem with Asterisk. /etc/init.d/asterisk status Asterisk dead but subsys locked I deleted all files in /var/run/asterisk folder and asterisk restart... It's ok for a while. But some days after Asterisk again is dead. Can anybody help me? Rgs / budacsik

Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric jacksch@tenebris.ca

I'm still trying to learn some parts of Asterisk, so sorry in advance for the dumb question! How do I set up an extension to dial out to the PSTN through my ZAP interfaces? I want the ability to have a ring group that will ring all of the phones in an office and then ring cell phones if nobody answers. I'm sure this is simple to do but I'm at a loss. I have tried the following

Greetings all, I've tried out chanspy, but what I'm really looking for is the ability to interrupt a call (i.e. barge in for emergency purposes). Has anyone found a way to do that with Asterisk? Regards, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060403/dfe6e99e/attachment.htm

Greetings all, I've been working on having Asterisk put a call through to two different numbers, and give the call to the first one that acknowledges by pressing the 1 key. I found an example on the wiki, but I can't get it working. When I answer the call I hear the message telling me to press 1 to connect, and as soon as the message is done, the call is connected. In other words, it

Hi! I need some help Well i have this cenario: 1 ip04 running asterisk [A] 1 pc running asterisk [B] I nedd to make calls from A to B, and B to A. Via sip The A-B calls are working. Now I need to configure the dial plan to call B-A either to sip numbers and Fxs. Anyone can help me? -------------- next part -------------- An HTML attachment was scrubbed... URL:

The problem I am running into is that when the call to my cellphone is made, it appears as though the call "completes" so it never rolls to asterisk voicemail. Here is my current config: exten => 102,1,Dial(${sipura},10,) exten => 102,n,playback(pls-wait-connect-call) exten => 102,n,Dial(IAX2/asterisk1/9139275900,10,r) exten => 102,n,VoiceMail(u102@default) exten =>

Hi, Let's say I do: Set(data=xxx,yyy) Gosub(my-sub,s,1(${data})) My subroutine will only receive "xxx" for ARG1. How can I pass a literal with a comma to a single argument in a subroutine? (The point is: when calling the subroutine I do not know if the variable has a comma or not.) Thanks, Daniel -------------- next part -------------- An HTML attachment was scrubbed... URL:

How do you enable call waiting on sip phones? Ive looked and googled and can only find call waiting pstn phones butnot for sip. Is their a way of setting this up within the dailplan?

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

Perhaps this will help... I have a phone connected to a QuickNet PhoneJack card. When I pick it up, I get a dial tone. When I dial a certain number of digits, the call is processed by Asterisk. The question: How does Asterisk determine how many numbers to let me dial? I'm banging my head against the desk here... _9XXXXXXX lets me make an outbound call, but _9X. only lets me dial 9 plus

I'm using a T100p to interface to a channel bank and from there to analog PSTN lines. Because of my particular setup I have to do post-connect inband DTMF dialing - which takes up to 5 seconds for a 10 digit number, assuming 0.5/sec per digit (ie. using "zap/g1/31|5|D(6045551212)". Even with an 'outside transfer' voice prompt before commencing dialing my users are getting

Hello All ! Recently I have upgraded NUT to 2.4.1, and now I can't monitor old IMV LanPro S3 UPS with old SNMP board. Previously it is monitored OK, but now snmp-ups refuses to start: [root at perforce /usr/local/etc/nut]# /usr/local/libexec/nut/snmp-ups -a lanproups -DDDDDDD Network UPS Tools - Generic SNMP UPS driver 0.44 (2.4.1) debug level is '7' SNMP UPS driver : entering

I sometimes need to allow sub-contracted admins root ssh access to my servers. Later, I always wonder what they did during access. Is there any shell that provides all shell abilities to the remote user but monitors/emails a designated user each command executed in the shell terminal and does not allow the user (even root) to modify the bash history file or similar shell history file, or maybe