similar to: Only Accept Call After Pressing a Key '#' or '*'

I'm using Asterisk to registry several DDI's to a sip proxy (pipecall.com). Everything works fine apart from several times a day my firewall (zywall70) reports a UDP port scan attack from the pipecall sip proxy. I can't seem to work out why this should be. All I could think was that the sip registry was expiring and causing some strange probing from the proxy, is it possible to alter

I've configured voicemail.conf to delete voice mails after they are emailed. The email work ok but the message don't delete. The config is as follows: [default] 3000 >= 1111,CS,cs@x.com,,delete=yes Thanks, Chris

Hi, I'd like to add a feature to my asterisk system that tries to find a user among a couple of locations, and then goes to internal voicemail if the user doesn't pick up. (e,g, an internal extension and a cell phone). The catch is that I want the user to manually accept the call to prevent it from going (for example) to the voice mail on my cell phone. Scenario * Call comes in,

Hello all, I am sure that this is possible - for helpdesk envioroments i.e. when you hear an announcment "Your call may be recorded for quality purposes" can asterisk record all calls onto disk or similar - hopefully as MP3? Thanks Nick

Is it possible to park a call through the manager interface? If yes; how? Regards Thorben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050318/ec2a5f90/attachment.htm

Hey everyone, I thought I''d seen a demo of an AJAX app that responded to control key presses, like performing a different action when the shift key is pressed. What I''d really like to do is have a div change when the shift key is pressed (to change the text of the link inside it to indicate what new action will occur when that link is clicked with shift held down). Is

I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227

Hi, if I dial remote user (over CAPI) and transfer it to conference, if it presses * or #, asterisk crashes with following core trace... This works OK with internal extensions .... bt full 0 0x0805dae6 in ast_queue_frame (chan=0x819b738, fin=0x40d99660) at channel.c:384 f = (struct ast_frame *) 0x810d7b8 prev = (struct ast_frame *) 0x0 cur = Variable "cur"

Hi I have setup forwarding - xfering - where you press # and then the extension. I add t to the dial cmd. My problem is that when you call something like internet banking they want #, but when # is pressed asterisk gets it instead. is there a way around this ? I haven't been able to get asterisk to listen to flash either Alex -- Why is it there are so many more horses' asses than

One of the features that I thought would be popular with the Web-MeetMe suite is the ability to start all non-admin callers in a muted state and selectively unmute them. For example any large conference that is of an announcment nature with a Q&A session. It's probably a feature I should have tested better, but I just discovered that a caller that is joined to a MeetMe with the |m flag

First let me say I'm really excited about the possibility of dynamic menu creation over http using php. When this is stable it will make maintinging some of our pxelinux infrastructure much easier. Id like to thank all involved; I'm really looking forward to the gpxe intigration. After hpa's announment for 3.70pre7 I begain testing. I dropped the new gpxelinux.0 and new c32 modules

Ok here is one for you. I know we all do the this for 911: exten => _911,1,Dial(Zap/1/911) exten => _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a "least cost" routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is

I see LAM-related announcment here, so i hope i'm not too much offtopic. I use LAM (LDAP Account manager) by many years, previously on OpenLDAP/Samba ''NT mode'' domains, now on AD mode. LAM works as expected in AD mode, but i've noted that i cannot use the ''tree view'' apart pointing it to some sub-OU. EG, if i set as base dn my domain base dn:

Hello, I am using macros to give the ability to a call-receiver to 'accept' a call. However, any keypress connects the caller. Anyone have any suggestions about how to re-engineer this so that the receiver can deny the call, or press other keys to do other actions, without connecting to the user? Thanks, Jason Wolfe

Apologies for resending this, I am not sure it made it through the first time.. --- Included below are my config files: sendmail.mc, dovecot.m4, dovecot.conf, dovecot-sql.conf. Whenever I try to send email into this system, it rejects it with user does not exist, however, the user does exist as far as dovecot (imap) is concerned, it is in the database. THANKS - Yossie # for i in

I'd like for our custom soft phone to be able to know what queue, and/or what DID is calling an Agent's phone before the agent picks up. The agent is using the AGENTCALLBACKLOGIN. One agent can be in multiple queues so it would be nice if they could get a pop up window telling them who's on the line before they pick up and hear the announcement telling them that. I'd like

Included below are my config files: sendmail.mc, dovecot.m4, dovecot.conf, dovecot-sql.conf. Whenever I try to send email into this system, it rejects it with user does not exist, however, the user does exist as far as dovecot (imap) is concerned, it is in the database. THANKS - Yossie # for i in /etc/dovecot/dovecot.conf /etc/dovecot/dovecot-sql.conf /

I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message

When using the callback feature on agents I notice that when the queue calls one of the agents and the agent picks up the call they hear nothing until pressing the # to accept the call. Only then does my announcement play back to the agent after which the call is immediately connected. Is there a way to have the announcement played to the agent before they press # to accept. I have ackcall=yes

I am looking for a way to Forward to a external or internal number and require a digit(s) in order to complete forward. Example: PSTN1 Calls * dials PSTN2 if PSTN2 presses proper digits bridge the PSTN1 and PSTN2 if no response return to a context Reasons: 2 actually 1. call is forwarded to cell phone but If cell is out of range, turned off,