similar to: application order when you make a call

Hi A few day ago, I notice that some applications missed in asterisk 1.6.1 release even if *.so file which normally create them were compiled during Asterisk install. SetCallerID(), SetCIDNum(), SetCIDName(), SetLanguage() ... and maybe so more. anyone already notice that to ? If it's not normal, anyone have an solution to it ? -------------- next part -------------- An HTML attachment was

Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ?

Hi to all. I'm trying to use meetme on asterisk 1.4.22.1. On a debian i've compiled (as i need h323 support) openh323_v1_18_0 pwlib_v1_10_0 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-1.4.22.1 All works fine, dahdi status is: asterik:/data/programmi# /etc/init.d/dahdi status ### Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: RTC) 1" (MASTER) asterik:/data/programmi#

Hi all, Our users have started to complain that some of the time they're unable to save Word documents to our Samba drive - Word tells them the disk is full. I had a look at the logs, and there are a lot of weird "Function not implemented" errors. These have been there for a while, but the "Operation not permitted" ones seem new. Nothing on the server has changed.

I'm a home user of CentOS (a desktop and a laptop) because I like the quality of the product and I love the cost. I'm a great believer in FOSS. That being said, I know that some of you use CentOS for production servers and (perhaps?) desktops as well. I've read at leat one comment like "I have 20 CentOS servers." My employer is a firm believer in RHEL - license costs are

Hi List; Where I determine the codec to be used for the SIP Trunk (between Asterik and another SIP softswitch)? Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000 at mydomain to call 9000 at my domain(which is the call center)

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

Hello everyone. I have the following problem that could help would appreciate. 1) Environment: HOST 1: xen-hypervisor-3.2-1-amd64 with all VM in LVM disk. HOST 2: XCP 0.5 2) The VM that I wish migrate is a domU in XEN and is debian lenny over LVM: ................................................................................................... # Configuration file for the Xen instance

Hi All, Have any way to asterisk forward the 487 Request Cancelled in SIP TO SIP call? In a SIP to SIP call when the called peer B send 487 to Asterik, Asterisk return to calling peer A 603 PEER A ASTERISK PEER B | INVITE ------------>| | |<------------TRYING| | |

Hi there, I need help in implementing Asterisk with LDAP. I' ve installed Asterik 1.4 with CentOS 5.2 and I would like to use with it an existing zimbra LPAD. thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081018/5f725eea/attachment.htm

Hi, folks, Well, it's ten years that I've been on this list, right when I started this job. But, it's time to move on... I'm retiring. (Yeah, that old.) So, though I'll be part time for a few months, and running CentOS at home (in spite of my manager's pushing me to do Ubuntu). This list is *so* much more useful than any of the ones I've seen for Ubuntu, or much

Hi people! Sorry for bringing up some annoying issue.. yes, it's AMD again... But I was searching the last days for a solution for my problem and didn't really find anything. Now I'm hoping that someone of you has maybe an idea for me. :) My setup: --------- I use the Asterik Manager API to generate outgoing calls (by using "Originate" messages). These outgoing calls

Hi I have a new question. Here the situation : I use softphone on 2 computers (soft1 and soft2) located on the same subnetwork. When I register on asterisk server using soft1 with one user (e.g JOHN) which I declared in sip.conf I can register again with this same user using soft2. Is it normal ? I notice that I can pass some call from both but incoming call for JOHN user only arrive on the last

Hi, I have to connect 3 asterisk servers,each of them being TLS server for his clients and connected in both way in TLS with both others asterisk, each having hi own Common Name. Is this possible? I set up 2 asterik's , one server and the other client, this is OK. But I can't deal with certificats generated on both servers. I tried to put tlscertfile ans tlscafile in the peer

Possible Caching Bug showing up as a MIME Boundary Issue I'm using Dovecot version 1.0.0. I was using Dovecot version 1.0.0 beta3 or alpha4. I upgraded to Dovecot 1.0.0 to make sure that was not the issue. Over the past few weeks on a server running a stable dovecot, I have seem a few emails arriving where the MIME document structure dividers are visible. I've included a

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is

Hello I have few questions : - what's the difference between a subscribe request et a register request ? - in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please someone could explain how doest it work because I think i'm a little bit confuse. - if I configure a sip terminal in sip.conf like this [john] type=friend username=JOHN secret=mypassword host=dynamic

Dear Colleague, I am writing to ask your help for a book I am writing about Life Practices. I am looking for examples of those simple things that we all do that are essential to expressing and maintaining our values as we face life's ups and downs - our Life Practices. What I am seeking is a bit like folklore. Something with a story and an easy to remember slogan or catchphrase. I am asking

Hi all, Eyebeam has a sip-chat function and it would be nice if I would be able to use it. But the problem is that I can't really find information about it. I can just try to send a message and on the Asterisk console a message like this appears: Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message: Received message to <sip:bla@voiphost> from "Bla