similar to: Events when the target of the call answer

why can't we compile the asterisk coading in windows, it's done in c++ so it should work in windows as well

Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus

Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with

What are the alternatives to FWD with IAX2 registration capability. FWD is great, but their IAX2 is not the priority and if it goes down it takes days to restore it. I want to use IAX2 protocol but the end point (Sipura unit) need to be able to register over SIP behind firewall. Line1 is registered with FWD PSTN need to be registered with somebody else. What are my alternatives? -- #Joseph

Hi Fran, you could do it using Adobe/Macromedia Flash Media Server 2, but I guess that's not the answer you are looking for. If you manage to do this and release it under GPL I'll kick in $50 for a bounty. Regards, Dean Collins dean@collins.net.pr +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on "Accept" button. Greg -------------- next part -------------- An

Hello everyone. I'm having a problem with convert-tool while trying to convert mbox folders to maildir. I'm running dovecot-1.1.16 on Centos 5.3 If the user has subfolders, convert-tool fails saying: "Mailbox conversion: Couldn't create mailbox sub/folder3: Invalid mailbox name" However, if I ran the convert-tool with the same arguments and same folders with dovecot

Hi folks! I'm having some trouble replicating a domain. I followed the steps in this guide: https://wiki.samba.org/index.php/Samba4/HOWTO/Join_a_domain_as_a_DC I followed the steps and everything worked fine for me. but... If I make changes on the PDC, these changes are replicated to the Secondary DC, but if I make changes to the Secondary, these are not seen in the PDC. Is there any way

Hi everyone, I''m Ezequiel from Buenos Aires. I''m new in this list, so please forgive me if my questions are not the correct one''s fro this alias. I''m working as a campus ambassador in Sun, while finishing my engineer degree, creating communities around opensolaris in Argentina. I discovered this project, and this group, looking for something that I could

This patch adds support for MSG_PEEK. In such a case, packets are not removed from the rx_queue and credit updates are not sent. Signed-off-by: Matias Ezequiel Vara Larsen <matiasevara at gmail.com> --- net/vmw_vsock/virtio_transport_common.c | 50 +++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 3 deletions(-) diff --git a/net/vmw_vsock/virtio_transport_common.c

This patch adds support for MSG_PEEK. In such a case, packets are not removed from the rx_queue and credit updates are not sent. Signed-off-by: Matias Ezequiel Vara Larsen <matiasevara at gmail.com> --- net/vmw_vsock/virtio_transport_common.c | 50 +++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 3 deletions(-) diff --git a/net/vmw_vsock/virtio_transport_common.c

Hi All, Asterisk 1.4.12 on CentOS 5 I'm trying to increment an AstDB key with the length of the last outgoing call. Here's what I've got for "01" UK geographical numbers: exten => _01.,1,Dial(${UKGeographical}/${EXTEN},,g) exten => _01.,n,Log(NOTICE,Call to ${EXTEN} lasted ${DIALEDTIME}) exten => _01.,n,Set(CALLTIME=${DIALEDTIME}) exten =>

My system is: Redhat 9.0 + Asterisk + ISDN4Linux + Teles 16.3 ISA Passive card I haven't sound card. Comunication between two SIP Clients is OK Comunication between PSTN and SIP Client is OneWay (i cant recive dtmf and voice from pstn) is it needed sound card ?

I have installed on a VirtualBox vm the crossbow ISO Now I''m trying to create the network map described in the "getting started" pdf I have this situation: On global zone, I have the real NIC: e1000g0 with ip 10.0.2.15 (VirtualBox NAT this IP, I can get web access with it) I have created an etherstub (etherstub n?2) with 2 vnics (vnic5 and vnic6), I''ve plumbed

Hi people, I have a print server using Debian, LPRNG and SAMBA working without any problems. Now I 'm trying to set up a new server with CentOS 6, CUPS and SAMBA. It 's a very simple setup: a few local users and a few printers, nothing special. The clients also map the printer using 'net use lpt1 \\mainserver\printerX' because they need to print over an old DOS application.

Hello everybody, I would like to implement "callback" function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

The best place for Astri Con 2006 would definatly be Omaha, Nebraska! ;) very central ...ah one could hope. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

Hi friends, I am using Real Time Asterisk Architecture where I have put the Sip users/peers and extensions defining the dialplan in tables in a mysql database. Currently, asterisk points to my single database server as configured: ------------------------------------------ /etc/asterisk/res_mysql.conf ------------------------------------------ [general] dbhost = xxx dbname =

Hi! I can make ldap authentication to work using LDAPS in CentOS 6. On CentOS 5, I just simply set tls_cheekpeer no to /etc/ldap.conf and it works! I tried all /etc/nslcd.conf /etc/pam_ldap.conf /etc/openldap/ldap.conf It 's really confusing on CentOS 6. Why so many files???? CentOS 5 LDAPTLS_REQCERT=never ldapsearch -x -H ldaps://xxxx (works!) CentOS 6 LDAPTLS_REQCERT=never ldapsearch -x