similar to: call transfer in CDR

Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? ango

HI, Does asterisk will ignore the setting in files if realtime is applied, say asterisk will ignore all the setting in sip.conf if realtime table sip_buddies is applied? ango

Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango

Hi all, I have a TDM400 with all FXO on it. When I make an outgoing call, I can hear callee but callee claims the volume is too low so that he/she can't hear very clear. Can I adjust to increase the volume in callee side? Is it increase the value of txgain can solve the problem? ango

Hi all, I am looking for a prepaid application. I found that there are many applications in the page http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications. Anyone recommendation among them? ango

In CDR, I found that there are 3 records after doing call transfer. However, 3 of them are individual record that is very difficult to identify they are related to call transfer. My question is how to identify the call with a clear flow, from CDR or by other means, is a call transfer.

Hi, I have an intel mother board dg43b (http://support.intel.com/Products/Desktop/Motherboards/DG43NB/DG43NB-overview.htm) which have an on board lan interface. In default, it can't be activated after CentOS5.2 installed. I can't find any driver or information about how to activate this interface. Can anyone tell me how to work it out? Thanks, ango

Hi all, I found that there will be a memory leak if asterisk running day by day without restart. Is it good to restart asterisk service daily? What is the better way to restart it daily like apache? ango

Hi all, I have installed Centos completely. However, the LAN doesn't work. Below is the message after I issue. How can I make it work? 00:19.0 Ethernet controller: Intel Corporation 82567V-2 Gigabit Network Connection Thanks!

Hi all, I have read many forums and discussion groups talking about fax support in asterisk. Some of them conclude that asterisk doesn't support fax. However, some of them conclude that there is no relationship between fax and asterisk as asterisk will only pass the fax signal to the fax machine. I have tried the fax in asterisk before but failed. Anyone can give me some guideline how to

Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on

hi, I have 2 asterisks with the following configuration. asterisk server 1 (S1) has an user 9002 asterisk server 2 (S2) has an user 9003 Both users can make call to each other without problem. Now I add both users to both servers, i.e. asterisk server 1 (S1) has users 9002,9003 asterisk server 2 (S2) has users 9002,9003 When 9002 dials 9003, Dial(SIP/9003@S2) or visa versa. Both processes

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

I have a TDM400 in the server. I want to press **1XX to pickup a call. It is ok if I pickup a call dialled from 1XX to 1YY (internal network call). However, it is failed to pick up a call from PSTN thro' TDM400 card. It seems I can't guess the correct context of it. How can I know the context of the call using CLI? The default context of the TDM400 is from-pstn but pickup still

HI, Recently, I found that asterisk fail to get the correct context of the sip phone. Below is the configuration and the log message. In the log message, asterisk fail to identify the calling party. As a result, it use a default context instead of int. Anyone know why and how to fix it? (testing environment) asterisk 1.4.22 & 1.4.24 asterisk-addon-1.4.7 Setting name=123 context=int

Hi, Feature originate can be used make call thro' the web. There is a parameter ,Async, in it. I set it to true but there is no effect. Actually, I want to do the following. What I know the function originate is: originate call ---> party A party A rings party A answers call party B rings, party A still hear ring party B answers and A & B connected. party A will feel weird when she

HI, I have 2 TDM400s plugged in a PC. I failed to use same channels to make a call to PSTN. It shows it can't establish connection after dial command issued. Below is the log. Actually, the call is established as I can hear voice from the called party but the softphone is still showing ringing. It seems the TDM card can't get an answered signal from PSTN. After 15 seconds, the call

Hi all, I got the following message in the log yesterday. After that, no more in/out bound call can be made. What is the meaning of the message? ango [Mar 12 09:26:17] ERROR[29565] chan_sip.c: We could NOT get the channel lock for SIP/2367-d8062fb0! [Mar 12 09:26:17] ERROR[29565] chan_sip.c: SIP transaction failed: 5999e928603c878945d3e7811d2393e8 at 210.14.27.50 [Mar 12 09:33:15] ERROR[29565]

HI, We are experiencing the hum noise when the conversion of 2 parties is established. How can we eliminate that noise? ango

Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango