similar to: quality after call transfer

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, 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, 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, I wonder how I can relate the CDR records for the case of call transfer. I can't find their relationship in CDR. Any can advice? ango

Hi all, I have a GXP2000 with BLF configured. I follow the configuration guide to enable the pickup cmd as follow and include it under corresponding content. [BLF_group_pickup] exten => _**1XX,1,Pickup(${EXTEN:2}) exten => _**1XX,n,Hangup The I press the single key to pickup the call to extension 100 when there is a call to it. From CLI, I can see it issue **100 to asterisk but failed

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, 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 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, I wonder who has the same voice quality problem as what we have. Below is our configuration. Company --- asterisk 1.4.22 (g729) --- CISCO --- T1 --- customer Sometimes, customers told me that they heard our voice not very clear, like a call from far far away. I heard the recording is ok and there is no such effect in it. Can I assume the following? -voice quality is ok in asterisk as

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 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 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

HI all, I got segfault in the system log that make asterisk crash. I still have no idea what cause this segfault. Is it a bug? Anyone has experience about it? phsip01 kernel: asterisk[3412]: segfault at 00002aabd10f2b40 rip 00000037e806ea75 rsp 0000000041d3cc70 error 6 version: asterisk1.4.12.1 usage: in/out bound call, queue, ivr, attended 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 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

HI, I want to allow user to press 0 to the voicemail if the user don't want to wait in the queue. Below is what I set but it doesn't work. Anyone can help? ango file: features.conf [applicationmap] opervm => 0,self/both,Macro,opervm file: extensions.conf ... exten => 5555,n(queue),Set(DYNAMIC_FEATURES=opervm) exten => 5555,n,Queue(5555|tThH|||180) ... [macro-opervm] exten

Hi, I have 3 queue set in the table as below. name,autopause 1000,1 2000,1 3000,1 In queue 1000, the autopause works after member failed to answer call. However, other queues don't work for the autopause function. queue 1000: -- Nobody picked up in 25000 ms -- Auto-Pausing Queue Member SIP/1234 in queue 1000 since they failed to answer. queue 2000/3000: -- Nobody picked up in