search for: rilawich

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 all, I will a TDM card with FXO modules on it. Below is the dial plan. When someone can 9123456, CLI will show dialing to 123456 and answered. After answered, the call hangup. I would like to know what will cause the case to happen. Anyone can give me some advice to solve it? exten => _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT}) exten => _9X.,n,Hangup zapata.conf

Hi, Can I play a prompt after hanging up a call? I have tried below but failed. ... exten => s,n,Dial(SIP/1234) ... exten => h,1,Playback(demo-instruct) -- Executing [h at macro-safedial:2] Playback("SIP/3601-09856bf0", "demo-instruct") in new stack [Aug 14 17:24:03] WARNING[2496]: file.c:738 ast_readaudio_callback: Failed to write frame --

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, 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

Any command to refresh or clear the whole ast database?

Recently, I got the following error messages in CLI periodically. Jan 20 17:43:18 ERROR[8641]: chan_sip.c:11002 handle_request_subscribe: Got SUBSCRIBE for extension XXXXXXXXXXX@from-int from 192.168.0.123, but there is no hint for that extension I have no idea what the error message tell me. I am sure I haven't that account XXXXXXXXXXX in my database and there is no hint extensions in dial

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, Recently, I got the following message from CLI and finally the asterisk will hang. Anyone can tell me how to fix the problem or why it will happen. Thanks. Jun 17 14:18:02 DEBUG[24573] channel.c: Avoiding initial deadlock for 'SIP/1127-008d65f0' Jun 17 14:22:45 ERROR[24696]: chan_sip.c:11337 sipsock_read: We could NOT get the channel lock for SIP/1589-0087cdd0! Jun 17

...In the case, I just wonder why we need to give up the app_realtime. Or we have another alternative to get a value easily and faster using func_realtime. Anyone using func_realtime can give me some advices? On 8/13/07, James FitzGibbon <james.fitzgibbon at gmail.com> wrote: > On 8/13/07, Rilawich Ango <maillisting at gmail.com> wrote: > > However, I found the the function will be depreciated in 1.4. There > > is a replacement using application REALTIME. I found that it is very > > troublesome to use it. > > I'm afraid I don't have any advice with rega...

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 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 all, I want to have a 16 FXO in a PC. Is it possible to use 4 x TDM404 or 2 TDM808 to get 16 FXO? What is the difference (in performance and control) in using 4 x TDM404 and 2 x TDM808 if possible? ango

Hi, We are using attended call transfer to transfer the call. In the direct call, the quality of the voice and dtmf are acceptable. After transfer, the quality becomes worst. Voice can't be heard clearly and dtmf wrong detection will occur sometime. I wonder call transfer will affect he quality of the call. Anyone has same experience? Anything to do in asterisk level can get a better

I have a TDM400 with all FXO module in it. Only one channel (say channel 3) is plugged to PSTN. In my understand, a dial command Dial(zap/g1/12345677) should search an available channel, which is 3, in group 1 to make a call. However, I found that it will still use channel 1 to make call even it hasn't plugged to the PSTN. Below are the conf files. --zapata.conf-- group=1 signalling=fxs_ks

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