similar to: No translator path exists for channel type dahdi

Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691918892|30|m") in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to create

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, 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, 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 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

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, 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 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

I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below I go off hook

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

Thanks. In app_realtime, it is very easy to get a value of a field by only applying the realtime application. However, in func_realtime, we need to get the key-value pair according to the position of it by using function CUT. After that, we need to apply another CUT to get the value. It will cause the following problems. http://www.voip-info.org/wiki/index.php?page=Asterisk+func+realtime 1.

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

ONLY ON MONDAY! Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?

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

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