similar to: some questions about atxfer usage

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. --

Hi List, I need to modify the callerID name of the call coming back when a parked call returns to the extension that parked it when it times out. Looking at app_parkandannounce.c /* Now place the call to the extention */ snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars =

Hello. I have a MinITX motherboard with only one pci slot and one onboard ethernet interface, I have a TDM04B card plugged into that motherboard and the /proc/interrupts: CPU0 0: 169626332 XT-PIC timer 1: 1270 XT-PIC i8042 2: 0 XT-PIC cascade 8: 4 XT-PIC rtc 12: 170166219 XT-PIC eth0, wctdm 14:

Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and "b" A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see

Hi, I'm testing attended transfer with 3 SIP phones. I noticed about 10% of my transfers make the call drop and I get this on my log: Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback: Failed to write frame -- Playing 'beep' (language 'it') Jul 5 10:42:32 WARNING[23960]: res_features.c:745 builtin_atxfer: Failed to play transfer sound! Moreover, every

hi, all I have test echo application for just fun. I can'nt understand why this is used below in .c file, format = ast_best_codec(chan->nativeformats); ast_set_write_format(chan, format); ast_set_read_format(chan, format); without this this application work fine. then why this is used. any suggestion?? Bhrugu mehta

the gdb of the core taken from the asterisk as the time of crash is as below I run asterisk-1.2.5 on fedora core 3 with chan_ss7 can someone help out? #0 ast_var_name (var=0x1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46

Hi, is it possible to remove the "no timeout" combo box in flash operator panel? How can I reduce the flash area? I set small buttons and half of the area is white and I want to resize it. TIA Giorgio Incantalupo

Hi all. I'm using a te110p in spain. ;zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 I'm getting problems dialing out through this span. ?How can I debug its behaviour? Thank you in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060404/70d2cb7f/attachment.htm

I released app_changrab.c lastnight really late... It includes a way to hijack a channel and originate calls from the CLI. /b --- Keep Your Friends Close, But Your Enemies Even Closer...

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to

Hi, I'm facing the following request : "When someone is starting an assisted transfer using Asterisk's features codes, he will ear a prompt saying "Transfer" and then a dialing inviting him to dial the number he tries to reach. This tone volume is qualified as a bit too load." Is it possible to change that and have a more delicate volume ? A quick look inside

Hi, We are testing new Asterisk 1.6.0.1 because we would like to use the Attended Transfer feature and we are trying to use the new action Atxfer developed for AMI. As far as we know, it is suposed to be in this release as it can be read in Digium's changelog /New command: Atxfer. See doc/manager_1_1.txt for more details or manager show command Atxfer from the CLI/ But, when we try to

Hi there i just can't find an answer on the featuremap config i want all phones to use the same method for transfering a call on all phones but i just can't get the atxfer or other functions to work on my grandsteam and sipura spa 2000 it's confusing for users with different phones to transfer a call i know you can use the transfer button but i wan't to use a code *1 not

Hi, We are having a problem that is preventing users from using *2 to manage an attended transfer. After dialling *2, asterisk places the call on hold, but you can only dial one digit before it times out, and the cli says:- [2011-06-29 18:33:55] WARNING[29877]: res_features.c:938 builtin_atxfer: Did not read data. There is already an issue in JIRA:

I would like to know if atxfer is supported somehow because there seems to be little documentation for this feature. I know most people expect a good SIP/IAX phone to do the job but I think it's nice to be able to do attended trasnfers with a simple ATA-connected analog phone. I have Asterisk 1.2/Freepbx and features.conf has a line regarding atxfer and I set it to *2 (Default). While # works

Hi everyone, I'm trying the new atxfer functionality. All seems to work fine at the beginning, but there is no audio between the party at the end of the transfer. Plus, after that, even normal calls won't work well (they can't hangup!). I'm using the Asterisk CVS from 2005-01-10 with Asterisk-OH323. Here is my dialplan: [default] exten => h,1,NoOp(bye) exten =>

Hi, I successfully installed asterisk 1.0 with Capi 0.35. In my pbx system I would like to use the atxfer function but is not included in the stable asterisk. Is there a way to include it in my version of asterisk: I did no used the last cvs because I can't compile the chan_capi .in it. :( Bye

Hi. I've been trying to develop this module since some time now. CVS already has a dial version with atxfer. When trying this, using the modifiers tT and having configures features.conf accordingly, i havent been able to use such a feature in the called party. I also tried using t and T separately. I've tried to understand why this happens, and started to watch the "copy" of

Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. In usual pbx's normaly there is no difference between an attended call transfer and a blind one: you just hit "transfer" then dial the extension you want the call to be transfered. If you stay on the phone you can talk to the other party, then, when you hangup, he will get the call. If you hang