[asterisk-users] no ringing tone making attended transfer whith an IAX client
Hi I have configured attended transfer in features.conf like this [general] parkext = 70 ; What ext. to dial to park parkpos = 00-99; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 300 ; Number of seconds a call can be parked for (default is 45 seconds) featuredigittimeout = 1000 courtesytone = beep xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr [featuremap] blindxfer = #9 automon = *1 ; One Touch Record atxfer = # When I'm making a transfer, just after dialing the transfering number, I don't listen any ringing tone, but the transfer is made correctly. In detail, when I press # I hear Transfer and the dial tone is played. I dial the extension I want to transfer and I don't hear any ringing tone but the transfer is beeing made. Is this OK? Thank you very much. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] using voip software client as public address system. Low volume
Hello all. We have an asterisk working perfectly but we need a sollution for the PA system. Before Asterisk PBX we had an expensive analog PBX which plugged an extension into an audio amplifier, and that was the PA system. Now, the Asterisk server is quite far from the audio amplifier and it has no audio card. So my idea is to plug the audio card of another linux server, which is over the amplifier, into the amplifier. I've configured a pjsua with auto answer but the audio is very poor, very low volume compared to a normal audio playing (like 'aplay ttt.wav'). Is there any way to increase the volume of sip calls? Is a client side configuration, a server side or both :) Any ideas? Please, I'm going mad. Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] some questions about atxfer usage
Hi I just press * to retrieve the caller again - Have you tried that? No, I haven't. Thanks, it's perfect for me. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] some questions about atxfer usage
OK
Thank you very much.
On 11/16/06, Alberto Pastore [EMAIL PROTECTED] wrote:
Antonio Almodóvar ha scritto:
Hi all.
I have enabled the attended transfer feature in features.conf. I'm
using it and I want to resolve some questions, I hope someone can help
me :)
When I transfer a call to an extension:
- The extension rings during 15 seconds and the call returns to the
transferer. Is there any possibility to recover the call before the
timeout of 15 seconds expires?
I mean, I would like to personalize the way of making transfers using
the feature of atxfer. How can I do that?
Thanks in advance.
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Hi Antonio.
Taking a look at the following code line from res_features.c:
newchan = ast_feature_request_and_dial(transferer,
Local,
ast_best_codec(transferer-nativeformats),
dialstr,
15000, // ---
outstate,
cid_num,
cid_name);
I assume that 15000 msecs is a hardcoded value...
You might want to replace it with some variable taken from
pbx_builtin_getvar_helper() results
but it involves recompiling at least the res_features.c module;
something more or less
like this (I haven't tested it!!!):
//these two lines go at the beginning of the if {} block
char *transfer_timeout_str;
int transfer_timeout = 15; //default value
//these lines replace the newchan = ast_feature_request_and_dial(...) one
//read the value (if any) from TRANSFER_TIMEOUT
//can be set in extensions.conf's [globals] (TRANSFER_TIMEOUT = 30)
transfer_timeout_str = pbx_builtin_getvar_helper(transferer,
TRANSFER_TIMEOUT);
if (transfer_timeout_str) {
transfer_timeout = atoi(transfer_timeout_str);
//sanity check
if (transfer_timeout = 0) transfer_timeout = 15;
}
newchan = ast_feature_request_and_dial(transferer,
Local,
ast_best_codec(transferer-nativeformats),
dialstr,
transfer_timeout * 1000, // ---
outstate,
cid_num,
cid_name);
Bye,
Alberto.
--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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[asterisk-users] some questions about atxfer usage
Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the transferer. Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like to personalize the way of making transfers using the feature of atxfer. How can I do that? Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ethernet interface shares interrupts with tdm card
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 rtc12: 170166219 XT-PIC eth0, wctdm14: 398500 XT-PIC ide0 NMI: 0 ERR: 0I've tried modifying parameters in the bios and I didn't managed to change the irq.Does anyone have a machine like mine?Have anyone changed the irq in order to not sharing irq's? Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ethernet interface shares interrupts with tdm card
Thank you for replying.Icannot assign specific IRQ to the PCI or ethernet slot via bios, I can assing IRQ to usb, serial,... but not to the PCI or Ethernet. Maybe both devices must share the same pci, I don't know. Anyone who has this motherboard? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: E1 te110p problem
Gracias por las respuestas.Estoy usando el AMP (ahora llamado FreePBX). Añadiendo las troncales usando esta herramienta. El plan de marcado es el que usa el AMP;zapata.conf:[channels]language=escontext=from-pstn switchtype = euroisdnusercallerid=yescallwaiting=nocallwatingcallerid=nothreewaycalling=notransfer=yescancallforward=nocallreturn=nocallerid=asreceivedechocancel=yes ;echocancelwhenbridged=yesrxgain=0.0txgain=0.0overlapdial=yessignalling=pri_cpegroup=1channel = 1-15channel = 17-31Los problemas creo que estaban relacionados con la empresa que nos suministra el primario e internet (ONO usando LMDS), porque internet también falló durante el no funcionamiento del primario, cosa de la que me enteré después. Ahora funciona todo bien pero no las llamadas al 902, respasaré el dialplan y comentaré los problemas que vaya encontrando en Español, si no hay ningún problemaGracias a todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 te110p problem
Hi all.I'm using a te110p in spain. ;zaptel.confspan=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16I'm getting problems dialing out through this span. ¿How can I debug its behaviour? Thank you in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
