[asterisk-users] Call Record and Playing
Hello List, I am using the following dialplan exten = s,1,NoOp(IVR recording test) exten = s,2,Answer() exten = s,3,Background(say-temp-msg-prs-pound) exten = s,4,Record(/var/lib/asterisk/sounds/custom/tempcustommessage.gsm) exten = s,5,Background(to-listen-to-itpress-1) exten = s,6,Background(to-accept-recordingpress-2) exten = s,7,Background(to-rerecord-announcepress-3) exten = 1,1,Playback(/var/lib/asterisk/sounds/custom/tempcustommessage) exten = 1,2,Goto(s,6) exten = 2,1,System(cp /var/lib/asterisk/sounds/custom/tempcustommessage.gsm /var/lib/asterisk/sounds/custom/ORIGINALMESSAGEname.gsm) exten = 2,2,Playback(auth-thankyou) exten = 2,3,Hangup() exten = 3,1,Goto(s,3) The recorded message cannot be played I get zero size file error as below:- -- Executing [1@test-pstn:1] Playback(DAHDI/i1/XX-b9, /var/lib/asterisk/sounds/custom/tempcustommessage) in new stack [Jun 2 17:27:30] WARNING[28250][C-1200]: file.c:492 filehelper: File /var/lib/asterisk/sounds/custom/tempcustommessage.gsm detected to have zero size. Has someone come across this issue and fixed it. -- Regards, Ashish Agarwal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Half-height PCIe analog FXO card
Thanks Eric Tim, I'll put Sangoma back on the shopping list then :) I also found a thing called a Jingletel A400EL which looks like an A400-module-compatible low-height PCIe card but the name suggests it's yet more Chinese knock-off cr*p. With postage to the UK, it's about the same price as a Sangoma, which in turn is about the same as I paid for the server (after cashback). Harumph. I wonder if I could hacksaw an A400E into fitting. If anyone in the UK has an old Sangoma A200 (I only need 1x FXO, if it comes with 2x FXS as well that's a brucie bonus) they'd be willing to sell for up to about 60 quid, please drop me a line. Cheers, Ade. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: 01 June 2012 15:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Half-height PCIe analog FXO card Last time I checked (a few years ago) Sangoma has half height brackets available. Contact their support or sales. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ade Vickers Sent: Friday, June 01, 2012 10:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Half-height PCIe analog FXO card Hi, Does anyone do a low profile PCIe FXO card? I just picked up an HP ProLiant microserver for $nuppence, which I'd hoped to migrate my Asterisk setup onto. I currently use an A400P analog card, but the ProLiant only has PCIe slots, and they're short ones too, so I can't use an A400E card. Even the Sangoma cards, which seem to be low profile, have full-height brackets on them - which, of course, won't fit in the box. Is it just me, or is this whole half-height PCIe thing a complete b***ocks? Any advice appreciated. I'd prefer not to have to spend mega$ on this, the server only cost $200, it seems silly to spend $1000 on a PCI to PCIe converter (Magma.com) to keep using a $100 card... Cheers, Ade. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music instead of Ring Ring
Hello, How can I achieve to play a music file instead of typical ring ring (something like MusicOnHold)... I have the following dialplan, the time when the user calls the context is executed and the system calls both the user and I hear a Ringing sound. [inc-call] exten = s,1,Dial(DAHDI/i1/USER2DAHDI/i1/USER1,20,A(sound-file)) Please suggest -- Regards, Ashish Agarwal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Escape digits recognition during recording
Dear, would you please let me know how to recognize DTMF during recording? for example escape digits ? in asterisk 1.6.2.X? .. $agi-record_file(a,gsm,1234567890); .. -- Pezhman Lali -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Le 30/05/2012 15:02, Andres a écrit : Considering that you made progress on your initial problem by setting nat=force_rport (resulting in connected calls with no audio) and now you're mentioning the use of externaddr, I'd recommend a very careful reading of the NAT SUPPORT section of sip.conf.sample in the configs directory of the Asterisk source tree. In Asterisk 1.8, there is a new configuration option named media_address which may be of particular interest. It sounds like a NAT issue to me too. Why don't you do a quick test and put the Asterisk box on a public IP if you can. If it works, you will have narrowed down the issue to a NAT problem. You could have a nat router with a broken SIP ALG. Back to the story: even out of VM -which means on a public IP- the timeout problem till appears. And more odd, if a communication start, the call get hanged up because of this timeout :-( All peers and users are setted with nat=yes, phones connected to Asterisk have directmedia=nonat and peers gateways have directmedia=yes. Remember, we only face this problem with Dellmont services and asterisk 1.8/10. Previous asterisk versions are working well. Does someone else use Dellmont services (VoipBuster, SipDiscount, Intenetcalls, Voicetrading, ...) with asterisk 1.8 or 10? If yes and without problem, would it be possible to share configurations? Thanks for your help. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi 2.6.1 with OSLEC support
In order solve my incoming caller ID problem, I upgrade the dahdi to version 2.6.1 from version 2.4.x. After upgrade, I found the echo cancellation doesn't working (I'm using Digium AEX800B PCI Express card). I can hear my self talking on the phone. How to solve this? I think I need to recompile dahdi 2.6.1 with OSLEC support? how? [root@callcenter ~]# dahdi_cfg -vvv DAHDI Tools Version - 2.6.1 DAHDI Version: 2.6.1 Echo Canceller(s): HWEC Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 04) Channel 05: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 05) Channel 06: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 06) Channel 07: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 07) Channel 08: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 08) Channel 09: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 09) Channel 10: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 10) Channel 11: FXS Kewlstart (Default) (Echo Canceler: oslec)(Slaves: 11) Channel 12: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 12) Channel 13: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 13) Channel 14: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 14) Channel 15: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 15) Channel 16: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 16) 16 channels to configure. Setting echocan for channel 1 to oslec DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) I try to change to /etc/dahdi/system.conf to fxsks=1 echocanceller=hwec,1 Still doesn't work and this error still occured: DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) [root@callcenter dahdi]# asterisk -rvvv Asterisk 1.8.7.0, Copyright (C) 1999 - 2011 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail s. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.8.7.0 currently running on callcenter (pid = 6908) Verbosity is at least 3 callcenter*CLI dahdi show channel 1 Channel: 1 File Descriptor: 12 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: Calling TON: 0 Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: no TDD: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Gains (RX/TX): 0.00/0.00 Dynamic Range Compression (RX/TX): 0.00/0.00 DND: no *Echo Cancellation: 128 taps (unless TDM bridged) currently OFF* Wait for dialtone: 0ms Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook callcenter*CLI -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
