similar to: Re: Asterisk-Users Digest, Vol 22, Issue 26

Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the channel, i cannot hear nothing. If I change the configuration and i send the call to an internal sip

OK, this is probably a dumb question for a lot of you, but I have no experience with digital lines outside of a tiny bit of ISDN, so I'll just bite the bullet and ask some newbie questions. I am attempting to plan an asterisk installation with about 20 SIP phones and the following incoming lines: 1) At least 6 (as many as 10) lines for voice to the SIP phones 2) 2 incoming/outgoing fax 3)

Hi, is there someone who speaks in Italian? I'll try to explain in english my problem, but if there is someone who speaks italian i think it would be better for me. I'd like to use asterisk only as IVR and call diverting. I have only one phone line, and no other phones, all the calls arrive at one number. I would like something that answare, and depending from the 'street'

I connected two plain old telephones to FXS lines of a TDM400P (defined as fxoks in zaptel.conf), one of them dials the other with Dial(Zap/2), I talk to myself for a while, hang up either of the phones, and the phone that remains off-hook gets the congestion tone until it goes on-hook (at least as long as I've cared to wait). I don't have a voltmeter on the line, but if I'm hearing

I had a similar issue on my Debian box. It seemed that setfacl didn't care for special characters. I changed the separator character to - (dash) instead of + or \ and it worked fine. Good luck! Tom Hallewell Radio Free Asia Washington DC USA > > (offlist replies discontinued due to increasing large number of people > involved) > > Gareth Davies wrote: > >

Hi Guys, I recently got a TDM400P 4 FXO for use in the UK, this at the time seemed like a good idea as I had good results with an X100P clone. Installation went great and call clarity is excellent no echo like I had on the clone card. My problems start with detecting hanging up the line. If a person calls into the system and speaks to me on a SIP phone when I hang up the call clears

Hello all! I have tried and tried to resolve this one to no avail. Hopefully one of you can help... The system in question is a Compaq Evo D600 (iirc) business desktop, with a 1.4GHz Pentium 4 and 512mb of RAM, running a stock install of PoundKey 1.2. It has two Digium cards installed: a TDM400P with four FXO modules, and a TE110P hooked to a Carrier Access Adit 600 which serves 8

Hi all! I have an issue with TDM400P FXO card. When a call enter into my IVR and select the proper option, the person that ansswer the call say your "thanks for contact us ..." but the caller cant hear this words because a delay between asterisk and caller part or between asterisk and the ATA device. What is the item on zapata.conf that can affect this delays. Thanks for any help

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, April 13,

Hi All, I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4 PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?, other detail is: this card have 4 card green. I need to know what is the best card for the following scenario: I need a IVR for my comapny and a PBX, but i want that my extension not use FXS I want IP phone . Thanks ins advanced,

Hi, I am setting up a test call center using *. I am using one Zap channel (Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip phones (SjPhone) for call agents. I have setup queues and agents. While testing I found that if the agent presses * key in soft phone while attending calls Zap channel gets hung up, and another customer can call. But if the caller hangs up (for example

Hi , I have an asterix (1.2.7.1) box with 3 TDM400P and 10 FXO modules, and zaptel (1.2.5). The problem is that 6 modules can't be initaized and i get the folowing error: May 7 09:17:37 WARNING[7739] chan_zap.c: Unable to specify channel 9: No such device May 7 09:17:37 ERROR[7739] chan_zap.c: Unable to open channel 9: No such device here = 0, tmp->channel = 9, channel = 9 ztcfg output

Hi ! I have some issues, i don't know exactly if it's a busy detection issue. When i dial into the Asterisk box, and if i hang up before the Asterisk answers with the IVR Welcome message, the Asterisk goes on with the call. But, if i wait for the Asterisk to answer, and if i hang up, the Asterisk hangs up too. I have this parameters on zapata.conf: busydetect=no

Hello everybody, Which one is a better choice 1. Gateway device with FXO <-> SIP ( example Addpac http://www.addpac.com/addpac_eng2/addpac_product_view_detail.php?class_id=19&item_id=59 ) 2. Digium (Wildcard TDM400P) 3. Sangoma (A200 Analog FXO/FXS) All i need is to put asterisk in place with 4-8 incomming lines (ordinary POTS ). With IVR, Voice mail and International Call via SIP.

Hi all, I have an existing panasonic analog pbx in use and a asterisk server with digium tdm400p(2 fxs and 2 fxo). channel 1 -> fxs -> telephone channel 2 -> fxs -> telephone channel 3 -> fxo -> extension 15 at panasonic pbx channel 4 -> fxo -> phone line from telco We call in to fxo (channel 4) and enter the ivr which prompt us to enter the extension number. After

I thought I had posted this, but I didn't see it in the archives, so I guess I hadn't. I've got FXS lines going to a legacy IVR. When I Dial into one of these lines and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I would like the IVR to hang up sooner. I could do this by either making the IVR recognize the standard Congestion tone, or changing the

With one IVR payment system, I noticed quite a difference in DTMF transmission between these two cards. The IVR missed nearly all digits from X100P, while receiving digits from TDM fine. Since neither card process or synthesize audio, what can the difference be? (This particular IVR has problem with some regular phone devices, too.) Yuan Liu

I have a line from BT (UK) connected to my asterisk system, on a TDM400P. I am able to see either distinctive ring cadences or caller ID but not both. If I try to enable both, all drings show up as 0,0,0. This is a pain because, if I make a call out over that line and the number I call is busy, I can elect to camp on it (ringback), which results in a different cadence of ring when the called

I realize I cannot use inband audio for phones (voicemail and internal ivr, password for external trunks and other thing not working) So I put everywhere rfc2833. Doing this, anyway, make any EXTERNAL IVR NOT working. I see a lot of posts about this, but no solution, becouse using inband audio (which works for outside...) breaks inside IVR Is it possible to define to use inband audio ONLY on