similar to: New feature request

Well, I think I discovered even further why there is no ringback tone available. The following message, is displayed on the console in asterisk. ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Looking more into it, I found that it was related to loading tones for a particular zone. The message is printed

I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with Asterisk. It works ... Partially. We are using the Inalp to connect ISDN phones, it basically acts like an ISDN ATA. First of all, when I make a SIP call to the unit with a simple Dial() command (no "r", so Asterisk shouldn't generate its ringback tone) I hear Asterisk's ringback tone anyway (I'm

Hi Bart, I guess that's because the value of the last argument of llvm.memcpy has to be known at compile time. Nicolas Bart Coppens wrote: > Hello, > > I'm getting the impression that I'm misunderstanding something about > intrinsics. I have the following reduced testcase in mind: > > declare void @llvm.memcpy.i32(i8*, i8*, i32, i32) nounwind > > define

I would just like to follow-up on the ringback problem I'm getting from *. As I've said in my previous post, I am not hearing the "real ringback" from the Cisco gateway terminating my call. I don't want to provide false ringback from * (r option of dial), because it'll still give me ringback even if I am suppose to hear announcement or fastbusy. Below is captured ISDN

I am trying to get Asterisk to emulate the sounds of the earlier telephone systems, and the settings in [us-old] are pretty helpful. The only thing lacking is ringback tone, which is not quite as complex as the real phone systems of the day. For example, it is true that a ringback tone commonly used is 420Hz modulated by 40Hz. This is what shows up in [us-old]. But that modulated tone was

Hi, I have the following situation: At a branch , there is a Cisco Call Manager with users all having Cisco phones. Now I put down a Asterisk 1.2.12 box at the branch, which talks H323 via chan_oh323 to the CCM. So calls go from the CCM, go H323 to the local Asterisk box, then I take it via SIP to another Asterisk box. From there I am hooked up to 2 different providers, for Local and

I don't expect my SIP provider to provide useful "Remote-Party-ID" information. Therefore, I am using "CONNECTEDLINE(num)=xxx" AND "CONNECTEDLINE(name)=yyy" to populate remote party information from a local database. I am also using the "I" (upper case "i") option for Dial. Generally I like to see to see the remote party name appear on the

Hi to all I'm a new user of TDM400P card. The configuration is OK and I have no problem for incoming call. When I try to place a outgoing call towards a PSTN number the call is not always answered. In other words TDM400P seems to not understand that the call has been accepted from the remote party. In other cases (different extension) the call is accepted succesfully. In my opinion TDM400P DSP

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users

Hello! An extension registered at asterisk 13.23 initiates an external call (pjsip). After the Invite, the callee (-> ISP) sends a 100 Trying 183 Session Progress (*without* SDP) Asterisk now sends to the extension: 183 Session Progress (*with* SDP) 183 Session Progress (*with* SDP) (really two times) The callee meanwhile sends 180 Ringing (*without* SDP) which is

For the past week or two, our customers who have Polycom phones have been experiencing a problem... but our customers with Cisco phones do not have this problem. The phones in question are: Polycom SoundPoint IP300 (firmware 1.3.1 or 1.3.4) Polycom SoundPoint IP500 (firmware 1.3.1 or 1.3.4) Cisco 7960 (firmware 7.2 or 7.3) The problem is this: when our Polycom users dial _some_ PSTN numbers,

On 12.12.18 at 19:43 Joshua C. Colp wrote: > On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote: > > <snip> > >> >> The problem: The extension doesn't create a ringback locally, because >> it most probably expects it to >> be sent by the callee - but the callee doesn't send anything (not >> surprising, because there has been >>

I can smile now. I made my * work with my Cisco. Finally. First problem was Ethernet on my Linux. After installing * on a different machine, I got rid of that "icmp udp unreachable" error. My next problem was the call stays on on Cisco gateway, but the ATA drops it. I figured out it was my mistake on dialplan in extensions.conf --- (it took me a day to notice it.. damn!). my config

Asterisk and the one that doesn't work returns 100 trying followed by 183 session progress. It is my understanding that 180 ringing causes ringback to be generated by the callee, while 183 means that the caller has early media and will send ringback through RTP. Anyone have any idea why I wouldn't get ringback in this case? Should Asterisk be passing through the early media to the first

Does anyone have disconnect supervision working on their TDM400P or X101P cards (to/from the telco)? Googling "telco disconnect supervision," I found the following description: http://mirror.lcs.mit.edu/telecom-archives/telecom-archives/TELECOM_Digest_Online/1559.html [cut] Re: How Do I Get "Kewlstart" From my Phone Company? Paul A Lee Tue, 24 Aug 2004 16:58:16 -0400 ... It

> > > > I would be grateful if anybody could tell me what I should > > tell Verizon > > in NJ so they would enable "disconnect supervision" for my lines. > > > > Apparently "remote hangup notification" or "disconnect > supervision" or > > "calling party control" is NOT the magic phrase for them. Although >

Hello, everyone. Could anyone explain me how does ringback detection works in asterisk. Sometimes, when making a call, my asterisk box doesn't detect a ringback and I just hear silence until the other party picks up the phone. I've checked the SIP messages and they are ok (I'm getting 183 "session in progress"), so I guess I should be debugging the RTP packets. From then on

I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in

I am new to the sip side of things and have a question regarding ringback. I don't hear ringback when using the sjphone softphone when dialing internal extensions. It's fine when dialing outside over the pstn. Is this a issue of the softphone, configuration or sip in general? Thank you, -gcc

I have an application that uses the Asterisk Management Interface to bridge two calls using the Originate command with Dial as the action. Using one SIP server, there is no ringback on the second leg of the call. The first person is called, answers, and hears silence until the second person picks up, even though the second person's phone is ringing. When the call goes to another SIP gateway,