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When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks.

...nsion: 183 Session Progress (*with* SDP) 183 Session Progress (*with* SDP) (really two times) The callee meanwhile sends 180 Ringing (*without* SDP) which is "forwarded" by Asterisk to the extension with 180 Ringing (*with* SDP) 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 no SDP). Or should Asterisk create the ringback (Asterisk doesn't send any RTP package)? Or should the phone create the ringback itself becau...

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the phone? We only use this simple dialplan to make outgoing call: exten => _9.,1,Set(CALLERID(all)=...

...mpa,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 Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections 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...

...lSets I can send calls to this extension with this: exten => 1,Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/4165555555,30) same => n,VoiceMail(4165555555 at VoiceMail,u) same => n,Hangup() Up to this point everything works as expected. I call in and I hear a ringback until the extension is picked up. Now I add a virtual PBX to the mix. [pbx-17842] exten => s,1,Verbose(0,${CALLERID(all)} Calling PBX 17842) same => n,Answer same => n,Wait(2) same => n(announce),Background($SOUNDS/pbx-17842/announce) same => n,WaitExten() sa...

Hello. I have another issue. When I call in, everything is processed correctly, including voicemail, but I don't hear any ringing/ringback. exten => s,1,Zapateller(answer|nocallerid) exten => s,2,NoOp exten => s,3,Playback(pls-wait-connect-call) exten => s,4,Dial(${PHONE1}&${PHONE2}&${PHONE3}&${PHONE4},15,Ttm) exten => s,5,Answer exten => s,6,Wait(1) exten => s,7,Voicemail(u${PHONE1VM}) exten =>...

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 h...

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 generally overlaid on top of real ringing, i.e. 20Hz. So, using the Ast...

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 Asterisk? The relevant settings in oh323.conf are:...

...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 International, both via SIP. The problem I am having is that the users dont get ringback (ringing indication) when they dial International numbers, yet it works perfectly when they dial Local numbers. Yet, to test, from a hardphone plugged into Asterisk2, I get ringback, so its not the Interntional provider, it must be the SIP trunk from Asterisk1 to Astrisk2....

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...

Hello All, I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using Asterisk@Home 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. >From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Does anyone know how to resolve this problem? Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/224c3e2a/attachment.htm

...artnode 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 sure it's Asterisk generating it because by changing the country it indications.conf, the ringing changes). That's what I see in the CLI: -- Executing Dial("SIP/2017-71be", "SIP/070@inalp|90") in new stack...

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 par...

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 >> no SDP). >> >> Or should Asterisk create the ringback (Asterisk doesn't send any RTP >&g...

Has anyone here implemented "Ring back when free" in Asterisk? The way it works in the UK is as follows: 1. A calls B. B is engaged (busy). 2. A hears "The number you called is busy. To use ringback, press 5" 3. A presses 5, and hears "Your ringback request has been accepted". 4. A hangs up. 5. Later, B hangs up. The system then calls A (if A is now busy, it waits until A is clear again). 6. If/when A answers, the system calls B on A's behalf and A hears ringing. Any imp...

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6...

...ry 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 can detect if the RIngBack tone is present or not. The call is declared answered if the ringback tone stops to occur. So my feeling is that the ringback ton detection fails form my country (italy) in most cases even if sometime it works fine. On Asterisk cli I can read 1) when the call is succesfully accepted host529*CLI&gt...

...r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party hears the please wait while I connect your call, but does not hear any ringing. I tried inserting exten => 1,1,Ringing but that does not work either. The same call flow from the pstn DOES generate ringback: [fromPSTN] exten => s,1,DigitTimeout(2) exten => s,2,ResponseTimeout(10) exten => s,3,Wait(1) exten => s,4,Background(custom/ivr-greeting) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},15,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1...