similar to: ringback detection

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

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 >>

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:

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 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,

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

Hello, everyone. I'm having a strange problem with my asterisk. After dialing and before the other side picks up the phone I should hear the tones (I'm not sure what are they called: piiiiiiiiiiii---piiiiiiiiiiiiii....) and in almost all cases that is true. However there is a range of numbers where I'm having this problem. There is no tones, just silence, until someone picks up the

I am noticing that calls that come from our IAX pstn gateway provider and terminate to our Asterisk IVR do not receive ringing when an extension is dialed. For example: 1. An inbound PSTN caller calls our number 2. Asterisk answers and provides greeting 3. PSTN user dials extension of internal SIP phone 4. No ringback is heard from PSTN callers perspective 5. SIP user picks up or the

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

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

Hi, I am using asterisk 1.8.22 and have a problem when calling in parallel several SIP endpoints and I am not sure how to resolve it. In this case Asterisk will not bridge any audio to the caller before the 200 OK. Which means any progress announcements, including remotely generated ringback, are not passed back to the caller. This behavior is completely correct, because there is no way to know

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, 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

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.

Hello! I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2. As customer of German Telekom, I have three numbers and therefore three trunks - each number is bound to one trunk. After migration, some callers complained about missing ringback tone: they didn't hear any ring tone and where surprised that they suddenly got me anyway :-). The connection afterwards was as expected. Deeper

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

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

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

Hello, My company is doing some * testing with our Class 5 softswitch and had some questions regarding ringback being provided to our PSTN users (off --> on net calling) Currently with MGCP subscribers, we know the PSTN ringing is provided by a digital PBX for example, However, it looks like with SIP, our softswitch is relying on MGCP signaling on our PSTN gateways to provide ringback

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