similar to: no ringback from SIP server when originating call

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

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

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

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,

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

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:

Hello List, I'm working on an autodialer project. At the moment I use the Originate application then I "throw" it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate finish its bit and I have lots of wasted time or even worse, the remote party hanging the call because instead of a human

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 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

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

Good morning everybody, Can someone explain to me the interconnection between these four things: indications.conf, SetLanguage(), zaptel.conf and ring-back ? If there is any !! :- ) I am having this case where some users cannot hear ring back from a DeadAGI script and it seems to be interconnected to these items. These users are from the iaxfriends table, they _can_ hear ring-back from a

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 >>

Has anyone used the G option with the Dial app? I'm looking for a way to control the called party leg. Specifically, I'd like to pass a few variables to the called side for some call control. Here's a synopsis of what I'm doing: Make outbound call w/ AMI Originate action. Called party answers ("Customer") Customer identifies himself, and now I use Dial w/ the G

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 originating a call using the "Originate" action in the Manager API. It calls one party, then when they answer does the "Dial" application and calls another party and connects the two. Is there a way using the Manager API to check back later on the status of this call (is it still up, etc.)? I have found the "Status" API action, but I don't know how to get

Chris Abel writes: >Hello everyone! > >I've had this problem for a while and cant figure it out. When an outside >caller calls an extension on my asterisk system, they do not hear any >sort of ringing. Inside extensions calling other extensions do hear >ringing. We have 3 other asterisk systems that are configured the same >way, but do not have this problem. We think it

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

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. 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