similar to: Cannot hear ringback music from telco

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! 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

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

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

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

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:

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

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

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

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

Hi All We have Grandstream 102's running ver X.18. When hanging up after a call has been made the grandstream seems not to disconnect the call and when you put the handset down the phone rings only to pick it up and be on the same call. This is happening quite often and gets very irritating. Can anyone help with this? Regards Doug

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 >>

Hi, I have PRI configured and up but when i am dialing outside i am not getting any ringback tone but my call is connected. following is my example SIP----------------->PRI ------------> mobile I have set progress=yes in chan_dahdi.conf but still not working if i call inbound from my mobile to internal extension ringing working please help me -S -------------- next

hi: When iam sending calls through sip a fake ringback tone is generated and then call status can't be viewed (if call is ringing,busy,offline) it just rings and rings. Can i disable this? Thanks in advance. _________________________________________________________________ Windows Live?: Keep your life in sync.

Hi What command i need to use to make a call with oh323 and hear the ringback sound Thanks

I'd like to add a test extension to implement ringback so that I can test a phone's ringer without having to use another channel in another room. The way I'd like to implement this is to dial a test extension, get a tone, hang up, then one second later, have the system call me back at that extension. There is a way to do this which is mentioned in the Asterisk white paper, but it

Hi, I use in Production : Asterisk 1.2.9.1 We Use Asterisk as a SIP Transit Server to record centrally all the calls. The call flow would be: incoming calls : PSTN -> GW -SIP-> Asterisk(Record) -SIP-> Softswitch -> IP Phone outgoing calls : IP Phone -> Softswitch -SIP-> Asterisk(Record) -SIP-> GW -> PSTN Dial plan in Asterisk is quite simple: [record] exten =>

In case it wasn't obvious in the DAHDI release announcement. Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140922/0bb998aa/attachment.html>

Good morning, i've noticed many times that there are IVRs that play a ring tone just before bridging me to an agent. My asterisk does not behave like this but i've always wanted to. I'm now playing with 1.6.2.9 and i've read in queue's doc: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue R ? stops moh and rings once an agent is ringing (Asterisk Trunk) (in