similar to: Providing Ringback

I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1 and call leg from gsm gateway is using codec gsm. I am having one way audio and getting below mentioned warning. Asterisk version is 1.8.11.0 [Jun 2 17:08:28] WARNING[21652]:

Hello, I'm fairly new to asterisk and Trixbox, I'm setting up a Trixbox based email to fax gateway. At this time, I have a ZAP PRI link between the eFax server and my VoIPSwitch. The ZAP channels are configured, the B and D channels are up, and I have green link lights on either end of my cabling, but when I dial the number I have assigned to my eFax server, the call never seems to route

I am using asterisk. I also have an access to VOIPSwitch ver 2 where I can see live calls. Many times I have seen that my calls are getting strucked and then it gets disconneected after 59 mins ( as settings are done accordingly in VOIPSwitch) What could be the reason ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, i need help. My problem is that when I want to call (using ISDN phone or internal SIP client) via the Sip provider a real phone number (ISDN phone or internal SIP Asterisk >> SIP ), I get a ring tone. When I now decide to hang up (e.g. if nobody answers), the called telephone continues to ring almost forever. the sip.conf: [2563105] accountcode = 2563105 username =

Hi all. I am having lots of trouble with random calls dropping after 20 seconds, and I finally managed to capture a full sip trace. I'll paste it in full below, but I'll give a summary first. It seems that Asterisk is not recognizing the ACK messages that it receives from the Grandstream ATA. This happens only on the ACK that follows the OK that marks a call as established. This makes

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, 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

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

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

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, 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

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

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

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

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

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