similar to: Ring back issue with asterisk 1.8.18.0

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of

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 has something to do with asterisk 1.6. The other

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,

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

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 particular zone. The message is printed

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

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

I don't expect my SIP provider to provide useful "Remote-Party-ID" information. Therefore, I am using "CONNECTEDLINE(num)=xxx" AND "CONNECTEDLINE(name)=yyy" to populate remote party information from a local database. I am also using the "I" (upper case "i") option for Dial. Generally I like to see to see the remote party name appear on the

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

How do you setup the timing in Meetme conference? I have a x100p and tdm4x card. When I dialing to my conference I get a request to schedule in the past error message. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 10, 2004 10:48 AM To:

Hello, I am using an incoming iax provider to bring calls to my server. When an incoming call comes in, the calling party does not hear a ring tone. I figured that this was no big deal, and that I just needed to enable the "r" flag to dial. This has not fixed the situation though. Just to try to make sure the line was being picked up properly, I tried the following: [inbound] exten

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 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

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

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 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

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 >>

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, 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

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