similar to: Ring file

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 using asterisk 1.6.0 and -- not all the time -- when a caller comes in and my ivrdials an extension, the ring he gets sounds like a modem handshake instead of the normal ring tone and it only sounds once even if the phone is not picked up. Anyone seeing this -- the logs look fine as far as I can tell. -- Your life is like a penny. You're going to lose it. The question is: How do you

Hi, Just noticed Asterisk is not playing 'ring' tone as defined in indications.conf when Dial command is used with 'r' option. For example: [test] exten => 123,1,PlayTones(ring) exten => 123,n,Wait(5) exten => 123,n,Playback(demo-congrats) exten => 123,n,Hangup() exten => 321,1,Dail(LOCAL/123 at test/n,60,r) When I now dial with a SIP phone - 123 I can hear nice

For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone. I am using ztdummy and do not have any cards installed. My configuration is as follows. I am using Trixbox 1.2.2. Can someone please guide me into the right direction? zaptel.conf loadzone = au defaultzone = au zapata.conf [channels] language=au indications.conf [general]

Hi Everyone, I have a SIP turnk which works fine with both inbound and outbound calling. However, the only issue is that there is no Ring Tone if someone calls us. The phones used are Aastra and Polycom connected to the PBX via VPN (SIP). I do get an outbound ring tone, so it's not that there is any media loss between the phones and the PBX. But when the DID is called there is dead silence

Hi there I've got a rather irritating problem with my Asterisk server. Whenever someone tries to call me, the don't get the usual "ring-tone" when they wait for me to pickup the phone. I don't know if I've disabled this feature somewhere in my configuration files. Since I'm in Denmark, I've got an entry in the indications.conf file pointing to

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937]

I have a TDM400P card all installed, and I can call out from a sip phone, but when I call from the PSTN into the asterisk machine, as soon as the Answer() gets called, the dial tone changes and is sounds like there is a lot of static on the line. Below is the part of the dial plan for answering the call. exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,Dial(Sip/pfriedel,20,tT)

Hi, I have configured 3 users in my iax.conf, i am using iaxcomm phones. Iaxcomm has excellent voice quality although there is no ringing tones(either ring back or ringing tone),but i can live without right now. I find that for each user i want registered i have to add his name and his ip address.I have been using "host = dynamic".Isnt there any way that i can define a dialmap such as

I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to "connected" even the callee is still ring. How could asterisk distinguish the "ringing" and "connected" in zap channel? thanks.

I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian Option 61C. Calls either way drop with error "Channel 0/23, span 1 got hangup, cause 100". Can anyone offer insight into the cause and solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading matching zaptel & libpri, put the problem is identical). For testing, I tried a call from the

Hi all, using * 1.2.9.1 and this week all of the sudden calls keep ringing even after they've been picked up... Here's one users summary: When I pick up the phone, I hear a dial tone and I am able to dial out. But for some odd reason, the receiving line picks up while the outgoing line is still ringing. And the receiving line can hear everything while the phone is still ringing. I tested

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work.... It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore.... also I couldn't find any ticket saying this parameter

Hi there i have some problems with some clients of my asterisk box, i have some cases when a client tried to make a call and there is no ring back only a silence and then the call hung up. I dont know why this is happening. I have the following stable asterisk version: CVS-v1-0-01/18/05-19:49:31 I did the an update a few days ago, the version that i had installed before was:

I am testing analog phone and fax machine plugged into Rhino Channel Bank which is connected to TE412P card. This site is in China. I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4 I ran into a problem which is analog phone can hear dial tone and can make outgoing calls. Another phone (ether internal or external) can call the analog phone ***but the phone does not

Andrew, I don't think it's a Polycom issue. We took Asterisk out of the picture and had our Polycom phones communicate directly with an Audiocodes PSTN gateway. Unlike Asterisk, the audiocodes do not send 180 Ringing before sending 183 Session Progress, and the polycom's play the correct tones in this case. We WANT Asterisk to send progress tones in band. In our case it IS needed.

Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible "Message waiting" indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks!