similar to: Generating a different countries ringtone on a per call basis

Hello list, when I sent an incoming call first to a queue and after the timeout to a dial-command, while the correspondent's phone rings there is no ringtone for the caller... So it goes like this : 1. dial(SIP/account1,20) 2. queue(myqueue,,,,20) 3. dial(SIP/account2) In step 1 there is a ringtone for the caller. In step 2 there is musiconhold (class default) for the caller. In step 3

Would it be possible to set the ringtone based on the number that was dialed? Example of what the goal is: Dial Denver number Incoming calls ring with ringtone 1 Dial main number Incoming calls ring with ringtone 2 We are currently using Digium D40, D50, D70 phones. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have uploaded a custom ringtone to our SPA-922's for the purpose of sounding like a door bell chime when the doorbell is pressed. I am using __alert_info to set this ringtone. It appears that I can only set the ringtone via alert_info for the ringtones that come from Linksys. Has anyone else seen this issue? I am doing the following: exten => 100,1,SetVar(_ALERT_INFO=doorbell)

i could add "r" option in dial command. this will generate a ringtone during connection. could i change this default ringtone? i tried indications.conf but not success. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/77a4d48f/attachment.htm>

Hi, is there a way to enable dtmf detection on zap channels? I am trying to pickup, play a ringtone and the dial out. I.e. exten => s,1,Wait,1 exten => s,1,Answer exten => s,2,Playtones(dial) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => _X,1,StopPlaytones exten => _X,2,Dial,Zap/g8/BYEXTENSION|10

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

Hi, i have a problem with iax2 and ringtone. Here is the call path pstn -> asterisk -> iax -> firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really

Hi, I'm having trouble setting the ringtone on my Polycom 501. The relevant entry in extensions.conf is: exten => 801,hint,SIP/creative1 exten => 801,1,SetVar(ALERT_INFO="Test") exten => 801,2,Dial(SIP/creative1,20,Ttr) In the sip.cfg: <alertInfo voIpProt.SIP.alertInfo.1.value="Test" voIpProt.SIP.alertInfo.1.class="13"/> and <TEST

I should've mentioned that before. I've tried doing that and it has no effect. I've tried both upper and lower-case 'r's. I've also tried a workaround that I thought would work, but it doesn't: Answering the call and then using the playtones(ringing) command before connecting to my cellphone. -----Original Message----- Date: Sat, 1 Apr 2006 19:59:46 +0100 From:

Hi Community! If this issue was already topic, please excuse or delete my request... Topic 1 "no ringtone": I configured a SIP registration with my SIP provider (SIPCall). Everything works fine except the ring tone for the caller. The caller hears silence until the called party takes up the phone. I used the DIAL command with the r and R option but no luck... :( Has anybody the same

Hello, would like to have distintive ringtone for internal calls, google gave me blurr answer. My extensions are 46**, any calls made within 46** I want to ring differently than external calls. Thanks in advance.

Hi I'm new to this list, so please forgive me off-topic or RTFM-questions. I have an asterisk/elastix driven phone-environment using Polycom SoundPoint IP 650 as extensions. When adding just one custom ringtone (~57KB) in a proper format (ML.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz) the phone boots well. But after I have chosen the custom ringtone as my

Hello Everyone. I usually find my own solutions for problems but this time, after several months, I've given up. My asterisk is set up so that incoming calls from my voip provider ring on both my sip extension and my cellphone at the same time. When the system receives an incoming call, ringtones indicating that the call is being connected play normally for the first 5 seconds to the

I have just successfully got the TFTP file remapping to work such that I can have unique ringtone files for each and every extension. I added the following to my server_args line in the xinetd configuration for TFTP: -m /home/asterisk/grandstream/ringmap.cfg Now the entire line reads: server_args = -v -s /home/asterisk/grandstream -u asterisk -m /home/asterisk/grandstream/ringmap.cfg (There

If you wan't to create a ringtone with makering.pl for firmware 1.0.50, be sure to create it as ring.bin and then rename it to ring1.bin / ring2.bin or ring3.bin. This seems to be the only change between the format from 1.0.4.68. Regards, Maron

I wouldn't normally post this to the asterisk mailing list but I'm really stuck... I've been trying to get meetme working on and off for a few months now but I always hit a brick wall when trying to compile. I keep seeing this... make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o

Have anybody experience with the following error on a linux system. The system is for the rest running perfect without problems. The system was full installed with Suse 9.1 and updated. Following uname is my kernel 2.6.5-7.104-default Greatings Han Suse 9.1 professional AMD Atlhon XP 2200 Asus A7V600-X bios 1005 1Gb memory 400Mhz Geforce MX 4000 64MB 40 GB Harddisk

I'm trying to find a way to generate an 'internal extensions' tonelist but I can't seem to find anything on how to do this. My idea was to start a Playtones(intercom) tonelist and not indicate ringing to the line (dead air). But then, somehow StopPlayTones needs to be run once the ringing telephone picks up. This seems like a dirty way to do this. I envision an option to the

I am getting this: make gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pritest.o -MF .pritest.o.d -MP -c -o pritest.o pritest.c pritest.c:50:28: fatal error: dahdi/tonezone.h: No such file or directory #include <dahdi/tonezone.h> I am using dahdi-linux-complete-2.11.0+2.11.0 I did "find . | grep timezone.h" from the dahdi root and its not

I have been looking for the specs (format, bit rate, ect) on custom ringtones for digium phones. Using the DPMA how would I deliver the ringtone to a digium phone? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120820/cb4927d0/attachment.htm>